From jonas.gauffin at gmail.com  Fri Oct  1 01:24:33 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Fri, 1 Oct 2010 10:24:33 +0200
Subject: [Freeswitch-users] ip-change detection
Message-ID: <AANLkTi=An5iCE=yNeuJFZRanp=J3bCNh9YdiiOB=TsPS@mail.gmail.com>

Hello,

How do I turn off ip-change detection?

Regards,
  Jonas
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From juanito1982 at gmail.com  Fri Oct  1 02:08:49 2010
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Fri, 1 Oct 2010 11:08:49 +0200
Subject: [Freeswitch-users] Error forcing iLBC to 30 ms
Message-ID: <AANLkTinnhr2JPN2Lki8m_TUWv_5tNV7HU=EOyk4+Cyww@mail.gmail.com>

Hello,

   I am trying to use wiki config about iLBC codec but I am not be able to
make it work. If I configure codec prefs as:

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA,iLBC"/>

I get:

-------------------------
2010-10-01 08:48:22.444357 [DEBUG] sofia_glue.c:3867 Audio Codec Compare
[iLBC:97:8000:30]/[GSM:3:8000:20]
2010-10-01 08:48:22.444357 [DEBUG] sofia_glue.c:3867 Audio Codec Compare
[iLBC:97:8000:30]/[PCMU:0:8000:20]
2010-10-01 08:48:22.444357 [DEBUG] sofia_glue.c:3867 Audio Codec Compare
[iLBC:97:8000:30]/[PCMA:8:8000:20]
2010-10-01 08:48:22.446360 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/dsp1 at 174.36.98.82) Running State Change CS_NEW
2010-10-01 08:48:22.446360 [DEBUG] sofia_glue.c:3867 Audio Codec Compare
[iLBC:97:8000:30]/[iLBC:98:8000:20]
-------------------------

Phone offers 30 ms, FS 20 ms and call gets cursed...

If I use:

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA,iLBC at 30i"/>
 or
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA,iLBC at 20i"/>

FS does not offer iLBC codec in Audio Codec Compare

Where could be the misscounfiguration? Which would be the best config?

Regards
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From steveayre at gmail.com  Fri Oct  1 03:15:24 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 1 Oct 2010 11:15:24 +0100
Subject: [Freeswitch-users] Error forcing iLBC to 30 ms
In-Reply-To: <AANLkTinnhr2JPN2Lki8m_TUWv_5tNV7HU=EOyk4+Cyww@mail.gmail.com>
References: <AANLkTinnhr2JPN2Lki8m_TUWv_5tNV7HU=EOyk4+Cyww@mail.gmail.com>
Message-ID: <AANLkTiminpQ0BS3-xwCioeDwsN_U7y_uEqzGhsmcf9Os@mail.gmail.com>

What version are you running? There was a version a couple of days ago
where the codecs were having problems negotiating different ptimes.

-Steve


2010/10/1 Juan Antonio Iba?ez Santorum <juanito1982 at gmail.com>:
> Hello,
> ?? I am trying to use wiki config about iLBC codec but I am not be able to
> make it work. If I configure codec prefs as:
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA,iLBC"/>
> I get:
> -------------------------
> 2010-10-01 08:48:22.444357 [DEBUG] sofia_glue.c:3867 Audio Codec Compare
> [iLBC:97:8000:30]/[GSM:3:8000:20]
> 2010-10-01 08:48:22.444357 [DEBUG] sofia_glue.c:3867 Audio Codec Compare
> [iLBC:97:8000:30]/[PCMU:0:8000:20]
> 2010-10-01 08:48:22.444357 [DEBUG] sofia_glue.c:3867 Audio Codec Compare
> [iLBC:97:8000:30]/[PCMA:8:8000:20]
> 2010-10-01 08:48:22.446360 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/dsp1 at 174.36.98.82) Running State Change CS_NEW
> 2010-10-01 08:48:22.446360 [DEBUG] sofia_glue.c:3867 Audio Codec Compare
> [iLBC:97:8000:30]/[iLBC:98:8000:20]
> -------------------------
> Phone offers 30 ms, FS 20 ms and call gets cursed...
> If I use:
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA,iLBC at 30i"/>
> ?or
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA,iLBC at 20i"/>
> FS does not offer iLBC codec in Audio Codec Compare
> Where could be the misscounfiguration? Which would be the best config?
> Regards
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From saeedahmad1981 at gmail.com  Fri Oct  1 03:17:04 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Fri, 1 Oct 2010 12:17:04 +0200
Subject: [Freeswitch-users] Duplicate CDRs
Message-ID: <AANLkTinra68rJ_sQcycX9wVX33tVkJcu2AbA_0QkuyZr@mail.gmail.com>

Dear List,

I am using xml_cdr to write the records on disk.

today for first time when there was a call pressure around 60-70 concurrent
calls (around 140 session), then FS started to write same CDR record two
times with different/unique file name and also different/unique UUIDs etc..
but the called number, start time, end time everything is same. so its
surely a duplicated CDR.

-> I am running FS on cent os, on a 6 core machine with 4  GM RAM.
-> FS version: FreeSWITCH Version 1.0.head (git-72baaf6 2010-09-21 19-22-28
-0500) (is anything regarding cdr changed in new version? so i can upgrade)

> Please note that system was running fine with few call attempts and i
never faced this prob, only in case of 6-70 councurrent calls

My xml_cr.conf:

<configuration name="xml_cdr.conf" description="XML CDR CURL logger">
  <settings>
    <!-- the url to post to if blank web posting is disabled  -->
    <!-- <param name="url" value="http://localhost/cdr_curl/post.php"/> -->

    <!-- optional: credentials to send to web server -->
    <!--    <param name="cred" value="user:pass"/> -->

    <!-- the total number of retries (not counting the first 'try') to post
to webserver incase of failure -->
*    <param name="retries" value="2"/>*

    <!-- delay between retries in seconds, default is 5 seconds -->
    <!-- <param name="delay" value="1"/> -->

    <!-- Log via http and on disk, default is false -->
    <!-- <param name="log-http-and-disk" value="true"/> -->

    <!-- optional: if not present we do not log every record to disk -->
    <!-- either an absolute path, a relative path assuming ${prefix}/logs or
a blank value will default to ${prefix}/logs/xml_cdr -->
*    <param name="log-dir" value="/opt/fscdr"/>*

    <!-- optional: if not present we do log the b leg -->
    <!-- true or false if we should create a cdr for the b leg of a call-->
    <!--param name="log-b-leg" value="false"/-->

    <!-- optional: if not present, all filenames are the uuid of the call
-->
    <!-- true or false if a leg files are prefixed "a_" -->
  *  <param name="prefix-a-leg" value="true"/>*

    <!-- encode the post data may be 'true' for url encoding, 'false' for no
encoding or 'base64' for base64 encoding -->
    *<param name="encode" value="true"/>*

    <!-- optional: set to true to disable Expect: 100-continue lighttpd
requires this setting -->
    <!--<param name="disable-100-continue" value="true"/>-->

    <!-- optional: full path to the error log dir for failed web posts if
not specified its the same as log-dir -->
    <!-- either an absolute path, a relative path assuming ${prefix}/logs or
a blank or omitted value will default to ${prefix}/logs/xml_cdr -->
    *<param name="err-log-dir" value="/opt/error_cdr"/>*

    <!-- which auhtentification scheme to use. Supported values are: basic,
digest, NTLM, GSS-NEGOTIATE or "any" for automatic detection -->
    <!--<param name="auth-scheme" value="basic"/>-->

    <!-- optional: this will enable the CA root certificate check by libcurl
to
         verify that the certificate was issued by a major Certificate
Authority.
         note: default value is disabled. only enable if you want this! -->
    <!--<param name="enable-cacert-check" value="true"/>-->
    <!-- optional: verify that the server is actually the one listed in the
cert -->
    <!-- <param name="enable-ssl-verifyhost" value="true"/> -->

    <!-- optional: these options can be used to specify custom SSL
certificates
         to use for HTTPS communications. Either use both options or
neither.
         Specify your public key with 'ssl-cert-path' and the private key
with
         'ssl-key-path'. If your private key has a password, specify it with
         'ssl-key-password'. -->
    <!-- <param name="ssl-cert-path"
value="$${base_dir}/conf/certs/public_key.pem"/> -->
    <!-- <param name="ssl-key-path"
value="$${base_dir}/conf/certs/private_key.pem"/> -->
    <!-- <param name="ssl-key-password" value="MyPrivateKeyPassword"/> -->

    <!-- optional: use a custom CA certificate in PEM format to verify the
peer
         with. This is useful if you are acting as your own certificate
authority.
         note: only makes sense if used in combination with
"enable-cacert-check." -->
    <!-- <param name="ssl-cacert-file"
value="$${base_dir}/conf/certs/cacert.pem"/> -->

    <!-- optional: specify the SSL version to force HTTPS to use. Valid
options are
         "SSLv3" and "TLSv1". Otherwise libcurl will auto-negotiate the
version. -->
    <!-- <param name="ssl-version" value="TLSv1"/> -->

    <!-- optional: enables cookies and stores them in the specified file.
-->
    <!-- <param name="cookie-file" value="/tmp/cookie-mod_xml_curl.txt"/>
-->
  </settings>
</configuration>


Many Thanks
- Saeed
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From ovvenkatesan at gmail.com  Fri Oct  1 03:41:49 2010
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Fri, 1 Oct 2010 16:11:49 +0530
Subject: [Freeswitch-users] Not able to connect mysql using Dbh
In-Reply-To: <AANLkTik-85W7+nwSqp5=-wg-bz_s-MCQPGE8_Ob_wPVK@mail.gmail.com>
References: <AANLkTi=cw3EMt38jJnc4uS7+MiYNimHOBSSKDPBAm1JT@mail.gmail.com>
	<AANLkTin4pHxfC1SAFF7zLki5u+GpderkNCKuxwWJdjM9@mail.gmail.com>
	<AANLkTikBYK5-NFbCF7b7PVjww2kh7Lc4_oSTGBGiHkMc@mail.gmail.com>
	<AANLkTik_syQb_Pg=acH6iogt8fpsKXphhmQc0qjcR+Xo@mail.gmail.com>
	<CD1DF16B-569E-44BA-B586-A34246D083D7@scarlet-internet.nl>
	<AANLkTimyFShcBkoAhmvcpfM=WWtO+N1jS2CQ9ObjkKHj@mail.gmail.com>
	<AANLkTik-85W7+nwSqp5=-wg-bz_s-MCQPGE8_Ob_wPVK@mail.gmail.com>
Message-ID: <AANLkTimvGGu32xVL3G90rPjALkOayV_TPgya2DPf9ReR@mail.gmail.com>

Hi Leon,

I did it. Its working now :) . Thank you very much.


Regards,
Venkat.

On Fri, Oct 1, 2010 at 11:48 AM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

>
>
> On Fri, Oct 1, 2010 at 3:05 PM, ovvenkat <ovvenkatesan at gmail.com> wrote:
>
>>
>>
>> On Thu, Sep 30, 2010 at 1:29 PM, Leon de Rooij <leon at scarlet-internet.nl>wrote:
>>
>>> Hi Venkat,
>>>
>>> Please try to get latest revision from git. The Dbh function was
>>> introduced in mod_lua since 7 sept. You don't need to install any extra
>>> packages in your os if you have odbc already working in FreeSWITCH itself.
>>>
>>>
>> Hi Leon, I have updated to latest git version.  My new version is
>>
>> FreeSWITCH Version 1.0.head (svn-18355M)
>>
>
> Clearly you don't have the latest git version. The above is from svn
> mirror, which is not being synced with git repo anymore.
> If you were using git, you would see something like this:
>
> FreeSWITCH Version 1.0.head (git-b6a81ba 2010-09-27 21-36-02 -0500)
>
>
>> Still, I am not able to  use freeswitch.Dbh. I am getting error like
>>
>> 2010-10-01 11:22:38.226735 [ERR] mod_lua.cpp:182
>> /usr/local/freeswitch/scripts/test.lua:23: attempt to call field 'Dbh' (a
>> nil value)
>>
>> stack traceback:
>>         /usr/local/freeswitch/scripts/test.lua:23: in main chunk
>>
>>
>> I can able to connect to mysql using java script, the script i have used
>>
>> use("ODBC");
>> var db = new ODBC("dsn_name", "venkat", "venkat123");
>>
>> Thanks in advance,
>> Venkat.
>>
>> Kind regards,
>>>
>>> Leon
>>>
>>>
>>>
>>>
>>> On Sep 30, 2010, at 9:47 AM, ovvenkat wrote:
>>>
>>> Hi to all,
>>>
>>> If I wanted to use "freeswitch.Dbh" to connect mysql,  Can you anyone
>>> please guide me, what are the packages I need to install ?
>>>
>>> As of now, I have installed mysql and unixodbc
>>>
>>> On Thu, Sep 30, 2010 at 11:03 AM, ovvenkat <ovvenkatesan at gmail.com>wrote:
>>>
>>>> Hi Sassan,
>>>>
>>>> Which version of FreeSWITCH?
>>>>>
>>>>>
>>>>     FreeSWITCH Version 1.0.6
>>>>
>>>>     Installed on CENT OS 5.3
>>>>
>>>>
>>>>
>>>> Regards,
>>>> Venkat.
>>>>
>>>>
>>>> Regards
>>>>> HASSAN
>>>>>
>>>>>
>>>>> On 2010-09-30, ovvenkat <ovvenkatesan at gmail.com> wrote:
>>>>> > Hi to all,
>>>>> > I could not able to connect to mySql using Dbh. Here is the coding I
>>>>> have
>>>>> > used to connect to my db.
>>>>> > *
>>>>> > local dbh = assert(freeswitch.Dbh("dsn_name","venkat","venkat123"))*
>>>>> >
>>>>> > I am getting following error while executing the lua test.lua in
>>>>> fs_cli
>>>>> >
>>>>> > 2010-09-29 17:57:29.924498 [ERR] mod_lua.cpp:182
>>>>> > /usr/local/freeswitch/scripts/test.lua:23: attempt to call field
>>>>> 'Dbh' (a
>>>>> > nil value)
>>>>> > stack traceback: /usr/local/freeswitch/scripts/test.lua:23: in main
>>>>> chunk
>>>>> >
>>>>> > */etc/odbc.ini* file contains
>>>>> >
>>>>> > [dsn_name]
>>>>> > Driver=MySQL
>>>>> > SERVER=localhost
>>>>> > PORT=3306
>>>>> > USER=venkat
>>>>> > PASSWORD=venkat123
>>>>> > DATABASE=venkat
>>>>> > Socket = /var/lib/mysql/mysql.sock
>>>>> >
>>>>> > *I can able to connect to my db using following command *
>>>>> >
>>>>> > isql dsn_name venkat venkat123
>>>>> >
>>>>> > I trying to figure it out for past two days and gone though wiki  not
>>>>> able
>>>>> > to crack it. Any one please help me to resolve my issue.
>>>>> >
>>>>> >
>>>>> > --
>>>>> >
>>>>> > Regards
>>>>> > Venkatesan OV.
>>>>> >
>>>>>
>>>>> --
>>>>> Sent from my mobile device
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> If you have come to help me, you are wasting your time.
>>>> If you have come to because your liberation is bound up in mine, we can
>>>> work together.
>>>>
>>>>
>>>> Regards
>>>> Venkatesan OV.
>>>>
>>>>
>>>
>>>
>>> --
>>>
>>> If you have come to help me, you are wasting your time.
>>> If you have come to because your liberation is bound up in mine, we can
>>> work together.
>>>
>>>
>>> Regards
>>> Venkatesan OV.
>>>
>>>  _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>>
>> If you have come to help me, you are wasting your time.
>> If you have come to because your liberation is bound up in mine, we can
>> work together.
>>
>>
>> Regards
>> Venkatesan OV.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From juanito1982 at gmail.com  Fri Oct  1 03:41:48 2010
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Fri, 1 Oct 2010 12:41:48 +0200
Subject: [Freeswitch-users] Error forcing iLBC to 30 ms
In-Reply-To: <AANLkTiminpQ0BS3-xwCioeDwsN_U7y_uEqzGhsmcf9Os@mail.gmail.com>
References: <AANLkTinnhr2JPN2Lki8m_TUWv_5tNV7HU=EOyk4+Cyww@mail.gmail.com>
	<AANLkTiminpQ0BS3-xwCioeDwsN_U7y_uEqzGhsmcf9Os@mail.gmail.com>
Message-ID: <AANLkTi=kUt=Ar3+ket=P3O7XXPSiS9mvzeu6DhnW5ikj@mail.gmail.com>

19dc1ab0f207d20a0934cd0c15b1e7757070be25 from Sep 16th

Regards

2010/10/1 Steven Ayre <steveayre at gmail.com>

> What version are you running? There was a version a couple of days ago
> where the codecs were having problems negotiating different ptimes.
>
> -Steve
>
>
> 2010/10/1 Juan Antonio Iba?ez Santorum <juanito1982 at gmail.com>:
> > Hello,
> >    I am trying to use wiki config about iLBC codec but I am not be able
> to
> > make it work. If I configure codec prefs as:
> > <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA,iLBC"/>
> > I get:
> > -------------------------
> > 2010-10-01 08:48:22.444357 [DEBUG] sofia_glue.c:3867 Audio Codec Compare
> > [iLBC:97:8000:30]/[GSM:3:8000:20]
> > 2010-10-01 08:48:22.444357 [DEBUG] sofia_glue.c:3867 Audio Codec Compare
> > [iLBC:97:8000:30]/[PCMU:0:8000:20]
> > 2010-10-01 08:48:22.444357 [DEBUG] sofia_glue.c:3867 Audio Codec Compare
> > [iLBC:97:8000:30]/[PCMA:8:8000:20]
> > 2010-10-01 08:48:22.446360 [DEBUG] switch_core_state_machine.c:314
> > (sofia/internal/dsp1 at 174.36.98.82) Running State Change CS_NEW
> > 2010-10-01 08:48:22.446360 [DEBUG] sofia_glue.c:3867 Audio Codec Compare
> > [iLBC:97:8000:30]/[iLBC:98:8000:20]
> > -------------------------
> > Phone offers 30 ms, FS 20 ms and call gets cursed...
> > If I use:
> > <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA,iLBC at 30i
> "/>
> >  or
> > <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA,iLBC at 20i
> "/>
> > FS does not offer iLBC codec in Audio Codec Compare
> > Where could be the misscounfiguration? Which would be the best config?
> > Regards
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.ponzone at ipeva.fr  Fri Oct  1 03:48:41 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 1 Oct 2010 12:48:41 +0200
Subject: [Freeswitch-users] Error forcing iLBC to 30 ms
In-Reply-To: <AANLkTi=kUt=Ar3+ket=P3O7XXPSiS9mvzeu6DhnW5ikj@mail.gmail.com>
References: <AANLkTinnhr2JPN2Lki8m_TUWv_5tNV7HU=EOyk4+Cyww@mail.gmail.com>
	<AANLkTiminpQ0BS3-xwCioeDwsN_U7y_uEqzGhsmcf9Os@mail.gmail.com>
	<AANLkTi=kUt=Ar3+ket=P3O7XXPSiS9mvzeu6DhnW5ikj@mail.gmail.com>
Message-ID: <23D83B33-41E4-4FB3-9FCC-5D343725EEAC@ipeva.fr>

Upgrade!

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
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Le 01/10/2010 ? 12:41, Juan Antonio Iba?ez Santorum a ?crit :

> 19dc1ab0f207d20a0934cd0c15b1e7757070be25 from Sep 16th
>
> Regards
>
> 2010/10/1 Steven Ayre <steveayre at gmail.com>
> What version are you running? There was a version a couple of days ago
> where the codecs were having problems negotiating different ptimes.
>
> -Steve
>
>
> 2010/10/1 Juan Antonio Iba?ez Santorum <juanito1982 at gmail.com>:
> > Hello,
> >    I am trying to use wiki config about iLBC codec but I am not be  
> able to
> > make it work. If I configure codec prefs as:
> > <X-PRE-PROCESS cmd="set"  
> data="global_codec_prefs=GSM,PCMU,PCMA,iLBC"/>
> > I get:
> > -------------------------
> > 2010-10-01 08:48:22.444357 [DEBUG] sofia_glue.c:3867 Audio Codec  
> Compare
> > [iLBC:97:8000:30]/[GSM:3:8000:20]
> > 2010-10-01 08:48:22.444357 [DEBUG] sofia_glue.c:3867 Audio Codec  
> Compare
> > [iLBC:97:8000:30]/[PCMU:0:8000:20]
> > 2010-10-01 08:48:22.444357 [DEBUG] sofia_glue.c:3867 Audio Codec  
> Compare
> > [iLBC:97:8000:30]/[PCMA:8:8000:20]
> > 2010-10-01 08:48:22.446360 [DEBUG] switch_core_state_machine.c:314
> > (sofia/internal/dsp1 at 174.36.98.82) Running State Change CS_NEW
> > 2010-10-01 08:48:22.446360 [DEBUG] sofia_glue.c:3867 Audio Codec  
> Compare
> > [iLBC:97:8000:30]/[iLBC:98:8000:20]
> > -------------------------
> > Phone offers 30 ms, FS 20 ms and call gets cursed...
> > If I use:
> > <X-PRE-PROCESS cmd="set"  
> data="global_codec_prefs=GSM,PCMU,PCMA,iLBC at 30i"/>
> >  or
> > <X-PRE-PROCESS cmd="set"  
> data="global_codec_prefs=GSM,PCMU,PCMA,iLBC at 20i"/>
> > FS does not offer iLBC codec in Audio Codec Compare
> > Where could be the misscounfiguration? Which would be the best  
> config?
> > Regards
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at ipeva.fr  Fri Oct  1 03:48:41 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 1 Oct 2010 12:48:41 +0200
Subject: [Freeswitch-users] Error forcing iLBC to 30 ms
In-Reply-To: <AANLkTi=kUt=Ar3+ket=P3O7XXPSiS9mvzeu6DhnW5ikj@mail.gmail.com>
References: <AANLkTinnhr2JPN2Lki8m_TUWv_5tNV7HU=EOyk4+Cyww@mail.gmail.com>
	<AANLkTiminpQ0BS3-xwCioeDwsN_U7y_uEqzGhsmcf9Os@mail.gmail.com>
	<AANLkTi=kUt=Ar3+ket=P3O7XXPSiS9mvzeu6DhnW5ikj@mail.gmail.com>
Message-ID: <23D83B33-41E4-4FB3-9FCC-5D343725EEAC@ipeva.fr>

Upgrade!

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 01/10/2010 ? 12:41, Juan Antonio Iba?ez Santorum a ?crit :

> 19dc1ab0f207d20a0934cd0c15b1e7757070be25 from Sep 16th
>
> Regards
>
> 2010/10/1 Steven Ayre <steveayre at gmail.com>
> What version are you running? There was a version a couple of days ago
> where the codecs were having problems negotiating different ptimes.
>
> -Steve
>
>
> 2010/10/1 Juan Antonio Iba?ez Santorum <juanito1982 at gmail.com>:
> > Hello,
> >    I am trying to use wiki config about iLBC codec but I am not be  
> able to
> > make it work. If I configure codec prefs as:
> > <X-PRE-PROCESS cmd="set"  
> data="global_codec_prefs=GSM,PCMU,PCMA,iLBC"/>
> > I get:
> > -------------------------
> > 2010-10-01 08:48:22.444357 [DEBUG] sofia_glue.c:3867 Audio Codec  
> Compare
> > [iLBC:97:8000:30]/[GSM:3:8000:20]
> > 2010-10-01 08:48:22.444357 [DEBUG] sofia_glue.c:3867 Audio Codec  
> Compare
> > [iLBC:97:8000:30]/[PCMU:0:8000:20]
> > 2010-10-01 08:48:22.444357 [DEBUG] sofia_glue.c:3867 Audio Codec  
> Compare
> > [iLBC:97:8000:30]/[PCMA:8:8000:20]
> > 2010-10-01 08:48:22.446360 [DEBUG] switch_core_state_machine.c:314
> > (sofia/internal/dsp1 at 174.36.98.82) Running State Change CS_NEW
> > 2010-10-01 08:48:22.446360 [DEBUG] sofia_glue.c:3867 Audio Codec  
> Compare
> > [iLBC:97:8000:30]/[iLBC:98:8000:20]
> > -------------------------
> > Phone offers 30 ms, FS 20 ms and call gets cursed...
> > If I use:
> > <X-PRE-PROCESS cmd="set"  
> data="global_codec_prefs=GSM,PCMU,PCMA,iLBC at 30i"/>
> >  or
> > <X-PRE-PROCESS cmd="set"  
> data="global_codec_prefs=GSM,PCMU,PCMA,iLBC at 20i"/>
> > FS does not offer iLBC codec in Audio Codec Compare
> > Where could be the misscounfiguration? Which would be the best  
> config?
> > Regards
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From dujinfang at gmail.com  Fri Oct  1 04:03:28 2010
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 1 Oct 2010 19:03:28 +0800
Subject: [Freeswitch-users] ip-change detection
In-Reply-To: <AANLkTi=An5iCE=yNeuJFZRanp=J3bCNh9YdiiOB=TsPS@mail.gmail.com>
References: <AANLkTi=An5iCE=yNeuJFZRanp=J3bCNh9YdiiOB=TsPS@mail.gmail.com>
Message-ID: <AANLkTi=npftVaroFi3oH2c860yLBwGma-oULZ0Rt9zo-@mail.gmail.com>

sofia.conf.xml:    <!-- <param name="auto-restart" value="false"/> -->

On Fri, Oct 1, 2010 at 4:24 PM, Jonas Gauffin <jonas.gauffin at gmail.com> wrote:
> Hello,
> How do I turn off ip-change detection?
> Regards,
> ??Jonas
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From jonas.gauffin at gmail.com  Fri Oct  1 04:30:34 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Fri, 1 Oct 2010 13:30:34 +0200
Subject: [Freeswitch-users] ip-change detection
In-Reply-To: <AANLkTi=npftVaroFi3oH2c860yLBwGma-oULZ0Rt9zo-@mail.gmail.com>
References: <AANLkTi=An5iCE=yNeuJFZRanp=J3bCNh9YdiiOB=TsPS@mail.gmail.com>
	<AANLkTi=npftVaroFi3oH2c860yLBwGma-oULZ0Rt9zo-@mail.gmail.com>
Message-ID: <AANLkTinMx2RaS-5PPZqsDBoTRj_Nxy3tvPNeDVO0+YYd@mail.gmail.com>

thanks

On Fri, Oct 1, 2010 at 1:03 PM, Seven Du <dujinfang at gmail.com> wrote:

> sofia.conf.xml:    <!-- <param name="auto-restart" value="false"/> -->
>
> On Fri, Oct 1, 2010 at 4:24 PM, Jonas Gauffin <jonas.gauffin at gmail.com>
> wrote:
> > Hello,
> > How do I turn off ip-change detection?
> > Regards,
> >   Jonas
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ovvenkatesan at gmail.com  Fri Oct  1 05:31:30 2010
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Fri, 1 Oct 2010 18:01:30 +0530
Subject: [Freeswitch-users] PRI inbound problem after updating freeswitch to
	current git version
Message-ID: <AANLkTikXsP9jvw7AQFQCkKHgQASLvkUexa1W=hn9-pMo@mail.gmail.com>

Hi to all,

I have upgraded my freeSwitch to current git version.

I am using below dial plan to handle the inbound call (SangomaA101 AND
openzap)

      <extension name="start_Talk">
      <condition field="destination_number" expression="^1100$">
        <action application="sleep" data="2000"/>
            <action application="ivr" data="demo_ivr"/>
      </condition>
      </extension>

When I am calling the PRI number, my phone is never answered, its keep on
ringing. But, fs_cli logs shows that, demo_ivr menu is playing.

here is my  fs_cli output,

http://pastebin.freeswitch.org/14098

Its happening after upgrading the freeswitch to git version. Any one plz
help me what may cause this problem?

Thanks.
-- 

Regards
Venkatesan OV.
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From ovvenkatesan at gmail.com  Fri Oct  1 05:42:56 2010
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Fri, 1 Oct 2010 18:12:56 +0530
Subject: [Freeswitch-users] PRI inbound problem after updating
	freeswitch to current git version
In-Reply-To: <AANLkTikXsP9jvw7AQFQCkKHgQASLvkUexa1W=hn9-pMo@mail.gmail.com>
References: <AANLkTikXsP9jvw7AQFQCkKHgQASLvkUexa1W=hn9-pMo@mail.gmail.com>
Message-ID: <AANLkTim_3cd=vvBLFXHCtd65tYR8P02q_TEtAX3j_D1g@mail.gmail.com>

Sorry, it was my mistake.

<action application="answer"/>

i did not answer the phone in my dial plan.





On Fri, Oct 1, 2010 at 6:01 PM, ovvenkat <ovvenkatesan at gmail.com> wrote:

> Hi to all,
>
> I have upgraded my freeSwitch to current git version.
>
> I am using below dial plan to handle the inbound call (SangomaA101 AND
> openzap)
>
>       <extension name="start_Talk">
>       <condition field="destination_number" expression="^1100$">
>         <action application="sleep" data="2000"/>
>             <action application="ivr" data="demo_ivr"/>
>       </condition>
>       </extension>
>
> When I am calling the PRI number, my phone is never answered, its keep on
> ringing. But, fs_cli logs shows that, demo_ivr menu is playing.
>
> here is my  fs_cli output,
>
> http://pastebin.freeswitch.org/14098
>
> Its happening after upgrading the freeswitch to git version. Any one plz
> help me what may cause this problem?
>
> Thanks.
> --
>
> Regards
> Venkatesan OV.
>
>


-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From brian at freeswitch.org  Fri Oct  1 06:56:16 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 1 Oct 2010 08:56:16 -0500
Subject: [Freeswitch-users] License of libzrtp for usage with FreeSWITCH?
In-Reply-To: <4CA53F77.5040106@puzzled.xs4all.nl>
References: <4CA53F77.5040106@puzzled.xs4all.nl>
Message-ID: <C41DB9FA-0019-40EB-9CDF-0F3C3BA6276C@freeswitch.org>

Patrick
	Since we do not ship with the lib its up to the end user to combine them.  So the license concern is not an issue for the FreeSWITCH project.  Its now on you to comply with all the licensing.

Thanks,
Brian

On Sep 30, 2010, at 8:55 PM, Patrick Lists wrote:

> Hi,
> 
> Does anyone know what the license is of libzrtp for usage with 
> FreeSWITCH? In the FreeSWITCH source I see a file docs/zrtp_agpl-3.0.txt 
> but the libzrtp source itself is not included.
> 
> So I got libzrtp-0.81.514 and in the source there is a file zrtp_legal.c 
> that says:
> 
>  * libZRTP SDK library, implements the ZRTP secure VoIP protocol.
>  * Copyright (c) 2006-2009 Philip R. Zimmermann.  All rights reserved.
>  *
>  * The version of the SDK library that is packaged with this legal 
> notice is
>  * NOT licensed under the GPL, AGPL, LGPL, or any other open source 
> license.
>  * For licensing terms or other information,
>  * contact Philip R. Zimmermann <prz at mit.edu>.
>  * For more contact information, see http://philzimmermann.com
>  *
>  * This file must be packaged together with the rest of the libZRTP SDK
>  * source code.  That's why it's in a .c file.
>  *
>  * This software may be subject to export controls by the US Commerce
>  * Department's Bureau of Industry and Security.  This software is provided
>  * "as is," with no warranty expressed or implied.
> 
> This is quite the opposite of what zrtp_agpl-3.0.txt suggests in the 
> FreeSWITCH source.
> 
> Any ideas?
> 
> Regards,
> Patrick
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Fri Oct  1 06:57:37 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 1 Oct 2010 08:57:37 -0500
Subject: [Freeswitch-users] serious problem in record_session
In-Reply-To: <AANLkTin0E5m-y6WcX5VE32yABBUfiuc7gYWNDSYpJQ=u@mail.gmail.com>
References: <14381.1285250251@ccs.covici.com>
	<4B839ACC-C9C3-4A7B-9D0C-6E1C44E433A1@freeswitch.org>
	<15271.1285252472@ccs.covici.com>
	<AANLkTikZsn+btek1f2Y-bfgUb=b9Te1La4ieKaMP6z7e@mail.gmail.com>
	<15773.1285254902@ccs.covici.com> <19074.1285456498@ccs.covici.com>
	<C3BDF5BAA4FC4D62849FD538BF7CBBBC@morindads>
	<41BCD9C9-B1BD-4EF1-8412-3C2D972C68D8@freeswitch.org>
	<59D9C73963DF441BB78ECAB357D6363F@morindads>
	<AANLkTin0E5m-y6WcX5VE32yABBUfiuc7gYWNDSYpJQ=u@mail.gmail.com>
Message-ID: <7B6107E5-7FA6-49C8-B269-21D2702751BB@freeswitch.org>

Yes its 100% solved since the conference works in raw audio anyway so no codec would be required to record the conference.

/b

On Sep 30, 2010, at 1:43 PM, Michael Collins wrote:

> LIke bkw and anthm said, at fs_cli:
> 
> conference <conf_name> record <file_name>
> 
> Problem solved, no?
> 
> -MC



From sos at sokhapkin.dyndns.org  Fri Oct  1 07:04:57 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 1 Oct 2010 10:04:57 -0400
Subject: [Freeswitch-users] Call Rejected Issue
In-Reply-To: <AANLkTikBSri2YDAY8FBVR4oiP5qVQcN_KeqwCpCVCdPH@mail.gmail.com>
References: <AANLkTikBSri2YDAY8FBVR4oiP5qVQcN_KeqwCpCVCdPH@mail.gmail.com>
Message-ID: <201010011004.57765.sos@sokhapkin.dyndns.org>

Enable SIP trace to see SIP messages exchanged.

On Friday 01 October 2010, Tech Man wrote:
> I'm trying to configure an iptel SIP account but everytime I call an
>  external number, it gets rejected.  I'm running FreeSwitch 1.0.6 on a Mac
>  OS X 10.5.8.  Any help would be greatly appreciated.  Thanks
> 
> Sam
> 



From brian at freeswitch.org  Fri Oct  1 07:11:11 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 1 Oct 2010 09:11:11 -0500
Subject: [Freeswitch-users] Call Rejected Issue
In-Reply-To: <AANLkTikBSri2YDAY8FBVR4oiP5qVQcN_KeqwCpCVCdPH@mail.gmail.com>
References: <AANLkTikBSri2YDAY8FBVR4oiP5qVQcN_KeqwCpCVCdPH@mail.gmail.com>
Message-ID: <69B2390F-D94F-4490-B4B4-B461189E3FA8@freeswitch.org>

The far side rejected the call turn on the sip trace and watch it... It'll be rather clear what is going on if you run "sofia profile external siptrace on"

/b

On Oct 1, 2010, at 1:12 AM, Tech Man wrote:

> I'm trying to configure an iptel SIP account but everytime I call an external number, it gets rejected.  I'm running FreeSwitch 1.0.6 on a Mac OS X 10.5.8.  Any help would be greatly appreciated.  Thanks
> 
> Sam



From jerry.richards at teotech.com  Fri Oct  1 08:22:29 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 1 Oct 2010 08:22:29 -0700
Subject: [Freeswitch-users] Intercept Ringing Call
Message-ID: <009801cb617c$763d7810$62b86830$@teotech.com>

Can I intercept a call in ringing state only?  The default **<number>
feature can steal a call from someone if multiple people try to answer a
ringing call.

Thanks,
Jerry



From covici at ccs.covici.com  Fri Oct  1 08:29:51 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 01 Oct 2010 11:29:51 -0400
Subject: [Freeswitch-users] voicemail email not working.
Message-ID: <15521.1285946991@ccs.covici.com>

Hi.  I left a test message and instead of using the variable I have in
my user directory, it tried to deliver to username at domain name which is
invalid -- so why did it not use the variable voicemail_email I have in
the user directory and it completely ignored the voicemail_notify_email
I have as well?

Any assistance would be appreciated.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From lists at infosecurity.ch  Fri Oct  1 08:32:08 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Fri, 01 Oct 2010 17:32:08 +0200
Subject: [Freeswitch-users] License of libzrtp for usage with FreeSWITCH?
In-Reply-To: <C41DB9FA-0019-40EB-9CDF-0F3C3BA6276C@freeswitch.org>
References: <4CA53F77.5040106@puzzled.xs4all.nl>
	<C41DB9FA-0019-40EB-9CDF-0F3C3BA6276C@freeswitch.org>
Message-ID: <4CA5FEF8.5040102@infosecurity.ch>

On http://zfoneproject.com/lic_policy.html it's written that the libzrtp
is licensed under AGPL and only zfone it's not under AGPL.

It sounds just like a typo / not updated statement in the code as the
website refer to AGPL licensed.

Fabio

On 01/10/10 15.56, Brian West wrote:
> Patrick
> 	Since we do not ship with the lib its up to the end user to combine them.  So the license concern is not an issue for the FreeSWITCH project.  Its now on you to comply with all the licensing.
>
> Thanks,
> Brian
>   


From neil.burgess at redmatter.com  Fri Oct  1 08:32:36 2010
From: neil.burgess at redmatter.com (Neil Burgess)
Date: Fri, 1 Oct 2010 16:32:36 +0100
Subject: [Freeswitch-users] Multiple "voices" under mod_say_en
Message-ID: <787302A89ACCE24DA8F56DA101E77C842B473C0A1B@THHS2E12BE1X.hostedservice2.net>

Hi,

I'd like to be able to use two sets of language files for US and UK within the say module, allowing either to be utilised with the spell, items, etc capabilities, based on setting some channel variable!

At the moment, I can't see how to achieve this since it appears that the "say_en" module for manipulating these strings is linked to the language name attribute as below.

<language name="en" sound-path="$${base_dir}/sounds/en/us/callie" tts-engine="unimrcp:loquendo7-mrcp2" tts-voice="kate">

Is there any way of supporting multiple "voices" under the same mod_say_en

Thanks,
Neil
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From fdelawarde at wirelessmundi.com  Fri Oct  1 08:34:21 2010
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Fri, 01 Oct 2010 17:34:21 +0200
Subject: [Freeswitch-users] Intercept Ringing Call
In-Reply-To: <009801cb617c$763d7810$62b86830$@teotech.com>
References: <009801cb617c$763d7810$62b86830$@teotech.com>
Message-ID: <1285947261.15336.55.camel@luna.tc.commsmundi.com>

You could use db functions to store possible pickup entries when calls
start, and remove the db entry with an "on_answer" hook.

Fran?ois.

On Fri, 2010-10-01 at 08:22 -0700, Jerry Richards wrote:
> Can I intercept a call in ringing state only?  The default **<number>
> feature can steal a call from someone if multiple people try to answer a
> ringing call.
> 
> Thanks,
> Jerry
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From steveayre at gmail.com  Fri Oct  1 08:45:42 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 1 Oct 2010 16:45:42 +0100
Subject: [Freeswitch-users] Duplicate CDRs
In-Reply-To: <AANLkTinra68rJ_sQcycX9wVX33tVkJcu2AbA_0QkuyZr@mail.gmail.com>
References: <AANLkTinra68rJ_sQcycX9wVX33tVkJcu2AbA_0QkuyZr@mail.gmail.com>
Message-ID: <AANLkTik4FikbJjQfSoQiJJze8XFLJYSVPAA0TUH6i7td@mail.gmail.com>

Are they A and B leg CDRs? For a bridged call you'll see a CDR for
both the A and B leg, which as separate channels will each have its
own UUID but will share the same caller id, destination number and
call duration.

<!--param name="log-b-leg" value="false"/-->
That parameter would disable logging the B-leg CDRs. You have it
commented out. I can't remember the default setting but from what you
describe, default behaviour might be to log both.

Look at the Direction that is submitted to tell the difference between
A and B legs. It'll either be inbound (A) or outbound (B). You could
then ignore B-legs in your PHP script, or handle them differently.

-Steve


On 1 October 2010 11:17, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> Dear List,
>
> I am using xml_cdr to write the records on disk.
>
> today for first time when there was a call pressure around 60-70 concurrent
> calls (around 140 session), then FS started to write same CDR record two
> times with different/unique file name and also different/unique UUIDs etc..
> but the called number, start time, end time everything is same. so its
> surely a duplicated CDR.
>
> -> I am running FS on cent os, on a 6 core machine with 4? GM RAM.
> -> FS version: FreeSWITCH Version 1.0.head (git-72baaf6 2010-09-21 19-22-28
> -0500) (is anything regarding cdr changed in new version? so i can upgrade)
>
>> Please note that system was running fine with few call attempts and i
>> never faced this prob, only in case of 6-70 councurrent calls
>
> My xml_cr.conf:
>
> <configuration name="xml_cdr.conf" description="XML CDR CURL logger">
> ? <settings>
> ??? <!-- the url to post to if blank web posting is disabled? -->
> ??? <!-- <param name="url" value="http://localhost/cdr_curl/post.php"/> -->
>
> ??? <!-- optional: credentials to send to web server -->
> ??? <!--??? <param name="cred" value="user:pass"/> -->
>
> ??? <!-- the total number of retries (not counting the first 'try') to post
> to webserver incase of failure -->
> ??? <param name="retries" value="2"/>
>
> ??? <!-- delay between retries in seconds, default is 5 seconds -->
> ??? <!-- <param name="delay" value="1"/> -->
>
> ??? <!-- Log via http and on disk, default is false -->
> ??? <!-- <param name="log-http-and-disk" value="true"/> -->
>
> ??? <!-- optional: if not present we do not log every record to disk -->
> ??? <!-- either an absolute path, a relative path assuming ${prefix}/logs or
> a blank value will default to ${prefix}/logs/xml_cdr -->
> ??? <param name="log-dir" value="/opt/fscdr"/>
>
> ??? <!-- optional: if not present we do log the b leg -->
> ??? <!-- true or false if we should create a cdr for the b leg of a call-->
> ??? <!--param name="log-b-leg" value="false"/-->
>
> ??? <!-- optional: if not present, all filenames are the uuid of the call
> -->
> ??? <!-- true or false if a leg files are prefixed "a_" -->
> ??? <param name="prefix-a-leg" value="true"/>
>
> ??? <!-- encode the post data may be 'true' for url encoding, 'false' for no
> encoding or 'base64' for base64 encoding -->
> ??? <param name="encode" value="true"/>
>
> ??? <!-- optional: set to true to disable Expect: 100-continue lighttpd
> requires this setting -->
> ??? <!--<param name="disable-100-continue" value="true"/>-->
>
> ??? <!-- optional: full path to the error log dir for failed web posts if
> not specified its the same as log-dir -->
> ??? <!-- either an absolute path, a relative path assuming ${prefix}/logs or
> a blank or omitted value will default to ${prefix}/logs/xml_cdr -->
> ??? <param name="err-log-dir" value="/opt/error_cdr"/>
>
> ??? <!-- which auhtentification scheme to use. Supported values are: basic,
> digest, NTLM, GSS-NEGOTIATE or "any" for automatic detection -->
> ??? <!--<param name="auth-scheme" value="basic"/>-->
>
> ??? <!-- optional: this will enable the CA root certificate check by libcurl
> to
> ???????? verify that the certificate was issued by a major Certificate
> Authority.
> ???????? note: default value is disabled. only enable if you want this! -->
> ??? <!--<param name="enable-cacert-check" value="true"/>-->
> ??? <!-- optional: verify that the server is actually the one listed in the
> cert -->
> ??? <!-- <param name="enable-ssl-verifyhost" value="true"/> -->
>
> ??? <!-- optional: these options can be used to specify custom SSL
> certificates
> ???????? to use for HTTPS communications. Either use both options or
> neither.
> ???????? Specify your public key with 'ssl-cert-path' and the private key
> with
> ???????? 'ssl-key-path'. If your private key has a password, specify it with
> ???????? 'ssl-key-password'. -->
> ??? <!-- <param name="ssl-cert-path"
> value="$${base_dir}/conf/certs/public_key.pem"/> -->
> ??? <!-- <param name="ssl-key-path"
> value="$${base_dir}/conf/certs/private_key.pem"/> -->
> ??? <!-- <param name="ssl-key-password" value="MyPrivateKeyPassword"/> -->
>
> ??? <!-- optional: use a custom CA certificate in PEM format to verify the
> peer
> ???????? with. This is useful if you are acting as your own certificate
> authority.
> ???????? note: only makes sense if used in combination with
> "enable-cacert-check." -->
> ??? <!-- <param name="ssl-cacert-file"
> value="$${base_dir}/conf/certs/cacert.pem"/> -->
>
> ??? <!-- optional: specify the SSL version to force HTTPS to use. Valid
> options are
> ???????? "SSLv3" and "TLSv1". Otherwise libcurl will auto-negotiate the
> version. -->
> ??? <!-- <param name="ssl-version" value="TLSv1"/> -->
>
> ??? <!-- optional: enables cookies and stores them in the specified file.
> -->
> ??? <!-- <param name="cookie-file" value="/tmp/cookie-mod_xml_curl.txt"/>
> -->
> ? </settings>
> </configuration>
>
>
> Many Thanks
> - Saeed
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From david.ponzone at ipeva.fr  Fri Oct  1 09:08:23 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 1 Oct 2010 18:08:23 +0200
Subject: [Freeswitch-users] Duplicate CDRs
In-Reply-To: <AANLkTik4FikbJjQfSoQiJJze8XFLJYSVPAA0TUH6i7td@mail.gmail.com>
References: <AANLkTinra68rJ_sQcycX9wVX33tVkJcu2AbA_0QkuyZr@mail.gmail.com>
	<AANLkTik4FikbJjQfSoQiJJze8XFLJYSVPAA0TUH6i7td@mail.gmail.com>
Message-ID: <0ACA1E73-98AC-4E83-8B4F-16FF1B4EA672@ipeva.fr>

Steven,

I confirm the default is log B-leg:

     <!-- optional: if not present we do log the b leg -->

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 01/10/2010 ? 17:45, Steven Ayre a ?crit :

> Are they A and B leg CDRs? For a bridged call you'll see a CDR for
> both the A and B leg, which as separate channels will each have its
> own UUID but will share the same caller id, destination number and
> call duration.
>
> <!--param name="log-b-leg" value="false"/-->
> That parameter would disable logging the B-leg CDRs. You have it
> commented out. I can't remember the default setting but from what you
> describe, default behaviour might be to log both.
>
> Look at the Direction that is submitted to tell the difference between
> A and B legs. It'll either be inbound (A) or outbound (B). You could
> then ignore B-legs in your PHP script, or handle them differently.
>
> -Steve
>
>
> On 1 October 2010 11:17, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
>> Dear List,
>>
>> I am using xml_cdr to write the records on disk.
>>
>> today for first time when there was a call pressure around 60-70  
>> concurrent
>> calls (around 140 session), then FS started to write same CDR  
>> record two
>> times with different/unique file name and also different/unique  
>> UUIDs etc..
>> but the called number, start time, end time everything is same. so  
>> its
>> surely a duplicated CDR.
>>
>> -> I am running FS on cent os, on a 6 core machine with 4  GM RAM.
>> -> FS version: FreeSWITCH Version 1.0.head (git-72baaf6 2010-09-21  
>> 19-22-28
>> -0500) (is anything regarding cdr changed in new version? so i can  
>> upgrade)
>>
>>> Please note that system was running fine with few call attempts  
>>> and i
>>> never faced this prob, only in case of 6-70 councurrent calls
>>
>> My xml_cr.conf:
>>
>> <configuration name="xml_cdr.conf" description="XML CDR CURL logger">
>>   <settings>
>>     <!-- the url to post to if blank web posting is disabled  -->
>>     <!-- <param name="url" value="http://localhost/cdr_curl/ 
>> post.php"/> -->
>>
>>     <!-- optional: credentials to send to web server -->
>>     <!--    <param name="cred" value="user:pass"/> -->
>>
>>     <!-- the total number of retries (not counting the first 'try')  
>> to post
>> to webserver incase of failure -->
>>     <param name="retries" value="2"/>
>>
>>     <!-- delay between retries in seconds, default is 5 seconds -->
>>     <!-- <param name="delay" value="1"/> -->
>>
>>     <!-- Log via http and on disk, default is false -->
>>     <!-- <param name="log-http-and-disk" value="true"/> -->
>>
>>     <!-- optional: if not present we do not log every record to  
>> disk -->
>>     <!-- either an absolute path, a relative path assuming $ 
>> {prefix}/logs or
>> a blank value will default to ${prefix}/logs/xml_cdr -->
>>     <param name="log-dir" value="/opt/fscdr"/>
>>
>>     <!-- optional: if not present we do log the b leg -->
>>     <!-- true or false if we should create a cdr for the b leg of a  
>> call-->
>>     <!--param name="log-b-leg" value="false"/-->
>>
>>     <!-- optional: if not present, all filenames are the uuid of  
>> the call
>> -->
>>     <!-- true or false if a leg files are prefixed "a_" -->
>>     <param name="prefix-a-leg" value="true"/>
>>
>>     <!-- encode the post data may be 'true' for url encoding,  
>> 'false' for no
>> encoding or 'base64' for base64 encoding -->
>>     <param name="encode" value="true"/>
>>
>>     <!-- optional: set to true to disable Expect: 100-continue  
>> lighttpd
>> requires this setting -->
>>     <!--<param name="disable-100-continue" value="true"/>-->
>>
>>     <!-- optional: full path to the error log dir for failed web  
>> posts if
>> not specified its the same as log-dir -->
>>     <!-- either an absolute path, a relative path assuming $ 
>> {prefix}/logs or
>> a blank or omitted value will default to ${prefix}/logs/xml_cdr -->
>>     <param name="err-log-dir" value="/opt/error_cdr"/>
>>
>>     <!-- which auhtentification scheme to use. Supported values  
>> are: basic,
>> digest, NTLM, GSS-NEGOTIATE or "any" for automatic detection -->
>>     <!--<param name="auth-scheme" value="basic"/>-->
>>
>>     <!-- optional: this will enable the CA root certificate check  
>> by libcurl
>> to
>>          verify that the certificate was issued by a major  
>> Certificate
>> Authority.
>>          note: default value is disabled. only enable if you want  
>> this! -->
>>     <!--<param name="enable-cacert-check" value="true"/>-->
>>     <!-- optional: verify that the server is actually the one  
>> listed in the
>> cert -->
>>     <!-- <param name="enable-ssl-verifyhost" value="true"/> -->
>>
>>     <!-- optional: these options can be used to specify custom SSL
>> certificates
>>          to use for HTTPS communications. Either use both options or
>> neither.
>>          Specify your public key with 'ssl-cert-path' and the  
>> private key
>> with
>>          'ssl-key-path'. If your private key has a password,  
>> specify it with
>>          'ssl-key-password'. -->
>>     <!-- <param name="ssl-cert-path"
>> value="$${base_dir}/conf/certs/public_key.pem"/> -->
>>     <!-- <param name="ssl-key-path"
>> value="$${base_dir}/conf/certs/private_key.pem"/> -->
>>     <!-- <param name="ssl-key-password"  
>> value="MyPrivateKeyPassword"/> -->
>>
>>     <!-- optional: use a custom CA certificate in PEM format to  
>> verify the
>> peer
>>          with. This is useful if you are acting as your own  
>> certificate
>> authority.
>>          note: only makes sense if used in combination with
>> "enable-cacert-check." -->
>>     <!-- <param name="ssl-cacert-file"
>> value="$${base_dir}/conf/certs/cacert.pem"/> -->
>>
>>     <!-- optional: specify the SSL version to force HTTPS to use.  
>> Valid
>> options are
>>          "SSLv3" and "TLSv1". Otherwise libcurl will auto-negotiate  
>> the
>> version. -->
>>     <!-- <param name="ssl-version" value="TLSv1"/> -->
>>
>>     <!-- optional: enables cookies and stores them in the specified  
>> file.
>> -->
>>     <!-- <param name="cookie-file" value="/tmp/cookie- 
>> mod_xml_curl.txt"/>
>> -->
>>   </settings>
>> </configuration>
>>
>>
>> Many Thanks
>> - Saeed
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From peder at networkoblivion.com  Fri Oct  1 09:12:06 2010
From: peder at networkoblivion.com (Peder)
Date: Fri, 1 Oct 2010 11:12:06 -0500
Subject: [Freeswitch-users] voicemail email not working.
In-Reply-To: <15521.1285946991@ccs.covici.com>
References: <15521.1285946991@ccs.covici.com>
Message-ID: <00d301cb6183$64a84330$2df8c990$@com>

My guess would be that you are using the wrong variable names.  From the
wiki:

http://wiki.freeswitch.org/wiki/Mod_voicemail


"vm-notify-mailto"
The address you want notification messages sent to default: same as
vm-mailto
This is defined in the directory for the particular user as a param.

"vm-mailto"
This is the user's email address default: undefined (originally called
email-addr. Use vm-mailto instead)
This is defined in the directory for the particular user as a param.
Multiple email addresses can be defined using a comma-separated list.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
covici at ccs.covici.com
Sent: Friday, October 01, 2010 10:30 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] voicemail email not working.

Hi.  I left a test message and instead of using the variable I have in
my user directory, it tried to deliver to username at domain name which is
invalid -- so why did it not use the variable voicemail_email I have in
the user directory and it completely ignored the voicemail_notify_email
I have as well?

Any assistance would be appreciated.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com

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



From vetali100 at gmail.com  Fri Oct  1 10:00:08 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Fri, 1 Oct 2010 20:00:08 +0300
Subject: [Freeswitch-users] Simultaneous ring of all phones registered with
 the same extension number
Message-ID: <AANLkTimRmtA=e4C3tVAHf_+pTu3GvWKU7eJ6_eFq3j50@mail.gmail.com>

Hi,
Could not find this answer, so would appreciate any hints.

I have registered 2 sip phones using the *same *extension number.

When I dial this extension number, only one sip phone rings.

However I need both of them to start ringing.

Is it possible, that when I dial the number - both other phones will ring,
not only one..?
And when I answer on a phone, another one would stop ringing.


Thank you,
Vitalie
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From msc at freeswitch.org  Fri Oct  1 10:04:44 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Oct 2010 10:04:44 -0700
Subject: [Freeswitch-users] my info as appearing in conference
In-Reply-To: <31987.1285897148@ccs.covici.com>
References: <28922.1285889502@ccs.covici.com>
	<AANLkTik7cq0s_H=ELuLXd-ifBib683mzGOj7X0AwSi8u@mail.gmail.com>
	<29684.1285894407@ccs.covici.com>
	<AANLkTi=bvTKjqrP_bztBSrZavPXfTn8=_0SGhhzOPPug@mail.gmail.com>
	<31987.1285897148@ccs.covici.com>
Message-ID: <AANLkTikYMSsung_ECbCZhNex5r67Xd4oCDOO9sXxow61@mail.gmail.com>

What, exactly, are you doing to establish the call?
-MC

On Thu, Sep 30, 2010 at 6:39 PM, <covici at ccs.covici.com> wrote:

> Well, in mod_portaudio they are set to the variables in vars.xml and if
> I call my cell phone it knows the callerid number and if I go into an
> asterisk conference which I can display I see both my number and name,
> so I am confused as to where the 0's are coming from.  I am using
> extension 9888 to get into the conference maybe no callerid is set
> there, I will check that, but it still seems strange.
>
> Michael Collins <msc at freeswitch.org> wrote:
>
> > Default dialplan? If not, pastebin your dp. At some point you need to set
> > the caller id name/number channel variables. Authenticated calls will
> have
> > those populated if you define them in the user directory, otherwise you
> will
> > need to manually set them in the dialplan, or at the very least modify
> the
> > default dialplan to have something more meaningful than all zeroes.
> >
> > -MC
> >
> > On Thu, Sep 30, 2010 at 5:53 PM, <covici at ccs.covici.com> wrote:
> >
> > > Yes they are in vars.xml -- that is why I am writing.
> > >
> > > William Suffill <william.suffill at gmail.com> wrote:
> > >
> > > > vars.xml should have the default callerid details used.
> > > >
> > > > ----------------------------------------------------
> > > > Alternatives:
> > > >
> > > > ----------------------------------------------------
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > http://www.freeswitch.org
> > >
> > > --
> > > Your life is like a penny.  You're going to lose it.  The question is:
> > > How do
> > > you spend it?
> > >
> > >         John Covici
> > >         covici at ccs.covici.com
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> >
> > ----------------------------------------------------
> > Alternatives:
> >
> > ----------------------------------------------------
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From saeedahmad1981 at gmail.com  Fri Oct  1 10:12:08 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Fri, 1 Oct 2010 19:12:08 +0200
Subject: [Freeswitch-users] Duplicate CDRs
In-Reply-To: <0ACA1E73-98AC-4E83-8B4F-16FF1B4EA672@ipeva.fr>
References: <AANLkTinra68rJ_sQcycX9wVX33tVkJcu2AbA_0QkuyZr@mail.gmail.com>
	<AANLkTik4FikbJjQfSoQiJJze8XFLJYSVPAA0TUH6i7td@mail.gmail.com>
	<0ACA1E73-98AC-4E83-8B4F-16FF1B4EA672@ipeva.fr>
Message-ID: <AANLkTinEKodoXw2rEbBycpF4+KQdFN66_2Hp9B2AqwQu@mail.gmail.com>

ahh finally we got to know that were loopback calls, that was totaly
unexpected.

our supplier was sending to our customer and back to us and with few of
calls were going to real time supplier and ... that was the reason...

thanks guys for taking interest.


On Fri, Oct 1, 2010 at 6:08 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Steven,
>
> I confirm the default is log B-leg:
>
>     <!-- optional: if not present we do log the b leg -->
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 01/10/2010 ? 17:45, Steven Ayre a ?crit :
>
> Are they A and B leg CDRs? For a bridged call you'll see a CDR for
> both the A and B leg, which as separate channels will each have its
> own UUID but will share the same caller id, destination number and
> call duration.
>
> <!--param name="log-b-leg" value="false"/-->
> That parameter would disable logging the B-leg CDRs. You have it
> commented out. I can't remember the default setting but from what you
> describe, default behaviour might be to log both.
>
> Look at the Direction that is submitted to tell the difference between
> A and B legs. It'll either be inbound (A) or outbound (B). You could
> then ignore B-legs in your PHP script, or handle them differently.
>
> -Steve
>
>
> On 1 October 2010 11:17, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
>
> Dear List,
>
>
> I am using xml_cdr to write the records on disk.
>
>
> today for first time when there was a call pressure around 60-70 concurrent
>
> calls (around 140 session), then FS started to write same CDR record two
>
> times with different/unique file name and also different/unique UUIDs etc..
>
> but the called number, start time, end time everything is same. so its
>
> surely a duplicated CDR.
>
>
> -> I am running FS on cent os, on a 6 core machine with 4  GM RAM.
>
> -> FS version: FreeSWITCH Version 1.0.head (git-72baaf6 2010-09-21 19-22-28
>
> -0500) (is anything regarding cdr changed in new version? so i can upgrade)
>
>
> Please note that system was running fine with few call attempts and i
>
> never faced this prob, only in case of 6-70 councurrent calls
>
>
> My xml_cr.conf:
>
>
> <configuration name="xml_cdr.conf" description="XML CDR CURL logger">
>
>   <settings>
>
>     <!-- the url to post to if blank web posting is disabled  -->
>
>     <!-- <param name="url" value="http://localhost/cdr_curl/post.php"/>
> -->
>
>
>     <!-- optional: credentials to send to web server -->
>
>     <!--    <param name="cred" value="user:pass"/> -->
>
>
>     <!-- the total number of retries (not counting the first 'try') to post
>
> to webserver incase of failure -->
>
>     <param name="retries" value="2"/>
>
>
>     <!-- delay between retries in seconds, default is 5 seconds -->
>
>     <!-- <param name="delay" value="1"/> -->
>
>
>     <!-- Log via http and on disk, default is false -->
>
>     <!-- <param name="log-http-and-disk" value="true"/> -->
>
>
>     <!-- optional: if not present we do not log every record to disk -->
>
>     <!-- either an absolute path, a relative path assuming ${prefix}/logs
> or
>
> a blank value will default to ${prefix}/logs/xml_cdr -->
>
>     <param name="log-dir" value="/opt/fscdr"/>
>
>
>     <!-- optional: if not present we do log the b leg -->
>
>     <!-- true or false if we should create a cdr for the b leg of a call-->
>
>     <!--param name="log-b-leg" value="false"/-->
>
>
>     <!-- optional: if not present, all filenames are the uuid of the call
>
> -->
>
>     <!-- true or false if a leg files are prefixed "a_" -->
>
>     <param name="prefix-a-leg" value="true"/>
>
>
>     <!-- encode the post data may be 'true' for url encoding, 'false' for
> no
>
> encoding or 'base64' for base64 encoding -->
>
>     <param name="encode" value="true"/>
>
>
>     <!-- optional: set to true to disable Expect: 100-continue lighttpd
>
> requires this setting -->
>
>     <!--<param name="disable-100-continue" value="true"/>-->
>
>
>     <!-- optional: full path to the error log dir for failed web posts if
>
> not specified its the same as log-dir -->
>
>     <!-- either an absolute path, a relative path assuming ${prefix}/logs
> or
>
> a blank or omitted value will default to ${prefix}/logs/xml_cdr -->
>
>     <param name="err-log-dir" value="/opt/error_cdr"/>
>
>
>     <!-- which auhtentification scheme to use. Supported values are: basic,
>
> digest, NTLM, GSS-NEGOTIATE or "any" for automatic detection -->
>
>     <!--<param name="auth-scheme" value="basic"/>-->
>
>
>     <!-- optional: this will enable the CA root certificate check by
> libcurl
>
> to
>
>          verify that the certificate was issued by a major Certificate
>
> Authority.
>
>          note: default value is disabled. only enable if you want this! -->
>
>     <!--<param name="enable-cacert-check" value="true"/>-->
>
>     <!-- optional: verify that the server is actually the one listed in the
>
> cert -->
>
>     <!-- <param name="enable-ssl-verifyhost" value="true"/> -->
>
>
>     <!-- optional: these options can be used to specify custom SSL
>
> certificates
>
>          to use for HTTPS communications. Either use both options or
>
> neither.
>
>          Specify your public key with 'ssl-cert-path' and the private key
>
> with
>
>          'ssl-key-path'. If your private key has a password, specify it
> with
>
>          'ssl-key-password'. -->
>
>     <!-- <param name="ssl-cert-path"
>
> value="$${base_dir}/conf/certs/public_key.pem"/> -->
>
>     <!-- <param name="ssl-key-path"
>
> value="$${base_dir}/conf/certs/private_key.pem"/> -->
>
>     <!-- <param name="ssl-key-password" value="MyPrivateKeyPassword"/> -->
>
>
>     <!-- optional: use a custom CA certificate in PEM format to verify the
>
> peer
>
>          with. This is useful if you are acting as your own certificate
>
> authority.
>
>          note: only makes sense if used in combination with
>
> "enable-cacert-check." -->
>
>     <!-- <param name="ssl-cacert-file"
>
> value="$${base_dir}/conf/certs/cacert.pem"/> -->
>
>
>     <!-- optional: specify the SSL version to force HTTPS to use. Valid
>
> options are
>
>          "SSLv3" and "TLSv1". Otherwise libcurl will auto-negotiate the
>
> version. -->
>
>     <!-- <param name="ssl-version" value="TLSv1"/> -->
>
>
>     <!-- optional: enables cookies and stores them in the specified file.
>
> -->
>
>     <!-- <param name="cookie-file" value="/tmp/cookie-mod_xml_curl.txt"/>
>
> -->
>
>   </settings>
>
> </configuration>
>
>
>
> Many Thanks
>
> - Saeed
>
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jcasale at activenetwerx.com  Fri Oct  1 10:26:48 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 1 Oct 2010 17:26:48 +0000
Subject: [Freeswitch-users] Simultaneous ring of all phones registered
 with the same extension number
In-Reply-To: <AANLkTimRmtA=e4C3tVAHf_+pTu3GvWKU7eJ6_eFq3j50@mail.gmail.com>
References: <AANLkTimRmtA=e4C3tVAHf_+pTu3GvWKU7eJ6_eFq3j50@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E906D85A63@prato.activenetwerx.local>

>I have registered 2 sip phones using the same extension number.
>
>When I dial this extension number, only one sip phone rings.
>
>However I need both of them to start ringing.

That doesn't sound wise to me, make two extensions, then use a ring/call group.
Goto Packt, download the free chapter from the FS book, start reading. It's in there...

jlc


From msc at freeswitch.org  Fri Oct  1 10:29:44 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Oct 2010 10:29:44 -0700
Subject: [Freeswitch-users] Simultaneous ring of all phones registered
 with the same extension number
In-Reply-To: <AANLkTimRmtA=e4C3tVAHf_+pTu3GvWKU7eJ6_eFq3j50@mail.gmail.com>
References: <AANLkTimRmtA=e4C3tVAHf_+pTu3GvWKU7eJ6_eFq3j50@mail.gmail.com>
Message-ID: <AANLkTikjwPxCGao1U=z4QtR-jJupVvcxnTwgbSh0T0jY@mail.gmail.com>

Okay, we need some information:
Are both phones on the same LAN as FreeSWITCH? If not, is NAT involved?
If you register each one individually and call the ext do they work
independently? (Sanity test)
When both phones are registered do they show up when you do "sofia status
profile xxx" ?

You'll definitely want to examine a SIP trace of calls to each phone
individually and then when they both are registered. Hopefully there will be
a clue as to what is happening.

-MC

On Fri, Oct 1, 2010 at 10:00 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> Hi,
> Could not find this answer, so would appreciate any hints.
>
> I have registered 2 sip phones using the *same *extension number.
>
> When I dial this extension number, only one sip phone rings.
>
> However I need both of them to start ringing.
>
> Is it possible, that when I dial the number - both other phones will ring,
> not only one..?
> And when I answer on a phone, another one would stop ringing.
>
>
> Thank you,
> Vitalie
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From aland at burngreave.net  Fri Oct  1 10:36:55 2010
From: aland at burngreave.net (Alan Dawson)
Date: Fri, 1 Oct 2010 18:36:55 +0100
Subject: [Freeswitch-users] Echo on sip trunk
Message-ID: <20101001173655.GB16497@apple.rat.burntout.org>

Hi, 

I have a freeswitch box, with a sip trunk to a voip provider.

When making / receiving external calls the external party gets an echo, which the freeswitch users dont suffer from.

What can I do to reduce that.  Thanks

Alan Dawson
-- 
GPG key: http://aland.burngreave.net/files/e81a4bba.gpg.pub.asc
Key Transition: http://aland.burngreave.net/files/keytransition.txt.asc
Further Reading: https://we.riseup.net/alster/openpgp-dsa1-key-rollover
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From peder at networkoblivion.com  Fri Oct  1 10:44:11 2010
From: peder at networkoblivion.com (Peder)
Date: Fri, 1 Oct 2010 12:44:11 -0500
Subject: [Freeswitch-users] Simultaneous ring of all phones registered
	with the same extension number
In-Reply-To: <AANLkTimRmtA=e4C3tVAHf_+pTu3GvWKU7eJ6_eFq3j50@mail.gmail.com>
References: <AANLkTimRmtA=e4C3tVAHf_+pTu3GvWKU7eJ6_eFq3j50@mail.gmail.com>
Message-ID: <012701cb6190$41b61cf0$c52256d0$@com>

Do you have multiple registration enabled?  I don't think it is on by
default, so that could definitely cause what you see.  Check internal.xml:

 

<!-- Can be 'true' or 'contact' -->

<!--<param name="multiple-registrations" value="contact"/>-->

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vitalii
Colosov
Sent: Friday, October 01, 2010 12:00 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Simultaneous ring of all phones registered with
the same extension number

 

Hi,

Could not find this answer, so would appreciate any hints.

 

I have registered 2 sip phones using the same extension number.

 

When I dial this extension number, only one sip phone rings.

 

However I need both of them to start ringing.

 

Is it possible, that when I dial the number - both other phones will ring,
not only one..? 

And when I answer on a phone, another one would stop ringing.

 

 

Thank you,

Vitalie

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From sos at sokhapkin.dyndns.org  Fri Oct  1 10:46:37 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 1 Oct 2010 13:46:37 -0400
Subject: [Freeswitch-users] Echo on sip trunk
In-Reply-To: <20101001173655.GB16497@apple.rat.burntout.org>
References: <20101001173655.GB16497@apple.rat.burntout.org>
Message-ID: <201010011346.38050.sos@sokhapkin.dyndns.org>

The question is not related to FS. Lower speaker volume on user's phone.

On Friday 01 October 2010, Alan Dawson wrote:
> Hi,
> 
> I have a freeswitch box, with a sip trunk to a voip provider.
> 
> When making / receiving external calls the external party gets an echo,
>  which the freeswitch users dont suffer from.
> 
> What can I do to reduce that.  Thanks
> 
> Alan Dawson
> 



From covici at ccs.covici.com  Fri Oct  1 10:58:13 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 01 Oct 2010 13:58:13 -0400
Subject: [Freeswitch-users] my info as appearing in conference
In-Reply-To: <AANLkTikYMSsung_ECbCZhNex5r67Xd4oCDOO9sXxow61@mail.gmail.com>
References: <28922.1285889502@ccs.covici.com>
	<AANLkTik7cq0s_H=ELuLXd-ifBib683mzGOj7X0AwSi8u@mail.gmail.com>
	<29684.1285894407@ccs.covici.com>
	<AANLkTi=bvTKjqrP_bztBSrZavPXfTn8=_0SGhhzOPPug@mail.gmail.com>
	<31987.1285897148@ccs.covici.com>
	<AANLkTikYMSsung_ECbCZhNex5r67Xd4oCDOO9sXxow61@mail.gmail.com>
Message-ID: <18462.1285955893@ccs.covici.com>

I fixed the problem -- in the extension 9888 and friends, the caller ids
were not set at all.  Now it seems to be coming in correctly.

Michael Collins <msc at freeswitch.org> wrote:

> What, exactly, are you doing to establish the call?
> -MC
> 
> On Thu, Sep 30, 2010 at 6:39 PM, <covici at ccs.covici.com> wrote:
> 
> > Well, in mod_portaudio they are set to the variables in vars.xml and if
> > I call my cell phone it knows the callerid number and if I go into an
> > asterisk conference which I can display I see both my number and name,
> > so I am confused as to where the 0's are coming from.  I am using
> > extension 9888 to get into the conference maybe no callerid is set
> > there, I will check that, but it still seems strange.
> >
> > Michael Collins <msc at freeswitch.org> wrote:
> >
> > > Default dialplan? If not, pastebin your dp. At some point you need to set
> > > the caller id name/number channel variables. Authenticated calls will
> > have
> > > those populated if you define them in the user directory, otherwise you
> > will
> > > need to manually set them in the dialplan, or at the very least modify
> > the
> > > default dialplan to have something more meaningful than all zeroes.
> > >
> > > -MC
> > >
> > > On Thu, Sep 30, 2010 at 5:53 PM, <covici at ccs.covici.com> wrote:
> > >
> > > > Yes they are in vars.xml -- that is why I am writing.
> > > >
> > > > William Suffill <william.suffill at gmail.com> wrote:
> > > >
> > > > > vars.xml should have the default callerid details used.
> > > > >
> > > > > ----------------------------------------------------
> > > > > Alternatives:
> > > > >
> > > > > ----------------------------------------------------
> > > > > _______________________________________________
> > > > > FreeSWITCH-users mailing list
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > > http://www.freeswitch.org
> > > >
> > > > --
> > > > Your life is like a penny.  You're going to lose it.  The question is:
> > > > How do
> > > > you spend it?
> > > >
> > > >         John Covici
> > > >         covici at ccs.covici.com
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > http://www.freeswitch.org
> > > >
> > >
> > > ----------------------------------------------------
> > > Alternatives:
> > >
> > > ----------------------------------------------------
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From covici at ccs.covici.com  Fri Oct  1 11:02:11 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 01 Oct 2010 14:02:11 -0400
Subject: [Freeswitch-users] voicemail email not working.
In-Reply-To: <00d301cb6183$64a84330$2df8c990$@com>
References: <15521.1285946991@ccs.covici.com>
	<00d301cb6183$64a84330$2df8c990$@com>
Message-ID: <18538.1285956131@ccs.covici.com>

hmmm, I looked at the template -- the notify template has to:
${voicemail_notify_email} and the other template has to:
${voicemail_email} -- do I need to change them as well?

Peder <peder at networkoblivion.com> wrote:

> My guess would be that you are using the wrong variable names.  From the
> wiki:
> 
> http://wiki.freeswitch.org/wiki/Mod_voicemail
> 
> 
> "vm-notify-mailto"
> The address you want notification messages sent to default: same as
> vm-mailto
> This is defined in the directory for the particular user as a param.
> 
> "vm-mailto"
> This is the user's email address default: undefined (originally called
> email-addr. Use vm-mailto instead)
> This is defined in the directory for the particular user as a param.
> Multiple email addresses can be defined using a comma-separated list.
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> covici at ccs.covici.com
> Sent: Friday, October 01, 2010 10:30 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] voicemail email not working.
> 
> Hi.  I left a test message and instead of using the variable I have in
> my user directory, it tried to deliver to username at domain name which is
> invalid -- so why did it not use the variable voicemail_email I have in
> the user directory and it completely ignored the voicemail_notify_email
> I have as well?
> 
> Any assistance would be appreciated.
> 
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From msc at freeswitch.org  Fri Oct  1 11:01:55 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Oct 2010 11:01:55 -0700
Subject: [Freeswitch-users] Simultaneous ring of all phones registered
 with the same extension number
In-Reply-To: <012701cb6190$41b61cf0$c52256d0$@com>
References: <AANLkTimRmtA=e4C3tVAHf_+pTu3GvWKU7eJ6_eFq3j50@mail.gmail.com>
	<012701cb6190$41b61cf0$c52256d0$@com>
Message-ID: <AANLkTik9cBmi1qbytdsBibo9+mPz2uxPVoKe7GSPweKv@mail.gmail.com>

Good catch. If you don't have multiple-registrations set to true then the
phones' registrations will clobber each other. I just tested with and
without multiple-registrations set to true and sure enough Peder is correct.


The other question for the OP is whether or not he really needs multiple
registrations or if JLC's suggestion about ring groups is a better way to
go. Only he can answer that, but at least now he has a few options.

Thanks guys,
MC

On Fri, Oct 1, 2010 at 10:44 AM, Peder <peder at networkoblivion.com> wrote:

>  Do you have multiple registration enabled?  I don?t think it is on by
> default, so that could definitely cause what you see.  Check internal.xml:
>
>
>
> <!-- Can be 'true' or 'contact' -->
>
> <!--<param name="multiple-registrations" value="contact"/>-->
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Vitalii
> Colosov
> *Sent:* Friday, October 01, 2010 12:00 PM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] Simultaneous ring of all phones registered
> with the same extension number
>
>
>
> Hi,
>
> Could not find this answer, so would appreciate any hints.
>
>
>
> I have registered 2 sip phones using the *same *extension number.
>
>
>
> When I dial this extension number, only one sip phone rings.
>
>
>
> However I need both of them to start ringing.
>
>
>
> Is it possible, that when I dial the number - both other phones will ring,
> not only one..?
>
> And when I answer on a phone, another one would stop ringing.
>
>
>
>
>
> Thank you,
>
> Vitalie
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Oct  1 11:03:15 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Oct 2010 11:03:15 -0700
Subject: [Freeswitch-users] my info as appearing in conference
In-Reply-To: <18462.1285955893@ccs.covici.com>
References: <28922.1285889502@ccs.covici.com>
	<AANLkTik7cq0s_H=ELuLXd-ifBib683mzGOj7X0AwSi8u@mail.gmail.com>
	<29684.1285894407@ccs.covici.com>
	<AANLkTi=bvTKjqrP_bztBSrZavPXfTn8=_0SGhhzOPPug@mail.gmail.com>
	<31987.1285897148@ccs.covici.com>
	<AANLkTikYMSsung_ECbCZhNex5r67Xd4oCDOO9sXxow61@mail.gmail.com>
	<18462.1285955893@ccs.covici.com>
Message-ID: <AANLkTikH1R8XDEVYhNEg_jCFzkLnaQiHt2RHvsaYAJfH@mail.gmail.com>

On Fri, Oct 1, 2010 at 10:58 AM, <covici at ccs.covici.com> wrote:

> I fixed the problem -- in the extension 9888 and friends, the caller ids
> were not set at all.  Now it seems to be coming in correctly.
>
> Good work. :)
-MC
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From freeswitch-list at puzzled.xs4all.nl  Fri Oct  1 11:21:47 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Fri, 01 Oct 2010 20:21:47 +0200
Subject: [Freeswitch-users] License of libzrtp for usage with FreeSWITCH?
In-Reply-To: <C41DB9FA-0019-40EB-9CDF-0F3C3BA6276C@freeswitch.org>
References: <4CA53F77.5040106@puzzled.xs4all.nl>
	<C41DB9FA-0019-40EB-9CDF-0F3C3BA6276C@freeswitch.org>
Message-ID: <4CA626BB.5060501@puzzled.xs4all.nl>

On 10/01/2010 03:56 PM, Brian West wrote:
> Patrick
> 	Since we do not ship with the lib its up to the end user to combine them.  So the license concern is not an issue for the FreeSWITCH project.  Its now on you to comply with all the licensing.
>
> Thanks,
> Brian

Hi Brian,

Thank you for your feedback.

Regards,
Patrick


From freeswitch-list at puzzled.xs4all.nl  Fri Oct  1 11:25:05 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Fri, 01 Oct 2010 20:25:05 +0200
Subject: [Freeswitch-users] License of libzrtp for usage with FreeSWITCH?
In-Reply-To: <4CA5FEF8.5040102@infosecurity.ch>
References: <4CA53F77.5040106@puzzled.xs4all.nl>	<C41DB9FA-0019-40EB-9CDF-0F3C3BA6276C@freeswitch.org>
	<4CA5FEF8.5040102@infosecurity.ch>
Message-ID: <4CA62781.2040909@puzzled.xs4all.nl>

On 10/01/2010 05:32 PM, Fabio Pietrosanti (naif) wrote:
> On http://zfoneproject.com/lic_policy.html it's written that the libzrtp
> is licensed under AGPL and only zfone it's not under AGPL.
>
> It sounds just like a typo / not updated statement in the code as the
> website refer to AGPL licensed.
>
> Fabio

Thanks Fabio. All clear now.

Regards,
Patrick


From msc at freeswitch.org  Fri Oct  1 12:05:19 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Oct 2010 12:05:19 -0700
Subject: [Freeswitch-users] Multiple "voices" under mod_say_en
In-Reply-To: <787302A89ACCE24DA8F56DA101E77C842B473C0A1B@THHS2E12BE1X.hostedservice2.net>
References: <787302A89ACCE24DA8F56DA101E77C842B473C0A1B@THHS2E12BE1X.hostedservice2.net>
Message-ID: <AANLkTinD61mjiCsnODpqfQqzib-YLz553rHjDwFoNym6@mail.gmail.com>

Neil,

Try setting channel variable ${sound_prefix} prior to calling your say
application. It's an imperfect solution until we have something more elegant
in place. Just note that if you need to switch between US and UK sounds on
the same call you'll need to change the sound_prefix on the channel each
time.

The default sound_prefix is:
/usr/local/freeswitch/sounds/en/us/callie

So change it to:
/usr/local/freeswitch/sounds/en/uk/<your_voice>

And then call say and it should use your uk voice.

-MC

On Fri, Oct 1, 2010 at 8:32 AM, Neil Burgess <neil.burgess at redmatter.com>wrote:

>  Hi,
>
>
>
> I?d like to be able to use two sets of language files for US and UK within
> the say module, allowing either to be utilised with the spell, items, etc
> capabilities, based on setting some channel variable!
>
>
>
> At the moment, I can?t see how to achieve this since it appears that the
> ?say_en? module for manipulating these strings is linked to the language
> name attribute as below.
>
>
>
> <language name="en" sound-path="$${base_dir}/sounds/en/us/callie"
> tts-engine="unimrcp:loquendo7-mrcp2" tts-voice="kate">
>
>
>
> Is there any way of supporting multiple ?voices? under the same mod_say_en
>
>
>
> Thanks,
>
> Neil
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peder at networkoblivion.com  Fri Oct  1 12:29:13 2010
From: peder at networkoblivion.com (Peder)
Date: Fri, 1 Oct 2010 14:29:13 -0500
Subject: [Freeswitch-users] voicemail email not working.
In-Reply-To: <18538.1285956131@ccs.covici.com>
References: <15521.1285946991@ccs.covici.com>	<00d301cb6183$64a84330$2df8c990$@com>
	<18538.1285956131@ccs.covici.com>
Message-ID: <01c201cb619e$ee16cdb0$ca446910$@com>

No.  Somewhere in the voicemail module, that appears to be added as a
variable with the setting from vm-mailto (no clue why though).  I just
verified that my directory entry uses vm-mailto and I do receive the message
to the correct address.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
covici at ccs.covici.com
Sent: Friday, October 01, 2010 1:02 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] voicemail email not working.

hmmm, I looked at the template -- the notify template has to:
${voicemail_notify_email} and the other template has to:
${voicemail_email} -- do I need to change them as well?

Peder <peder at networkoblivion.com> wrote:

> My guess would be that you are using the wrong variable names.  From the
> wiki:
> 
> http://wiki.freeswitch.org/wiki/Mod_voicemail
> 
> 
> "vm-notify-mailto"
> The address you want notification messages sent to default: same as
> vm-mailto
> This is defined in the directory for the particular user as a param.
> 
> "vm-mailto"
> This is the user's email address default: undefined (originally called
> email-addr. Use vm-mailto instead)
> This is defined in the directory for the particular user as a param.
> Multiple email addresses can be defined using a comma-separated list.
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> covici at ccs.covici.com
> Sent: Friday, October 01, 2010 10:30 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] voicemail email not working.
> 
> Hi.  I left a test message and instead of using the variable I have in
> my user directory, it tried to deliver to username at domain name which is
> invalid -- so why did it not use the variable voicemail_email I have in
> the user directory and it completely ignored the voicemail_notify_email
> I have as well?
> 
> Any assistance would be appreciated.
> 
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com

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



From aland at burngreave.net  Fri Oct  1 13:01:44 2010
From: aland at burngreave.net (Alan Dawson)
Date: Fri, 1 Oct 2010 21:01:44 +0100
Subject: [Freeswitch-users] Echo on sip trunk
In-Reply-To: <201010011346.38050.sos@sokhapkin.dyndns.org>
References: <20101001173655.GB16497@apple.rat.burntout.org>
	<201010011346.38050.sos@sokhapkin.dyndns.org>
Message-ID: <20101001200143.GC16497@apple.rat.burntout.org>

On Fri, Oct 01, 2010 at 01:46:37PM -0400, Sergey Okhapkin wrote:
> The question is not related to FS. Lower speaker volume on user's phone.

hi, interesting response.

When we dont use the FS box, and just make a direct connection to the VOIP provider no echo occurs.

This leads me to believe that it is an artefact related to the PBX

-- 
GPG key: http://aland.burngreave.net/files/e81a4bba.gpg.pub.asc
Key Transition: http://aland.burngreave.net/files/keytransition.txt.asc
Further Reading: https://we.riseup.net/alster/openpgp-dsa1-key-rollover
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From msc at freeswitch.org  Fri Oct  1 13:01:32 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Oct 2010 13:01:32 -0700
Subject: [Freeswitch-users] Intercept Ringing Call
In-Reply-To: <009801cb617c$763d7810$62b86830$@teotech.com>
References: <009801cb617c$763d7810$62b86830$@teotech.com>
Message-ID: <AANLkTim1-+x27rZwZ33YRPAvDPtJc3sa7braw58S7nHC@mail.gmail.com>

I cannot reproduce this symptom on latest git. I have this scenario:

x1001: Snom 300
x1002: eyeBeam
x1005: Poly 320

Call from x1002 to x1001, x1001 is ringin
At x1005 dial **1001, x1005 is now talking to x1002

If I have the scenario wrong please let me know.
-MC

On Fri, Oct 1, 2010 at 8:22 AM, Jerry Richards
<jerry.richards at teotech.com>wrote:

> Can I intercept a call in ringing state only?  The default **<number>
> feature can steal a call from someone if multiple people try to answer a
> ringing call.
>
> Thanks,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Oct  1 13:32:22 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Oct 2010 13:32:22 -0700
Subject: [Freeswitch-users] Echo on sip trunk
In-Reply-To: <20101001200143.GC16497@apple.rat.burntout.org>
References: <20101001173655.GB16497@apple.rat.burntout.org>
	<201010011346.38050.sos@sokhapkin.dyndns.org>
	<20101001200143.GC16497@apple.rat.burntout.org>
Message-ID: <AANLkTimCZkOz=MaHqZG+jAvSOEOpJNfzVVOX7OPL557U@mail.gmail.com>

This will probably take some good old fashioned detective work on your part.
I'd recommend gathering all the information about your topology, devices,
etc. and putting it in pastebin.freeswitch.org. I'd also get a pcap of he
traffic on the FS box (including media) and analyze in Wireshark. Assuming
that you don't have an ATA causing the echo then you're looking at the more
traditional network issues like delay.

You might want to start going down the rabbit hole by checking out this
article and the links therein:
http://networkinstruments.wordpress.com/2007/10/25/voip-troubleshooting-exploring-echo-echo-and-delay/

I'm not saying it's not FreeSWITCH, but since we use it all day, every day,
and the only echo issues we ever have are related to network issues, I'd
recommend barking up that tree first.

-MC

On Fri, Oct 1, 2010 at 1:01 PM, Alan Dawson <aland at burngreave.net> wrote:

> On Fri, Oct 01, 2010 at 01:46:37PM -0400, Sergey Okhapkin wrote:
> > The question is not related to FS. Lower speaker volume on user's phone.
>
> hi, interesting response.
>
> When we dont use the FS box, and just make a direct connection to the VOIP
> provider no echo occurs.
>
> This leads me to believe that it is an artefact related to the PBX
>
> --
> GPG key: http://aland.burngreave.net/files/e81a4bba.gpg.pub.asc
> Key Transition: http://aland.burngreave.net/files/keytransition.txt.asc
> Further Reading: https://we.riseup.net/alster/openpgp-dsa1-key-rollover
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.10 (GNU/Linux)
>
> iQIcBAEBCAAGBQJMpj4mAAoJEAGiBQHoGku6pgsP/iLRaa7sMpyaw8Ybq3bD+svA
> BXP9W9rrs8Ss4MydAV6Kic15kHwKLxrfdqaxw+Jg/WGHiw5IOSefMcYxoiDrvbTK
> FAUxGo+CIwH0/hg56bTvrzjixFjq7nlx31dYA+ftcR6Q+0UyjMdSW2oJgOfGHKfI
> PpDqUDd6L7BEKSYY5RKUrNa+ZO0fU9yBXVAN3ONzUCFBGj9B3fzIErSUa7c10Le9
> e5rPUzrPoVsyWFyjT/drKPQ3e/bSoStKHgc/xwZ6bvvBzVxA6s2LMq3XLwofL8TI
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> tmWnM2aRx9XZh66X7ChCPUv+lS2NESRt/X+pLHC+GfQ/Iy4gN5WeSTQGSiUiFPCx
> sE4ytLd5rSpL0RSdEs3eAaR+MzAMOyWJ0m+QX7HtfuiOA/1CnXGifdVlmNmmEDsk
> Qr51oGwmczNCBaUN7mKdyfunFUILj5f1rLfa1tkCBBOxiP5WOO4GtUtiey4IFA6Z
> aASDHygJ6k2FYxTjeuSE2fdxoj58d240EhNFA9MaOBIN47Q6aXyewKbbE7prlIMV
> Nu92XS2wNwhUda+gftiEN/Sy3m2XPyP/WiiF2TBGjM5XtYORTdH/3GZYhf4+FkBy
> hzd/qJ1c5SE0twHfrRiy
> =BtSr
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From covici at ccs.covici.com  Fri Oct  1 13:45:30 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 01 Oct 2010 16:45:30 -0400
Subject: [Freeswitch-users] voicemail email not working.
In-Reply-To: <00d301cb6183$64a84330$2df8c990$@com>
References: <15521.1285946991@ccs.covici.com>
	<00d301cb6183$64a84330$2df8c990$@com>
Message-ID: <20205.1285965930@ccs.covici.com>

OK, I have vm-mailto working, but I cannot get the vm-notify-mailto to
work -- it completely ignores this one.  Looking at the source it seems
to need send_notify, but I can't find a way to set that.  I did try to
set it as avariable, but no joy.

Peder <peder at networkoblivion.com> wrote:

> My guess would be that you are using the wrong variable names.  From the
> wiki:
> 
> http://wiki.freeswitch.org/wiki/Mod_voicemail
> 
> 
> "vm-notify-mailto"
> The address you want notification messages sent to default: same as
> vm-mailto
> This is defined in the directory for the particular user as a param.
> 
> "vm-mailto"
> This is the user's email address default: undefined (originally called
> email-addr. Use vm-mailto instead)
> This is defined in the directory for the particular user as a param.
> Multiple email addresses can be defined using a comma-separated list.
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> covici at ccs.covici.com
> Sent: Friday, October 01, 2010 10:30 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] voicemail email not working.
> 
> Hi.  I left a test message and instead of using the variable I have in
> my user directory, it tried to deliver to username at domain name which is
> invalid -- so why did it not use the variable voicemail_email I have in
> the user directory and it completely ignored the voicemail_notify_email
> I have as well?
> 
> Any assistance would be appreciated.
> 
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From vetali100 at gmail.com  Fri Oct  1 13:46:20 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Fri, 1 Oct 2010 23:46:20 +0300
Subject: [Freeswitch-users] Simultaneous ring of all phones registered
 with the same extension number
In-Reply-To: <AANLkTik9cBmi1qbytdsBibo9+mPz2uxPVoKe7GSPweKv@mail.gmail.com>
References: <AANLkTimRmtA=e4C3tVAHf_+pTu3GvWKU7eJ6_eFq3j50@mail.gmail.com>
	<012701cb6190$41b61cf0$c52256d0$@com>
	<AANLkTik9cBmi1qbytdsBibo9+mPz2uxPVoKe7GSPweKv@mail.gmail.com>
Message-ID: <AANLkTi=fVTMd5c5NNqeRKDOZJ-ywtt7opy3wtDedg7EH@mail.gmail.com>

I have enabled multiple registration, but I had the same result - only 1
phone was ringing.

Finally I found one link and it solved the problem - now all the phones are
ringing:

http://freeswitch-users.2379917.n2.nabble.com/Multiple-Registrations-randomly-only-one-device-ring-td4707557.html

I need to use:
<action application="bridge" data="${sofia_contact(${dialed_extension}@
${domain_name})}"/>

Instead of the default one I had:
<action application="bridge"
data="sofia/internal/${dialed_extension}%${domain_name}"/>


<http://freeswitch-users.2379917.n2.nabble.com/Multiple-Registrations-randomly-only-one-device-ring-td4707557.html>Any
comments would be much appreciated :-)


Thank you,
Vitalie

2010/10/1 Michael Collins <msc at freeswitch.org>

> Good catch. If you don't have multiple-registrations set to true then the
> phones' registrations will clobber each other. I just tested with and
> without multiple-registrations set to true and sure enough Peder is correct.
>
>
> The other question for the OP is whether or not he really needs multiple
> registrations or if JLC's suggestion about ring groups is a better way to
> go. Only he can answer that, but at least now he has a few options.
>
> Thanks guys,
> MC
>
> On Fri, Oct 1, 2010 at 10:44 AM, Peder <peder at networkoblivion.com> wrote:
>
>>  Do you have multiple registration enabled?  I don?t think it is on by
>> default, so that could definitely cause what you see.  Check internal.xml:
>>
>>
>>
>> <!-- Can be 'true' or 'contact' -->
>>
>> <!--<param name="multiple-registrations" value="contact"/>-->
>>
>>
>>
>>
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Vitalii
>> Colosov
>> *Sent:* Friday, October 01, 2010 12:00 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* [Freeswitch-users] Simultaneous ring of all phones registered
>> with the same extension number
>>
>>
>>
>> Hi,
>>
>> Could not find this answer, so would appreciate any hints.
>>
>>
>>
>> I have registered 2 sip phones using the *same *extension number.
>>
>>
>>
>> When I dial this extension number, only one sip phone rings.
>>
>>
>>
>> However I need both of them to start ringing.
>>
>>
>>
>> Is it possible, that when I dial the number - both other phones will ring,
>> not only one..?
>>
>> And when I answer on a phone, another one would stop ringing.
>>
>>
>>
>>
>>
>> Thank you,
>>
>> Vitalie
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Oct  1 13:55:29 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Oct 2010 13:55:29 -0700
Subject: [Freeswitch-users] Simultaneous ring of all phones registered
 with the same extension number
In-Reply-To: <AANLkTi=fVTMd5c5NNqeRKDOZJ-ywtt7opy3wtDedg7EH@mail.gmail.com>
References: <AANLkTimRmtA=e4C3tVAHf_+pTu3GvWKU7eJ6_eFq3j50@mail.gmail.com>
	<012701cb6190$41b61cf0$c52256d0$@com>
	<AANLkTik9cBmi1qbytdsBibo9+mPz2uxPVoKe7GSPweKv@mail.gmail.com>
	<AANLkTi=fVTMd5c5NNqeRKDOZJ-ywtt7opy3wtDedg7EH@mail.gmail.com>
Message-ID: <AANLkTikx9Ah0QgwHHXYouoyo=uvczmURQMaiWdwqJk73@mail.gmail.com>

Go to fs_cli and type:

sofia_contact USER at DOMAIN

where USER is ${dialed_extension} and DOMAIN is ${domain_name}
You'll see that it gives you a nicely formatted dialstring. :)

-MC

On Fri, Oct 1, 2010 at 1:46 PM, Vitalii Colosov <vetali100 at gmail.com> wrote:

> I have enabled multiple registration, but I had the same result - only 1
> phone was ringing.
>
> Finally I found one link and it solved the problem - now all the phones are
> ringing:
>
>
> http://freeswitch-users.2379917.n2.nabble.com/Multiple-Registrations-randomly-only-one-device-ring-td4707557.html
>
> I need to use:
> <action application="bridge" data="${sofia_contact(${dialed_extension}@
> ${domain_name})}"/>
>
> Instead of the default one I had:
> <action application="bridge"
> data="sofia/internal/${dialed_extension}%${domain_name}"/>
>
>
>
> <http://freeswitch-users.2379917.n2.nabble.com/Multiple-Registrations-randomly-only-one-device-ring-td4707557.html>Any
> comments would be much appreciated :-)
>
>
> Thank you,
> Vitalie
>
> 2010/10/1 Michael Collins <msc at freeswitch.org>
>
> Good catch. If you don't have multiple-registrations set to true then the
>> phones' registrations will clobber each other. I just tested with and
>> without multiple-registrations set to true and sure enough Peder is correct.
>>
>>
>> The other question for the OP is whether or not he really needs multiple
>> registrations or if JLC's suggestion about ring groups is a better way to
>> go. Only he can answer that, but at least now he has a few options.
>>
>> Thanks guys,
>> MC
>>
>> On Fri, Oct 1, 2010 at 10:44 AM, Peder <peder at networkoblivion.com> wrote:
>>
>>>  Do you have multiple registration enabled?  I don?t think it is on by
>>> default, so that could definitely cause what you see.  Check internal.xml:
>>>
>>>
>>>
>>> <!-- Can be 'true' or 'contact' -->
>>>
>>> <!--<param name="multiple-registrations" value="contact"/>-->
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Vitalii
>>> Colosov
>>> *Sent:* Friday, October 01, 2010 12:00 PM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* [Freeswitch-users] Simultaneous ring of all phones registered
>>> with the same extension number
>>>
>>>
>>>
>>> Hi,
>>>
>>> Could not find this answer, so would appreciate any hints.
>>>
>>>
>>>
>>> I have registered 2 sip phones using the *same *extension number.
>>>
>>>
>>>
>>> When I dial this extension number, only one sip phone rings.
>>>
>>>
>>>
>>> However I need both of them to start ringing.
>>>
>>>
>>>
>>> Is it possible, that when I dial the number - both other phones will
>>> ring, not only one..?
>>>
>>> And when I answer on a phone, another one would stop ringing.
>>>
>>>
>>>
>>>
>>>
>>> Thank you,
>>>
>>> Vitalie
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peder at networkoblivion.com  Fri Oct  1 13:56:25 2010
From: peder at networkoblivion.com (Peder)
Date: Fri, 1 Oct 2010 15:56:25 -0500
Subject: [Freeswitch-users] voicemail email not working.
In-Reply-To: <20205.1285965930@ccs.covici.com>
References: <15521.1285946991@ccs.covici.com>	<00d301cb6183$64a84330$2df8c990$@com>
	<20205.1285965930@ccs.covici.com>
Message-ID: <023a01cb61ab$1c9834b0$55c89e10$@com>

"vm-notify-email-all-messages"
Setting to true will send a notify email to vm-notify-mailto when a vm is
left (never has attachment) default: false


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
covici at ccs.covici.com
Sent: Friday, October 01, 2010 3:46 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] voicemail email not working.

OK, I have vm-mailto working, but I cannot get the vm-notify-mailto to
work -- it completely ignores this one.  Looking at the source it seems
to need send_notify, but I can't find a way to set that.  I did try to
set it as avariable, but no joy.

Peder <peder at networkoblivion.com> wrote:

> My guess would be that you are using the wrong variable names.  From the
> wiki:
> 
> http://wiki.freeswitch.org/wiki/Mod_voicemail
> 
> 
> "vm-notify-mailto"
> The address you want notification messages sent to default: same as
> vm-mailto
> This is defined in the directory for the particular user as a param.
> 
> "vm-mailto"
> This is the user's email address default: undefined (originally called
> email-addr. Use vm-mailto instead)
> This is defined in the directory for the particular user as a param.
> Multiple email addresses can be defined using a comma-separated list.
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> covici at ccs.covici.com
> Sent: Friday, October 01, 2010 10:30 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] voicemail email not working.
> 
> Hi.  I left a test message and instead of using the variable I have in
> my user directory, it tried to deliver to username at domain name which is
> invalid -- so why did it not use the variable voicemail_email I have in
> the user directory and it completely ignored the voicemail_notify_email
> I have as well?
> 
> Any assistance would be appreciated.
> 
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com

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



From Peter.Hinman at ParcelPool.com  Fri Oct  1 14:50:03 2010
From: Peter.Hinman at ParcelPool.com (Peter Hinman)
Date: Fri, 01 Oct 2010 15:50:03 -0600
Subject: [Freeswitch-users] Alphanumeric case sensitivity
Message-ID: <4CA6578B.6090704@ParcelPool.com>

  I'm trying to get FS working with alphanumeric UserIDs.  I'm running 
into a problem when user1 registers using mixed case (User1 at example.com) 
and then user2 tries to reach user1 by dialing "user1" instead of "User1".

FS fails to make the connection between the two extensions and reports:
[ERR] switch_ivr_originate.c:2648 Cannot create outgoing channel of type 
[error] cause: [USER_NOT_REGISTERED]

The dial plan string for bridging my internal users is:
<action application="bridge" 
data="user/${dialed_extension}@${domain_name}"/>

Is there a way to use alphanumeric UserIDs and bridge calls between 
internal extensions when the dialing user doesn't know exactly how 
(upper, lower, mixed case) the destination user/extension registered 
with FS?

-- 
Peter Hinman
ParcelPool.com
Office: 801-434-7400
Cell: 801-830-8292




From covici at ccs.covici.com  Fri Oct  1 15:20:34 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 01 Oct 2010 18:20:34 -0400
Subject: [Freeswitch-users] voicemail email not working.
In-Reply-To: <023a01cb61ab$1c9834b0$55c89e10$@com>
References: <15521.1285946991@ccs.covici.com>
	<00d301cb6183$64a84330$2df8c990$@com>
	<20205.1285965930@ccs.covici.com>
	<023a01cb61ab$1c9834b0$55c89e10$@com>
Message-ID: <6610.1285971634@ccs.covici.com>

OK, must have missed that one -- thanks.

Peder <peder at networkoblivion.com> wrote:

> "vm-notify-email-all-messages"
> Setting to true will send a notify email to vm-notify-mailto when a vm is
> left (never has attachment) default: false
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> covici at ccs.covici.com
> Sent: Friday, October 01, 2010 3:46 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] voicemail email not working.
> 
> OK, I have vm-mailto working, but I cannot get the vm-notify-mailto to
> work -- it completely ignores this one.  Looking at the source it seems
> to need send_notify, but I can't find a way to set that.  I did try to
> set it as avariable, but no joy.
> 
> Peder <peder at networkoblivion.com> wrote:
> 
> > My guess would be that you are using the wrong variable names.  From the
> > wiki:
> > 
> > http://wiki.freeswitch.org/wiki/Mod_voicemail
> > 
> > 
> > "vm-notify-mailto"
> > The address you want notification messages sent to default: same as
> > vm-mailto
> > This is defined in the directory for the particular user as a param.
> > 
> > "vm-mailto"
> > This is the user's email address default: undefined (originally called
> > email-addr. Use vm-mailto instead)
> > This is defined in the directory for the particular user as a param.
> > Multiple email addresses can be defined using a comma-separated list.
> > 
> > 
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> > covici at ccs.covici.com
> > Sent: Friday, October 01, 2010 10:30 AM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: [Freeswitch-users] voicemail email not working.
> > 
> > Hi.  I left a test message and instead of using the variable I have in
> > my user directory, it tried to deliver to username at domain name which is
> > invalid -- so why did it not use the variable voicemail_email I have in
> > the user directory and it completely ignored the voicemail_notify_email
> > I have as well?
> > 
> > Any assistance would be appreciated.
> > 
> > -- 
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> > 
> >          John Covici
> >          covici at ccs.covici.com
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From msc at freeswitch.org  Fri Oct  1 15:35:24 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Oct 2010 15:35:24 -0700
Subject: [Freeswitch-users] Alphanumeric case sensitivity
In-Reply-To: <4CA6578B.6090704@ParcelPool.com>
References: <4CA6578B.6090704@ParcelPool.com>
Message-ID: <AANLkTim+FOi70+zkeg4ODbz=Dd1TYYW3WFWH3otaShjx@mail.gmail.com>

What's the user_id of "User1 at example.com"? Is it "User1" or "user1"? (BTW,
this is a classic example of why alphanumeric usernames are a bad idea in a
telephony environment.)

I would recommend that you enforce some policy, like always use all lower
case. (I've poked around in the source code and I don't see any easy way to
make alpha usernames be case insensitive.)

-MC

On Fri, Oct 1, 2010 at 2:50 PM, Peter Hinman <Peter.Hinman at parcelpool.com>wrote:

>  I'm trying to get FS working with alphanumeric UserIDs.  I'm running
> into a problem when user1 registers using mixed case (User1 at example.com)
> and then user2 tries to reach user1 by dialing "user1" instead of "User1".
>
> FS fails to make the connection between the two extensions and reports:
> [ERR] switch_ivr_originate.c:2648 Cannot create outgoing channel of type
> [error] cause: [USER_NOT_REGISTERED]
>
> The dial plan string for bridging my internal users is:
> <action application="bridge"
> data="user/${dialed_extension}@${domain_name}"/>
>
> Is there a way to use alphanumeric UserIDs and bridge calls between
> internal extensions when the dialing user doesn't know exactly how
> (upper, lower, mixed case) the destination user/extension registered
> with FS?
>
> --
> Peter Hinman
> ParcelPool.com
> Office: 801-434-7400
> Cell: 801-830-8292
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Fri Oct  1 16:38:36 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 1 Oct 2010 18:38:36 -0500
Subject: [Freeswitch-users] Alphanumeric case sensitivity
In-Reply-To: <AANLkTim+FOi70+zkeg4ODbz=Dd1TYYW3WFWH3otaShjx@mail.gmail.com>
References: <4CA6578B.6090704@ParcelPool.com>
	<AANLkTim+FOi70+zkeg4ODbz=Dd1TYYW3WFWH3otaShjx@mail.gmail.com>
Message-ID: <416C46B3-380C-4D22-9E8F-DBA409584821@freeswitch.org>

It can't be case insensitive.  The A1-Hash calculations would not work for the auth/challenge.

/b

On Oct 1, 2010, at 5:35 PM, Michael Collins wrote:

> What's the user_id of "User1 at example.com"? Is it "User1" or "user1"? (BTW, this is a classic example of why alphanumeric usernames are a bad idea in a telephony environment.)
> 
> I would recommend that you enforce some policy, like always use all lower case. (I've poked around in the source code and I don't see any easy way to make alpha usernames be case insensitive.)
> 
> -MC



From xduvox at gmail.com  Fri Oct  1 16:57:31 2010
From: xduvox at gmail.com (Octavio Duarte)
Date: Fri, 1 Oct 2010 18:57:31 -0500
Subject: [Freeswitch-users] distributor in LCR
Message-ID: <AANLkTin2CnjpVaWfuGhy19tAhrwRQKJs5d25D9umNNK5@mail.gmail.com>

Hello
Here is my problem, i have a several gateways and each one of them gw has a
different calls' capacity  but the same rate , what i wanna do is to use
something like mod distributor to be able to distribute call among gw?s
according it's capacity, i dont want to full each gateway to send calls to
another empty.
In LCR module i read that i can use random to get different order, but that
is not what i need! i see that i can add custom sql but  i wanted to know if
there is something already done!
Finally   I dont see the way to use mod limit and distributor together

i am really new on FS but i seems to be an amazing project!

Thank in advance for you help and time!
Best Regards from Mexico
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From peter.schrock at gmail.com  Fri Oct  1 19:43:58 2010
From: peter.schrock at gmail.com (Peter Schrock)
Date: Fri, 1 Oct 2010 19:43:58 -0700
Subject: [Freeswitch-users] error on install on os x 10.6.4
In-Reply-To: <AANLkTinxB3Rc3TxR0=BvBnhZ-cjXz02b9VxWY2k8D8WV@mail.gmail.com>
References: <AANLkTikb_MyGHMGet9682meMwbVXiJG_NeB6fQipYSr2@mail.gmail.com>
	<AANLkTi=W71voJZJDd=5stGJZX3_S7_d93Gune368+gBa@mail.gmail.com>
	<-2648041760610933328@unknownmsgid>
	<AANLkTinxB3Rc3TxR0=BvBnhZ-cjXz02b9VxWY2k8D8WV@mail.gmail.com>
Message-ID: <AANLkTi=-0-jJx3qTH+fG4w2ik4hwV1yvzqLGzCEyzoCk@mail.gmail.com>

Okay, so I have been toying around with a few idea that I have found and I
am having trouble with compiling spandsp. This is the error I am getting.

http://pastebin.freeswitch.org/14104

What is "movq"? I am thinking that gsm0610_rpe.lo is what is giving me
trouble, but I am not sure. Anyway, if anyone can help me understand what is
happening here and what I can do or look at to fix this. Mind you, this is
all from the current git tree.

Peter

On Thu, Sep 30, 2010 at 12:54 AM, Jeffrey Leung <curriegrad2004 at gmail.com>wrote:

> I checked out the scribd site, it's not too different from the wiki,
> except the fact that you'll need to run a ./bootstrap.sh before you
> start configuring and compiling for all git tree builds.
>
> On Thu, Sep 30, 2010 at 12:45 AM, Peter Schrock <peter.schrock at gmail.com>
> wrote:
> > Yeah, I have all the prerequisites.
> >
> > That's just it, I am not sure what else to post because it doesn't
> > really say much.
> >
> > I will try the git tree.
> >
> > Did you check out the scribd site? It will tell you what I did. Of
> > course, some things were updated since it was posted, so I modified
> > the instructions to those updates, like 1.0.6, not 1.0.1.
> >
> > PeterS
> >
> > On Sep 30, 2010, at 12:30 AM, Jeffrey Leung <curriegrad2004 at gmail.com>
> wrote:
> >
> >> Can you provide more details than that? I can't really see what's
> >> going on except make telling Error 2 has occurred. Are you sure you
> >> have all prerequisites installed on your Mac OS X machine?
> >>
> >> It'd be a great idea to clone the latest git tree and build it from
> >> there as there are many improvements since 1.0.1, including support
> >> for Google Voice via mod_dingaling.
> >>
> >> On Thu, Sep 30, 2010 at 12:09 AM, Peter Schrock <
> peter.schrock at gmail.com> wrote:
> >>> Since I was having troubles with getting my sip phone to connect, I
> decided
> >>> to try a different resource for instruction on how to install FS. The
> >>> instructions are found here:
> >>> http://www.scribd.com/doc/17425068/Free-Switch-in-Real-Life
> >>> Everything was going fine until I came up with this error on the make
> >>> install:
> >>> http://pastebin.freeswitch.org/14067
> >>> Any thoughts?
> >>> PeterS
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Fri Oct  1 20:42:36 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 1 Oct 2010 22:42:36 -0500
Subject: [Freeswitch-users] error on install on os x 10.6.4
In-Reply-To: <AANLkTi=-0-jJx3qTH+fG4w2ik4hwV1yvzqLGzCEyzoCk@mail.gmail.com>
References: <AANLkTikb_MyGHMGet9682meMwbVXiJG_NeB6fQipYSr2@mail.gmail.com>
	<AANLkTi=W71voJZJDd=5stGJZX3_S7_d93Gune368+gBa@mail.gmail.com>
	<-2648041760610933328@unknownmsgid>
	<AANLkTinxB3Rc3TxR0=BvBnhZ-cjXz02b9VxWY2k8D8WV@mail.gmail.com>
	<AANLkTi=-0-jJx3qTH+fG4w2ik4hwV1yvzqLGzCEyzoCk@mail.gmail.com>
Message-ID: <B08F4A6E-12DB-40FE-9739-1FAA61B606C5@freeswitch.org>

I'm going to only guess you have installed darwin ports or fink which has totally screwed your dev evn.  We only support building with the utils apple provides NOTHING MORE.

/b

On Oct 1, 2010, at 9:43 PM, Peter Schrock wrote:

> Okay, so I have been toying around with a few idea that I have found and I am having trouble with compiling spandsp. This is the error I am getting.
> 
> http://pastebin.freeswitch.org/14104
> 
> What is "movq"? I am thinking that gsm0610_rpe.lo is what is giving me trouble, but I am not sure. Anyway, if anyone can help me understand what is happening here and what I can do or look at to fix this. Mind you, this is all from the current git tree.
> 
> Peter
> 
> On Thu, Sep 30, 2010 at 12:54 AM, Jeffrey Leung <curriegrad2004 at gmail.com> wrote:
> I checked out the scribd site, it's not too different from the wiki,
> except the fact that you'll need to run a ./bootstrap.sh before you
> start configuring and compiling for all git tree builds.
> 
> On Thu, Sep 30, 2010 at 12:45 AM, Peter Schrock <peter.schrock at gmail.com> wrote:
> > Yeah, I have all the prerequisites.
> >
> > That's just it, I am not sure what else to post because it doesn't
> > really say much.
> >
> > I will try the git tree.
> >
> > Did you check out the scribd site? It will tell you what I did. Of
> > course, some things were updated since it was posted, so I modified
> > the instructions to those updates, like 1.0.6, not 1.0.1.
> >
> > PeterS
> >
> > On Sep 30, 2010, at 12:30 AM, Jeffrey Leung <curriegrad2004 at gmail.com> wrote:
> >
> >> Can you provide more details than that? I can't really see what's
> >> going on except make telling Error 2 has occurred. Are you sure you
> >> have all prerequisites installed on your Mac OS X machine?
> >>
> >> It'd be a great idea to clone the latest git tree and build it from
> >> there as there are many improvements since 1.0.1, including support
> >> for Google Voice via mod_dingaling.
> >>
> >> On Thu, Sep 30, 2010 at 12:09 AM, Peter Schrock <peter.schrock at gmail.com> wrote:
> >>> Since I was having troubles with getting my sip phone to connect, I decided
> >>> to try a different resource for instruction on how to install FS. The
> >>> instructions are found here:
> >>> http://www.scribd.com/doc/17425068/Free-Switch-in-Real-Life
> >>> Everything was going fine until I came up with this error on the make
> >>> install:
> >>> http://pastebin.freeswitch.org/14067
> >>> Any thoughts?
> >>> PeterS
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From bwibowo at gmail.com  Fri Oct  1 22:52:28 2010
From: bwibowo at gmail.com (Budi wibowo)
Date: Sat, 2 Oct 2010 05:52:28 +0000
Subject: [Freeswitch-users] mod skype legal
In-Reply-To: <AANLkTinF+-hUSTa=1FfMzjfodL5P8vowGTCLGiZ2DyOQ@mail.gmail.com>
References: <AANLkTi=Eha03JxMxYWZqyShCgvEw+g+AgyFJi3QJnXMz@mail.gmail.com><AANLkTinF+-hUSTa=1FfMzjfodL5P8vowGTCLGiZ2DyOQ@mail.gmail.com>
Message-ID: <732012231-1285998749-cardhu_decombobulator_blackberry.rim.net-1973678712-@bda034.bisx.prodap.on.blackberry>

Thx a lot for the positive feedback


Regards

Budi
-----Original Message-----
From: Shamun toha md <shamun.toha at gmail.com>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Thu, 30 Sep 2010 23:55:41 
To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] mod skype legal

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




From peter.schrock at gmail.com  Fri Oct  1 23:59:24 2010
From: peter.schrock at gmail.com (Peter Schrock)
Date: Fri, 1 Oct 2010 23:59:24 -0700
Subject: [Freeswitch-users] error on install on os x 10.6.4
In-Reply-To: <B08F4A6E-12DB-40FE-9739-1FAA61B606C5@freeswitch.org>
References: <AANLkTikb_MyGHMGet9682meMwbVXiJG_NeB6fQipYSr2@mail.gmail.com>
	<AANLkTi=W71voJZJDd=5stGJZX3_S7_d93Gune368+gBa@mail.gmail.com>
	<-2648041760610933328@unknownmsgid>
	<AANLkTinxB3Rc3TxR0=BvBnhZ-cjXz02b9VxWY2k8D8WV@mail.gmail.com>
	<AANLkTi=-0-jJx3qTH+fG4w2ik4hwV1yvzqLGzCEyzoCk@mail.gmail.com>
	<B08F4A6E-12DB-40FE-9739-1FAA61B606C5@freeswitch.org>
Message-ID: <-6194582537160853113@unknownmsgid>

And yet it says on this page:
http://wiki.freeswitch.org/wiki/Git_Install
to "Install MacPorts and run as root" to install git-core (to meet the
requirement of having git to build a git tree). As far as I know MacPorts is
formerly known as DarwinPorts.
If this is the case, why is it that the wiki is so keen on saying to make
sure the prerequisites are met and it isn't mentioned to avoid fink and
darwin? Those are suppose to be the equivalent of "yum" for Linux on the OS
X, why wouldn't I think to check those sources for any required
prerequisites?
This is aggravating!

On Oct 1, 2010, at 8:49 PM, Brian West <brian at freeswitch.org> wrote:

I'm going to only guess you have installed darwin ports or fink which has
totally screwed your dev evn.  We only support building with the utils apple
provides NOTHING MORE.

/b

On Oct 1, 2010, at 9:43 PM, Peter Schrock wrote:

Okay, so I have been toying around with a few idea that I have found and I
am having trouble with compiling spandsp. This is the error I am getting.

http://pastebin.freeswitch.org/14104

What is "movq"? I am thinking that gsm0610_rpe.lo is what is giving me
trouble, but I am not sure. Anyway, if anyone can help me understand what is
happening here and what I can do or look at to fix this. Mind you, this is
all from the current git tree.

Peter

On Thu, Sep 30, 2010 at 12:54 AM, Jeffrey Leung <curriegrad2004 at gmail.com>wrote:

> I checked out the scribd site, it's not too different from the wiki,
> except the fact that you'll need to run a ./bootstrap.sh before you
> start configuring and compiling for all git tree builds.
>
> On Thu, Sep 30, 2010 at 12:45 AM, Peter Schrock <peter.schrock at gmail.com>
> wrote:
> > Yeah, I have all the prerequisites.
> >
> > That's just it, I am not sure what else to post because it doesn't
> > really say much.
> >
> > I will try the git tree.
> >
> > Did you check out the scribd site? It will tell you what I did. Of
> > course, some things were updated since it was posted, so I modified
> > the instructions to those updates, like 1.0.6, not 1.0.1.
> >
> > PeterS
> >
> > On Sep 30, 2010, at 12:30 AM, Jeffrey Leung <curriegrad2004 at gmail.com>
> wrote:
> >
> >> Can you provide more details than that? I can't really see what's
> >> going on except make telling Error 2 has occurred. Are you sure you
> >> have all prerequisites installed on your Mac OS X machine?
> >>
> >> It'd be a great idea to clone the latest git tree and build it from
> >> there as there are many improvements since 1.0.1, including support
> >> for Google Voice via mod_dingaling.
> >>
> >> On Thu, Sep 30, 2010 at 12:09 AM, Peter Schrock <
> peter.schrock at gmail.com> wrote:
> >>> Since I was having troubles with getting my sip phone to connect, I
> decided
> >>> to try a different resource for instruction on how to install FS. The
> >>> instructions are found here:
> >>> http://www.scribd.com/doc/17425068/Free-Switch-in-Real-Life
> >>> Everything was going fine until I came up with this error on the make
> >>> install:
> >>> http://pastebin.freeswitch.org/14067
> >>> Any thoughts?
> >>> PeterS
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

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


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From shamun.toha at gmail.com  Sat Oct  2 02:11:15 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Sat, 2 Oct 2010 11:11:15 +0200
Subject: [Freeswitch-users] mod skype legal
In-Reply-To: <732012231-1285998749-cardhu_decombobulator_blackberry.rim.net-1973678712-@bda034.bisx.prodap.on.blackberry>
References: <AANLkTi=Eha03JxMxYWZqyShCgvEw+g+AgyFJi3QJnXMz@mail.gmail.com>
	<AANLkTinF+-hUSTa=1FfMzjfodL5P8vowGTCLGiZ2DyOQ@mail.gmail.com>
	<732012231-1285998749-cardhu_decombobulator_blackberry.rim.net-1973678712-@bda034.bisx.prodap.on.blackberry>
Message-ID: <AANLkTik0AZQrznDshpLVCQrgY1Jpc8Bd4_SMCX5QgQeN@mail.gmail.com>

You are welcome, I am trying to research for my thesis, and increase Skype
Instance to 100 or 200 or 300 Channels in 4 socket mother board with 12 core
on each. 12 core processor multiply by 4 = 48 core processors.

So, please keep growing this mod_skypopen, without any afraid, very soon, if
the ALSA driver bugs can be reduced, we can have like 500 Channels on small
CPU, because in my analyze i saw ALSA driver is causing many problems, not
the Skype API.

Thanks & Regards



On Sat, Oct 2, 2010 at 7:52 AM, Budi wibowo <bwibowo at gmail.com> wrote:

> Thx a lot for the positive feedback
>
>
> Regards
>
> Budi
> -----Original Message-----
> From: Shamun toha md <shamun.toha at gmail.com>
> Sender: freeswitch-users-bounces at lists.freeswitch.org
> Date: Thu, 30 Sep 2010 23:55:41
> To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] mod skype legal
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From babak.freeswitch at gmail.com  Sat Oct  2 03:50:33 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sat, 2 Oct 2010 14:20:33 +0330
Subject: [Freeswitch-users] user reachability
Message-ID: <AANLkTi=7gYjUcwWdqOsVvpm9ucTzxf0DeYVGgkdDm8sU@mail.gmail.com>

Hi
Is there anything like nat-options-ping and unregister-on-options-fail to
use for all registered users (not just users behind nat)? cause sometimes
the network connection between users and fs breaks and I don't want to send
calls to those users.
thanx
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From abid_freeswitch at live.com  Sat Oct  2 05:44:44 2010
From: abid_freeswitch at live.com (Abid Saleem)
Date: Sat, 2 Oct 2010 18:44:44 +0600
Subject: [Freeswitch-users] g729 passthrough mode
Message-ID: <COL109-W13224EAAAE107BFB3C60A1F46A0@phx.gbl>


Hi All,
Could you please tell me what is meant by 9729 codec in passthrough mode in the class 4 routing case. For example if Originating provider supports g729 and is sending call to FS which recieves and routes the call to termination provider who also supports g729 codec. So is the license required in this case or not?
Thanks for your time.
Regards--------------Abid SaleemSr. Product ManagerTerminus Technologies 		 	   		  
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From slim at thegreek.com  Fri Oct  1 14:01:31 2010
From: slim at thegreek.com (Jeroen C. van Gelderen)
Date: Fri, 1 Oct 2010 16:01:31 -0500
Subject: [Freeswitch-users] FreeTDM Partial Spans?
Message-ID: <8A7648CAD02A4BEBB8B4BF15A8733F37@mbnet.local>


Hi Guys,

This might be a dumb question but I can't seem to find an answer by
Googling.

I have oodles of FXO ports configured on Xorcom Astribanks. On the Astribank
each set of 8 FXO ports forms a zaptel/openzap/freetdm span.

Now if a trunk consists of an integral multiple of 8 FXO ports it is easy to
just dial out on the first available channel:

<action application="bridge" data="freetdm/1/a/$1|freetdm/2/a/$1"/>

This grabs the first available channel out of 16 FXO ports.

But what is the "right" way to do it when I need to use 1.5 spans (i.e. 12
FXO ports)?

Is there such a thing as a virtual span that can be built out of individual
FXO ports?

I have a feeling I'm missing something rather obvious.

Thanks,
-Slim
--
Jeroen C. "Slim" van Gelderen

Cheers,
-Slim
--
Jeroen C. "Slim" van Gelderen
Olympic Sports Data Services
Email: slim at thegreek.com
Phone: +1 876 953 6182 x128




From Nabble at slickdeals.endjunk.com  Fri Oct  1 12:23:15 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 1 Oct 2010 12:23:15 -0700 (PDT)
Subject: [Freeswitch-users] License of libzrtp for usage with FreeSWITCH?
In-Reply-To: <4CA5FEF8.5040102@infosecurity.ch>
References: <4CA53F77.5040106@puzzled.xs4all.nl>
	<C41DB9FA-0019-40EB-9CDF-0F3C3BA6276C@freeswitch.org>
	<4CA5FEF8.5040102@infosecurity.ch>
Message-ID: <1285960995535-5592596.post@n2.nabble.com>



Fabio Pietrosanti (naif) wrote:
> On http://zfoneproject.com/lic_policy.html it's written that the libzrtp
> is licensed under AGPL and only zfone it's not under AGPL.
Cool and thanks for the link. I believe this will be available for the next
releases of FS. This way, we all can do a secured communication and not
worry being evesdropped.

BTW, is this AGPL for libzrtp governed by US export law? If yes, then I
reckon it may not be a good idea to include in the next FS releases.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/License-of-libzrtp-for-usage-with-FreeSWITCH-tp5590066p5592596.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Fri Oct  1 07:17:46 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 1 Oct 2010 07:17:46 -0700 (PDT)
Subject: [Freeswitch-users] License of libzrtp for usage with FreeSWITCH?
In-Reply-To: <4CA53F77.5040106@puzzled.xs4all.nl>
References: <4CA53F77.5040106@puzzled.xs4all.nl>
Message-ID: <1285942666086-5591552.post@n2.nabble.com>



Patrick Lists wrote:
> ... I see a file docs/zrtp_agpl-3.0.txt 
> but the libzrtp source itself is not included.
Perhaps, the docs/zrtp_agpl-3.0.txt license is meant for the ZRTP sound
files included in the FS distro.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/License-of-libzrtp-for-usage-with-FreeSWITCH-tp5590066p5591552.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From justlikeef at gmail.com  Fri Oct  1 13:30:26 2010
From: justlikeef at gmail.com (Rob Hutton)
Date: Fri, 1 Oct 2010 16:30:26 -0400
Subject: [Freeswitch-users] Errors compiling mod_opal
Message-ID: <201010011630.26508.justlikeef@gmail.com>

OpenSuse 11.3 x86_64.  No clue where to start, willing to help...

Compiling 
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp...
quiet_libtool: compile:  g++ -g -ggdb -I. -
I/usr/src/freeswitch/freeswitch/src/include -
I/usr/src/freeswitch/freeswitch/src/include -
I/usr/src/freeswitch/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden 
-DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DP_64BIT=1 
-DPTRACING=1 -D_REENTRANT -D_GNU_SOURCE=1 -fno-exceptions -I/usr/include/opal 
-I/usr/include/SDL -DHAVE_CONFIG_H -c 
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp  -fPIC 
-DPIC -o .libs/mod_opal.o
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In 
constructor ?FSConnection::FSConnection(OpalCall&, FSEndPoint&, void*, 
unsigned int, OpalConnection::StringOptions*, switch_caller_profile_t*, 
switch_core_session_t*, switch_channel_t*)?:
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:564:26: 
error: no matching function for call to 
?OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, void*&, 
unsigned int&, OpalConnection::StringOptions*&)?
/usr/include/opal/opal/localep.h:249:5: note: candidates are: 
OpalLocalConnection::OpalLocalConnection(OpalCall&, OpalLocalEndPoint&, void*)
/usr/include/opal/opal/localep.h:242:1: note:                 
OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&)
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In 
member function ?switch_status_t FSMediaStream::read_frame(switch_frame_t**, 
switch_io_flag_t)?:
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1296:25: 
error: ?class OpalMediaPatch? has no member named ?OnStartMediaPatch?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In 
member function ?switch_status_t FSMediaStream::write_frame(const 
switch_frame_t*, switch_io_flag_t)?:
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1418:25: 
error: ?class OpalMediaPatch? has no member named ?OnStartMediaPatch?
make[5]: *** [mod_opal.lo] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_opal-all] Error 1
make[2]: *** [all-recursive] Error 1


-- 
Thanks,
Rob


From justlikeef at gmail.com  Fri Oct  1 09:58:41 2010
From: justlikeef at gmail.com (Rob Hutton)
Date: Fri, 1 Oct 2010 12:58:41 -0400
Subject: [Freeswitch-users] Errors compiling mod_h323
Message-ID: <201010011258.41592.justlikeef@gmail.com>

I am getting the following error compiling mod_h323 on OpenSuse 11.3 X86_64.  
I don't even know where to begin on this, but would me more than happy to help 
work it out...

Compiling 
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp...
quiet_libtool: compile:  g++ -g -ggdb -I/usr/local/include/ptlib -
I/usr/local/include/openh323 -I. -DPTRACING=1 -D_REENTRANT -fno-exceptions -
I/usr/src/freeswitch/freeswitch/src/include -
I/usr/src/freeswitch/freeswitch/src/include -
I/usr/src/freeswitch/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden 
-DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -
DHAVE_CONFIG_H -c 
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp  -fPIC 
-DPIC -o .libs/mod_h323.o
In file included from 
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:31:0:
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:110:1: 
error: too many initializers for ?const char* const [8]?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:121:1: 
error: too many initializers for ?const char* const [5]?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:212:40: 
error: expected class-name before ?{? token
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:213:2: 
error: ?Comparison? does not name a type
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h: In 
member function ?virtual BOOL FSProcess::InternalIsDescendant(const char*) 
const?:
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:213:2: 
error: ?PLibraryProcess? has not been declared
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h: In 
member function ?virtual const char* FSProcess::GetClass(unsigned int) const?:
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:213:2: 
error: ?PLibraryProcess? has not been declared
In file included from 
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:31:0:
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h: At 
global scope:
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:313:7: 
error: ?PBoolean? has not been declared
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:318:10: 
error: ?PBoolean? does not name a type
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:319:10: 
error: ?PBoolean? does not name a type
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:326:10: 
error: ?PBoolean? does not name a type
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:320:15: 
error: conflicting return type specified for ?virtual bool 
FSH323Connection::OnReceivedSignalSetup(const H323SignalPDU&)?
/usr/include/openh323/h323con.h:766:18: error:   overriding ?virtual BOOL 
H323Connection::OnReceivedSignalSetup(const H323SignalPDU&)?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:321:15: 
error: conflicting return type specified for ?virtual bool 
FSH323Connection::OnReceivedCallProceeding(const H323SignalPDU&)?
/usr/include/openh323/h323con.h:798:18: error:   overriding ?virtual BOOL 
H323Connection::OnReceivedCallProceeding(const H323SignalPDU&)?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:323:15: 
error: conflicting return type specified for ?virtual bool 
FSH323Connection::OnReceivedProgress(const H323SignalPDU&)?
/usr/include/openh323/h323con.h:810:18: error:   overriding ?virtual BOOL 
H323Connection::OnReceivedProgress(const H323SignalPDU&)?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:324:15: 
error: conflicting return type specified for ?virtual bool 
FSH323Connection::OnSendCallProceeding(H323SignalPDU&)?
/usr/include/openh323/h323con.h:1243:18: error:   overriding ?virtual BOOL 
H323Connection::OnSendCallProceeding(H323SignalPDU&)?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:325:15: 
error: conflicting return type specified for ?virtual bool 
FSH323Connection::OnSendReleaseComplete(H323SignalPDU&)?
/usr/include/openh323/h323con.h:1258:18: error:   overriding ?virtual BOOL 
H323Connection::OnSendReleaseComplete(H323SignalPDU&)?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:329:15: 
error: conflicting return type specified for ?virtual bool 
FSH323Connection::OnAlerting(const H323SignalPDU&, const PString&)?
/usr/include/openh323/h323con.h:1272:18: error:   overriding ?virtual BOOL 
H323Connection::OnAlerting(const H323SignalPDU&, const PString&)?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:333:15: 
error: conflicting return type specified for ?virtual bool 
FSH323Connection::OnRequestModeChange(const H245_RequestMode&, 
H245_RequestModeAck&, H245_RequestModeReject&, PINDEX&)?
/usr/include/openh323/h323con.h:2167:18: error:   overriding ?virtual BOOL 
H323Connection::OnRequestModeChange(const H245_RequestMode&, 
H245_RequestModeAck&, H245_RequestModeReject&, PINDEX&)?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:337:15: 
error: conflicting return type specified for ?virtual bool 
FSH323Connection::OnSendSignalSetup(H323SignalPDU&)?
/usr/include/openh323/h323con.h:1231:18: error:   overriding ?virtual BOOL 
H323Connection::OnSendSignalSetup(H323SignalPDU&)?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:341:15: 
error: conflicting return type specified for ?virtual bool 
FSH323Connection::OnReceivedCapabilitySet(const H323Capabilities&, const 
H245_MultiplexCapability*, H245_TerminalCapabilitySetReject&)?
/usr/include/openh323/h323con.h:1544:18: error:   overriding ?virtual BOOL 
H323Connection::OnReceivedCapabilitySet(const H323Capabilities&, const 
H245_MultiplexCapability*, H245_TerminalCapabilitySetReject&)?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:404:10: 
error: ?PBoolean? does not name a type
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:405:10: 
error: ?PBoolean? does not name a type
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:406:10: 
error: ?PBoolean? does not name a type
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:407:10: 
error: ?PBoolean? does not name a type
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:455:15: 
error: conflicting return type specified for ?virtual bool 
BaseG7231Capab::OnSendingPDU(H245_AudioCapability&, unsigned int) const?
/usr/include/openh323/h323caps.h:834:18: error:   overriding ?virtual BOOL 
H323AudioCapability::OnSendingPDU(H245_AudioCapability&, unsigned int) const?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:462:15: 
error: conflicting return type specified for ?virtual bool 
BaseG7231Capab::OnReceivedPDU(const H245_AudioCapability&, unsigned int&)?
/usr/include/openh323/h323caps.h:888:18: error:   overriding ?virtual BOOL 
H323AudioCapability::OnReceivedPDU(const H245_AudioCapability&, unsigned 
int&)?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:520:15: 
error: conflicting return type specified for ?virtual bool 
BaseGSM0610Cap::OnSendingPDU(H245_AudioCapability&, unsigned int) const?
/usr/include/openh323/h323caps.h:834:18: error:   overriding ?virtual BOOL 
H323AudioCapability::OnSendingPDU(H245_AudioCapability&, unsigned int) const?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:528:15: 
error: conflicting return type specified for ?virtual bool 
BaseGSM0610Cap::OnReceivedPDU(const H245_AudioCapability&, unsigned int&)?
/usr/include/openh323/h323caps.h:888:18: error:   overriding ?virtual BOOL 
H323AudioCapability::OnReceivedPDU(const H245_AudioCapability&, unsigned 
int&)?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp: In 
function ?const OpalMediaFormat& GetOpalT38_IFP_COR()?:
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:84:6: 
error: invalid conversion from ?char*? to ?BOOL?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:84:6: 
error:   initializing argument 4 of ?OpalMediaFormat::OpalMediaFormat(const 
char*, unsigned int, RTP_DataFrame::PayloadTypes, BOOL, unsigned int, PINDEX, 
unsigned int, unsigned int, time_t)?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp: In 
function ?const OpalMediaFormat& GetOpalT38_IFP_PRE()?:
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:100:6: 
error: invalid conversion from ?char*? to ?BOOL?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:100:6: 
error:   initializing argument 4 of ?OpalMediaFormat::OpalMediaFormat(const 
char*, unsigned int, RTP_DataFrame::PayloadTypes, BOOL, unsigned int, PINDEX, 
unsigned int, unsigned int, time_t)?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp: In 
constructor ?FSProcess::FSProcess()?:
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:346:4: 
error: class ?FSProcess? does not have any field named ?PLibraryProcess?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:346:46: 
error: ?AlphaCode? was not declared in this scope
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp: At 
global scope:
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:752:42: 
error: ?PBoolean? has not been declared
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:961:1: 
error: ?PBoolean? does not name a type
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:968:1: 
error: ?PBoolean? does not name a type
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:1083:1: 
error: ?PBoolean? does not name a type
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp: In 
member function ?virtual void 
FSH323Connection::AnsweringCall(H323Connection::AnswerCallResponse)?:
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:1152:5: 
error: ?PBoolean? was not declared in this scope
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:1152:14: 
error: expected ?;? before ?sendPDU?
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:1187:9: 
error: ?sendPDU? was not declared in this scope
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp: At 
global scope:
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:1242:1: 
error: ?PBoolean? does not name a type
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:1904:1: 
error: ?PBoolean? does not name a type
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:2213:1: 
error: ?PBoolean? does not name a type
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:2227:1: 
error: ?PBoolean? does not name a type
/usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:2233:1: 
error: ?PBoolean? does not name a type
make[1]: *** [mod_h323.lo] Error 1
make: *** [all] Error 1


-- 
Thanks,
Rob


From justlikeef at gmail.com  Fri Oct  1 09:31:52 2010
From: justlikeef at gmail.com (Rob Hutton)
Date: Fri, 1 Oct 2010 12:31:52 -0400
Subject: [Freeswitch-users] Error compiling mod_callcenter
Message-ID: <201010011231.52583.justlikeef@gmail.com>

When trying to compile mod_callcenter I am getting the following error:

making all mod_callcenter
Compiling 
/usr/src/freeswitch/freeswitch/src/mod/applications/mod_callcenter/mod_callcenter.c...
quiet_libtool: compile:  gcc -I/usr/src/freeswitch/freeswitch/src/include -
I/usr/src/freeswitch/freeswitch/src/include -
I/usr/src/freeswitch/freeswitch/libs/libteletone/src -fPIC -Werror -
fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -g -
O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -
DHAVE_CONFIG_H -c 
/usr/src/freeswitch/freeswitch/src/mod/applications/mod_callcenter/mod_callcenter.c  
-fPIC -DPIC -o .libs/mod_callcenter.o
cc1: warnings being treated as errors
/usr/src/freeswitch/freeswitch/src/mod/applications/mod_callcenter/mod_callcenter.c: 
In function ?cc_config_api_function?:
/usr/src/freeswitch/freeswitch/src/mod/applications/mod_callcenter/mod_callcenter.c:2466:6: 
error: case value ?11? not in enumerated type ?cc_agent_status_t?
make[5]: *** [mod_callcenter.lo] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_callcenter-all] Error 1
make[2]: *** [all-recursive] Error 1

According to cc_agent_status_t; 0, 1, 2, 3, and 4 are valid.  

The switch statement this refers to tests for CC_STATUS_INVALID_KEY which is 
defined in cc_status_t as the numeric value 11.

The switch statement calls cc_agent_get which defines the return value as 
cc_agent_status_t.   In looking at that function, it seems that the return 
values referenced all come from cc_status_t, so I think just changing the 
defined returned value fixes the issue, but there are some select statements 
that I am not sure what the possible values are.

-- 
Thanks,
Rob


From mnhassan at usa.net  Sat Oct  2 05:55:40 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Sat, 2 Oct 2010 18:55:40 +0600
Subject: [Freeswitch-users] g729 passthrough mode
In-Reply-To: <COL109-W13224EAAAE107BFB3C60A1F46A0@phx.gbl>
References: <COL109-W13224EAAAE107BFB3C60A1F46A0@phx.gbl>
Message-ID: <AANLkTimh+NiKS_q60Un7AEEPrRq_AKQ+feUQWME+T8JS@mail.gmail.com>

No.


On Sat, Oct 2, 2010 at 18:44, Abid Saleem <abid_freeswitch at live.com> wrote:

>  Hi All,
>
> Could you please tell me what is meant by 9729 codec in passthrough mode in
> the class 4 routing case. For example if Originating provider supports g729
> and is sending call to FS which recieves and routes the call to termination
> provider who also supports g729 codec. So is the license required in this
> case or not?
>
> Thanks for your time.
>
> Regards
> --------------
> Abid Saleem
> Sr. Product Manager
> Terminus Technologies
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From testeador01 at gmail.com  Sat Oct  2 06:09:04 2010
From: testeador01 at gmail.com (Milena)
Date: Sat, 2 Oct 2010 08:09:04 -0500
Subject: [Freeswitch-users] Multiple "voices" under mod_say_en
In-Reply-To: <AANLkTinD61mjiCsnODpqfQqzib-YLz553rHjDwFoNym6@mail.gmail.com>
References: <787302A89ACCE24DA8F56DA101E77C842B473C0A1B@THHS2E12BE1X.hostedservice2.net>
	<AANLkTinD61mjiCsnODpqfQqzib-YLz553rHjDwFoNym6@mail.gmail.com>
Message-ID: <AANLkTikNfDrPUpt1Uba9nZ0-2=DSM7sKCqE5ok35Azui@mail.gmail.com>

Hi,
you forgot to read the wiki:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_intercept

<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_intercept>(last
example)

-Milena

On Fri, Oct 1, 2010 at 2:05 PM, Michael Collins <msc at freeswitch.org> wrote:

> Neil,
>
> Try setting channel variable ${sound_prefix} prior to calling your say
> application. It's an imperfect solution until we have something more elegant
> in place. Just note that if you need to switch between US and UK sounds on
> the same call you'll need to change the sound_prefix on the channel each
> time.
>
> The default sound_prefix is:
> /usr/local/freeswitch/sounds/en/us/callie
>
> So change it to:
> /usr/local/freeswitch/sounds/en/uk/<your_voice>
>
> And then call say and it should use your uk voice.
>
> -MC
>
> On Fri, Oct 1, 2010 at 8:32 AM, Neil Burgess <neil.burgess at redmatter.com>wrote:
>
>>  Hi,
>>
>>
>>
>> I?d like to be able to use two sets of language files for US and UK within
>> the say module, allowing either to be utilised with the spell, items, etc
>> capabilities, based on setting some channel variable!
>>
>>
>>
>> At the moment, I can?t see how to achieve this since it appears that the
>> ?say_en? module for manipulating these strings is linked to the language
>> name attribute as below.
>>
>>
>>
>> <language name="en" sound-path="$${base_dir}/sounds/en/us/callie"
>> tts-engine="unimrcp:loquendo7-mrcp2" tts-voice="kate">
>>
>>
>>
>> Is there any way of supporting multiple ?voices? under the same mod_say_en
>>
>>
>>
>> Thanks,
>>
>> Neil
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From testeador01 at gmail.com  Sat Oct  2 06:39:43 2010
From: testeador01 at gmail.com (Milena)
Date: Sat, 2 Oct 2010 08:39:43 -0500
Subject: [Freeswitch-users] Multiple "voices" under mod_say_en
In-Reply-To: <AANLkTikNfDrPUpt1Uba9nZ0-2=DSM7sKCqE5ok35Azui@mail.gmail.com>
References: <787302A89ACCE24DA8F56DA101E77C842B473C0A1B@THHS2E12BE1X.hostedservice2.net>
	<AANLkTinD61mjiCsnODpqfQqzib-YLz553rHjDwFoNym6@mail.gmail.com>
	<AANLkTikNfDrPUpt1Uba9nZ0-2=DSM7sKCqE5ok35Azui@mail.gmail.com>
Message-ID: <AANLkTinp0+mjS06S6TUZJtEY3DF-OhwturVmB-y4J55p@mail.gmail.com>

wrong thread for my previous email :)

about this topic:
you can also create a whole language called 'en_UK' based on 'en' from
conf/lang
and set the default language (default_language=en_UK)
or (default_language=en).
This would allow you to have different phrases defined for the language
according to user preferences.
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From testeador01 at gmail.com  Sat Oct  2 06:40:45 2010
From: testeador01 at gmail.com (Milena)
Date: Sat, 2 Oct 2010 08:40:45 -0500
Subject: [Freeswitch-users] Intercept Ringing Call
In-Reply-To: <AANLkTim1-+x27rZwZ33YRPAvDPtJc3sa7braw58S7nHC@mail.gmail.com>
References: <009801cb617c$763d7810$62b86830$@teotech.com>
	<AANLkTim1-+x27rZwZ33YRPAvDPtJc3sa7braw58S7nHC@mail.gmail.com>
Message-ID: <AANLkTikL4HWhw-k7zvkZ2BuGJO9iHhi+v44Muc-0NS4v@mail.gmail.com>

Hi,
you forgot to read the wiki:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_intercept

<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_intercept>(last
example)

-Milena

On Fri, Oct 1, 2010 at 8:22 AM, Jerry Richards
<jerry.richards at teotech.com>wrote:
>
>> Can I intercept a call in ringing state only?
>
>
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From saeedahmad1981 at gmail.com  Sat Oct  2 07:32:00 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Sat, 2 Oct 2010 16:32:00 +0200
Subject: [Freeswitch-users] mod skype legal
In-Reply-To: <AANLkTik0AZQrznDshpLVCQrgY1Jpc8Bd4_SMCX5QgQeN@mail.gmail.com>
References: <AANLkTi=Eha03JxMxYWZqyShCgvEw+g+AgyFJi3QJnXMz@mail.gmail.com>
	<AANLkTinF+-hUSTa=1FfMzjfodL5P8vowGTCLGiZ2DyOQ@mail.gmail.com>
	<732012231-1285998749-cardhu_decombobulator_blackberry.rim.net-1973678712-@bda034.bisx.prodap.on.blackberry>
	<AANLkTik0AZQrznDshpLVCQrgY1Jpc8Bd4_SMCX5QgQeN@mail.gmail.com>
Message-ID: <AANLkTimCm9h+1FoendyAmUiyG0DehXURip_LCLyghUQZ@mail.gmail.com>

Hi,

Recently we saw that skype blocked fring and nimbuzz... could it also be the
case that they see calls from FS and block them?


On Sat, Oct 2, 2010 at 11:11 AM, Shamun toha md <shamun.toha at gmail.com>wrote:

> You are welcome, I am trying to research for my thesis, and increase Skype
> Instance to 100 or 200 or 300 Channels in 4 socket mother board with 12 core
> on each. 12 core processor multiply by 4 = 48 core processors.
>
> So, please keep growing this mod_skypopen, without any afraid, very soon,
> if the ALSA driver bugs can be reduced, we can have like 500 Channels on
> small CPU, because in my analyze i saw ALSA driver is causing many problems,
> not the Skype API.
>
> Thanks & Regards
>
>
>
>
> On Sat, Oct 2, 2010 at 7:52 AM, Budi wibowo <bwibowo at gmail.com> wrote:
>
>> Thx a lot for the positive feedback
>>
>>
>> Regards
>>
>> Budi
>> -----Original Message-----
>> From: Shamun toha md <shamun.toha at gmail.com>
>> Sender: freeswitch-users-bounces at lists.freeswitch.org
>> Date: Thu, 30 Sep 2010 23:55:41
>> To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
>> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-users] mod skype legal
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Sat Oct  2 07:40:09 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 2 Oct 2010 09:40:09 -0500
Subject: [Freeswitch-users] error on install on os x 10.6.4
In-Reply-To: <-6194582537160853113@unknownmsgid>
References: <AANLkTikb_MyGHMGet9682meMwbVXiJG_NeB6fQipYSr2@mail.gmail.com>
	<AANLkTi=W71voJZJDd=5stGJZX3_S7_d93Gune368+gBa@mail.gmail.com>
	<-2648041760610933328@unknownmsgid>
	<AANLkTinxB3Rc3TxR0=BvBnhZ-cjXz02b9VxWY2k8D8WV@mail.gmail.com>
	<AANLkTi=-0-jJx3qTH+fG4w2ik4hwV1yvzqLGzCEyzoCk@mail.gmail.com>
	<B08F4A6E-12DB-40FE-9739-1FAA61B606C5@freeswitch.org>
	<-6194582537160853113@unknownmsgid>
Message-ID: <4048EE03-F887-4D0C-A8A9-A72672164CFF@freeswitch.org>

That should be fixed.  Installing any of that will destroy your build environment.   See this is a community wiki... so someone put that on there.  It should be removed and clarified. 

Default build tools plus http://code.google.com/p/git-osx-installer/ are all you need.

/b

On Oct 2, 2010, at 1:59 AM, Peter Schrock wrote:

> And yet it says on this page:
> http://wiki.freeswitch.org/wiki/Git_Install
> to "Install MacPorts and run as root" to install git-core (to meet the requirement of having git to build a git tree). As far as I know MacPorts is formerly known as DarwinPorts. 
> If this is the case, why is it that the wiki is so keen on saying to make sure the prerequisites are met and it isn't mentioned to avoid fink and darwin? Those are suppose to be the equivalent of "yum" for Linux on the OS X, why wouldn't I think to check those sources for any required prerequisites?
> This is aggravating!
> 
> On Oct 1, 2010, at 8:49 PM, Brian West <brian at freeswitch.org> wrote:
> 
>> I'm going to only guess you have installed darwin ports or fink which has totally screwed your dev evn.  We only support building with the utils apple provides NOTHING MORE.
>> 
>> /b

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From freeswitch-list at puzzled.xs4all.nl  Sat Oct  2 07:44:45 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Sat, 02 Oct 2010 16:44:45 +0200
Subject: [Freeswitch-users] Errors compiling mod_opal
In-Reply-To: <201010011630.26508.justlikeef@gmail.com>
References: <201010011630.26508.justlikeef@gmail.com>
Message-ID: <4CA7455D.8080808@puzzled.xs4all.nl>

On 10/01/2010 10:30 PM, Rob Hutton wrote:
> OpenSuse 11.3 x86_64.  No clue where to start, willing to help...

Afaik you need to have the following installed (also the -devel packages):

For mod_opal:
ptlib version 2.8.1 (or 2.8.2)
opal  version 3.8.1 (or 3.8.2)

For mod_h323:
ptlib version 2.8.1 (or 2.8.2)
h323+ version cvs from 20100525

Regards,
Patrick


From brian at freeswitch.org  Sat Oct  2 07:46:07 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 2 Oct 2010 09:46:07 -0500
Subject: [Freeswitch-users] Error compiling mod_callcenter
In-Reply-To: <201010011231.52583.justlikeef@gmail.com>
References: <201010011231.52583.justlikeef@gmail.com>
Message-ID: <E238BBAD-B5D7-47B2-A40D-098BA465C955@freeswitch.org>

Can you EMAIL only once please... If you have compile issues with modules that are NOT in the default modules.conf that is installed then please contact the authors of the modules listed at the tops of those files.  We don't maintain those and if they will not then removal of the modules is our only corse of action.  As for the h323 modules you're going to have to compile the opan and h323 libs from source... packages on SUSE will not do.

/b

On Oct 1, 2010, at 11:31 AM, Rob Hutton wrote:

> When trying to compile mod_callcenter I am getting the following error:
> 
> making all mod_callcenter
> Compiling 
> /usr/src/freeswitch/freeswitch/src/mod/applications/mod_callcenter/mod_callcenter.c...
> quiet_libtool: compile:  gcc -I/usr/src/freeswitch/freeswitch/src/include -
> I/usr/src/freeswitch/freeswitch/src/include -
> I/usr/src/freeswitch/freeswitch/libs/libteletone/src -fPIC -Werror -
> fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -g -
> O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -
> DHAVE_CONFIG_H -c 
> /usr/src/freeswitch/freeswitch/src/mod/applications/mod_callcenter/mod_callcenter.c  
> -fPIC -DPIC -o .libs/mod_callcenter.o
> cc1: warnings being treated as errors
> /usr/src/freeswitch/freeswitch/src/mod/applications/mod_callcenter/mod_callcenter.c: 
> In function ?cc_config_api_function?:
> /usr/src/freeswitch/freeswitch/src/mod/applications/mod_callcenter/mod_callcenter.c:2466:6: 
> error: case value ?11? not in enumerated type ?cc_agent_status_t?
> make[5]: *** [mod_callcenter.lo] Error 1
> make[4]: *** [all] Error 1
> make[3]: *** [mod_callcenter-all] Error 1
> make[2]: *** [all-recursive] Error 1
> 
> According to cc_agent_status_t; 0, 1, 2, 3, and 4 are valid.  
> 
> The switch statement this refers to tests for CC_STATUS_INVALID_KEY which is 
> defined in cc_status_t as the numeric value 11.
> 
> The switch statement calls cc_agent_get which defines the return value as 
> cc_agent_status_t.   In looking at that function, it seems that the return 
> values referenced all come from cc_status_t, so I think just changing the 
> defined returned value fixes the issue, but there are some select statements 
> that I am not sure what the possible values are.
> 
> -- 
> Thanks,
> Rob
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Sat Oct  2 07:46:41 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 2 Oct 2010 09:46:41 -0500
Subject: [Freeswitch-users] Intercept Ringing Call
In-Reply-To: <AANLkTikL4HWhw-k7zvkZ2BuGJO9iHhi+v44Muc-0NS4v@mail.gmail.com>
References: <009801cb617c$763d7810$62b86830$@teotech.com>
	<AANLkTim1-+x27rZwZ33YRPAvDPtJc3sa7braw58S7nHC@mail.gmail.com>
	<AANLkTikL4HWhw-k7zvkZ2BuGJO9iHhi+v44Muc-0NS4v@mail.gmail.com>
Message-ID: <A7814BB2-FB31-4ACB-A4A0-56CB3503BEB3@freeswitch.org>

could just be the wiki search sucks :P

/b

On Oct 2, 2010, at 8:40 AM, Milena wrote:

> Hi, 
> you forgot to read the wiki:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_intercept
> 
> (last example)
> 
> -Milena

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From gmaruzz at gmail.com  Sat Oct  2 07:49:34 2010
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 2 Oct 2010 16:49:34 +0200
Subject: [Freeswitch-users] mod skype legal
In-Reply-To: <AANLkTimCm9h+1FoendyAmUiyG0DehXURip_LCLyghUQZ@mail.gmail.com>
References: <AANLkTi=Eha03JxMxYWZqyShCgvEw+g+AgyFJi3QJnXMz@mail.gmail.com>
	<AANLkTinF+-hUSTa=1FfMzjfodL5P8vowGTCLGiZ2DyOQ@mail.gmail.com>
	<732012231-1285998749-cardhu_decombobulator_blackberry.rim.net-1973678712-@bda034.bisx.prodap.on.blackberry>
	<AANLkTik0AZQrznDshpLVCQrgY1Jpc8Bd4_SMCX5QgQeN@mail.gmail.com>
	<AANLkTimCm9h+1FoendyAmUiyG0DehXURip_LCLyghUQZ@mail.gmail.com>
Message-ID: <AANLkTinTr=QmqHq1hOwev4AKDK0rbmVA4gws+eP1N+KC@mail.gmail.com>

No way, it's just not possible (I believe).

Mod-skypopen generates regular skype calls using the standard skype
client (mod-skypopen acts like a remote control on the standard skype
client).
Those calls are "true" skype calls, in no way different from other
regular skype calls.
-giovanni

On 10/2/10, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> Hi,
>
> Recently we saw that skype blocked fring and nimbuzz... could it also be the
> case that they see calls from FS and block them?
>
>
> On Sat, Oct 2, 2010 at 11:11 AM, Shamun toha md
> <shamun.toha at gmail.com>wrote:
>
>> You are welcome, I am trying to research for my thesis, and increase Skype
>> Instance to 100 or 200 or 300 Channels in 4 socket mother board with 12
>> core
>> on each. 12 core processor multiply by 4 = 48 core processors.
>>
>> So, please keep growing this mod_skypopen, without any afraid, very soon,
>> if the ALSA driver bugs can be reduced, we can have like 500 Channels on
>> small CPU, because in my analyze i saw ALSA driver is causing many
>> problems,
>> not the Skype API.
>>
>> Thanks & Regards
>>
>>
>>
>>
>> On Sat, Oct 2, 2010 at 7:52 AM, Budi wibowo <bwibowo at gmail.com> wrote:
>>
>>> Thx a lot for the positive feedback
>>>
>>>
>>> Regards
>>>
>>> Budi
>>> -----Original Message-----
>>> From: Shamun toha md <shamun.toha at gmail.com>
>>> Sender: freeswitch-users-bounces at lists.freeswitch.org
>>> Date: Thu, 30 Sep 2010 23:55:41
>>> To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
>>> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Subject: Re: [Freeswitch-users] mod skype legal
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From moises.silva at gmail.com  Sat Oct  2 08:06:23 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Sat, 2 Oct 2010 11:06:23 -0400
Subject: [Freeswitch-users] FreeTDM Partial Spans?
In-Reply-To: <8A7648CAD02A4BEBB8B4BF15A8733F37@mbnet.local>
References: <8A7648CAD02A4BEBB8B4BF15A8733F37@mbnet.local>
Message-ID: <AANLkTinK+1i4s+tVqLMua6aNd_u+M9Fdx9_5pYxcnUrm@mail.gmail.com>

On Fri, Oct 1, 2010 at 5:01 PM, Jeroen C. van Gelderen
<slim at thegreek.com> wrote:
> <action application="bridge" data="freetdm/1/a/$1|freetdm/2/a/$1"/>
>
> This grabs the first available channel out of 16 FXO ports.
>
> But what is the "right" way to do it when I need to use 1.5 spans (i.e. 12
> FXO ports)?
>
> Is there such a thing as a virtual span that can be built out of individual
> FXO ports?
>
> I have a feeling I'm missing something rather obvious.

You can create spans with channels from any other span (as long as the
signaling is the same).

[span zt xorcomSpan]
; channels from 1.5 trunks
fxo-channel => 1-12

Then dial:

<action application="bridge" data="freetdm/xorcomSpan/a/$1/>

You can also just define a group for those channels and use the group
for dialing.

[span zt xorcomTrunk1]
trunk_type => FXO
group => xorcomg1
fxo-channels => 1-8

[span zt xorcomTrunk2]
trunk_type => FXO
group => xorcomg1
fxo-channels => 9-12

Then dial using the group name.

<action application="bridge" data="freetdm/xorcomg1/a/$1/>

This means spans and groups are strings within the same name space (as
far as hunting is concerned), and you should not make them conflict.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From b_ball_henry at hotmail.com  Sat Oct  2 08:20:10 2010
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Sat, 2 Oct 2010 23:20:10 +0800
Subject: [Freeswitch-users] Intercept Ringing Call
In-Reply-To: <A7814BB2-FB31-4ACB-A4A0-56CB3503BEB3@freeswitch.org>
References: <009801cb617c$763d7810$62b86830$@teotech.com>
	<AANLkTim1-+x27rZwZ33YRPAvDPtJc3sa7braw58S7nHC@mail.gmail.com>
	<AANLkTikL4HWhw-k7zvkZ2BuGJO9iHhi+v44Muc-0NS4v@mail.gmail.com>
	<A7814BB2-FB31-4ACB-A4A0-56CB3503BEB3@freeswitch.org>
Message-ID: <AANLkTin4F2=shvUf=yWHUViiaMPde_dUpFa7FCia+w0=@mail.gmail.com>

Possibly, the wiki search takes exact terms. One have to be precise and know
the exact term that they are searching for.

Henry Huang Unified Communication System R&D, Founder of UniC Solution
US: +1 (626) 606-3306 | ??(Taiwan): +886 933847619
Contact Me [image:
LinkedIn]<http://www.linkedin.com/pub/henry-huang/1/654/578> [image:
Facebook] <http://www.facebook.com/profile.php?id=595148342> [image:
Twitter] <http://twitter.com/unicsolution>
IM [image: Google Talk/] red_rain_seven at gmail.com [image:
Skype/]unicsolution [image:
MSN/] b_ball_henry at hotmail.com
?"My play was a complete success. The audience was a failure." - Ashleigh
Brilliant<http://www.quotesdaddy.com/quote/59255/ashleigh-brilliant/my-play-was-a-complete-success-the-audience-was-a>
?  Get this email app!
<http://www.wisestamp.com/apps/quotes?utm_source=extension&utm_medium=email&utm_term=quotes&utm_campaign=apps>



On Sat, Oct 2, 2010 at 10:46 PM, Brian West <brian at freeswitch.org> wrote:

> could just be the wiki search sucks :P
>
> /b
>
> On Oct 2, 2010, at 8:40 AM, Milena wrote:
>
> Hi,
> you forgot to read the wiki:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_intercept
>
> <http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_intercept>(last
> example)
>
> -Milena
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From moises.silva at gmail.com  Sat Oct  2 08:34:52 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Sat, 2 Oct 2010 11:34:52 -0400
Subject: [Freeswitch-users] FreeTDM Partial Spans?
In-Reply-To: <AANLkTinK+1i4s+tVqLMua6aNd_u+M9Fdx9_5pYxcnUrm@mail.gmail.com>
References: <8A7648CAD02A4BEBB8B4BF15A8733F37@mbnet.local>
	<AANLkTinK+1i4s+tVqLMua6aNd_u+M9Fdx9_5pYxcnUrm@mail.gmail.com>
Message-ID: <AANLkTinAp=RSSLqPi16gejMoGBihg7YnnB2wj0kz7vJC@mail.gmail.com>

Btw, I hope you don't mind, I started a Mailing list FAQ section in
the FreeTDM wiki page and added your question there (and some more
information):

http://wiki.freeswitch.org/wiki/FreeTDM#Mailing_List_FAQ

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From moises.silva at gmail.com  Sat Oct  2 08:36:53 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Sat, 2 Oct 2010 11:36:53 -0400
Subject: [Freeswitch-users] Openzap rxgain and txgain not working well
In-Reply-To: <5e91d1a8d2e95fe3823da1883f958bb6@thom.fr.eu.org>
References: <AANLkTik=wrCRqKdbKkSeC4jv+uXSaHXgK+HkkYLRxAUH@mail.gmail.com>
	<AANLkTi=8eaUEKW+f-ThpW1LxRjeEg_s_c_d4_+vVESXD@mail.gmail.com>
	<AANLkTim0=-DEYxZpqWr++ts=QWMvBkP3g+R3T5QBmi=n@mail.gmail.com>
	<24986.1285340573@ccs.covici.com>
	<AANLkTinrebwEw4SP=VBuwdWkf3diPyEEDUgX8b0QV18x@mail.gmail.com>
	<6e8db087d1eb31e6d11e4b2aa2af6220@thom.fr.eu.org>
	<AANLkTimb6d_AuEhdniBpKoa3PiRBJBkqsBmSrpXp6+yC@mail.gmail.com>
	<5e91d1a8d2e95fe3823da1883f958bb6@thom.fr.eu.org>
Message-ID: <AANLkTim8xR0GFDfV_TtDpzDMkur4NDOT-eaxL=k4KxhF@mail.gmail.com>

On Wed, Sep 29, 2010 at 3:04 AM, Fran?ois Legal <devel at thom.fr.eu.org> wrote:
> Unfortunately, there is no (not yet ?) exact link. The page
> "Configuration examples" from Openzap would need to be ported/reworked
> for freetdm, then the link could be added on the freetdm page.
>
> Fran?ois

I worked a bit on that. Please help adding any new information at
http://wiki.freeswitch.org/wiki/FreeTDM

I added a mailing list FAQ too. It's supposed to include any frequent
question with an answer in the mailing list in a synthesized form.

http://wiki.freeswitch.org/wiki/FreeTDM#Mailing_List_FAQ

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From testeador01 at gmail.com  Sat Oct  2 08:56:43 2010
From: testeador01 at gmail.com (Milena)
Date: Sat, 2 Oct 2010 10:56:43 -0500
Subject: [Freeswitch-users] Intercept Ringing Call
In-Reply-To: <AANLkTin4F2=shvUf=yWHUViiaMPde_dUpFa7FCia+w0=@mail.gmail.com>
References: <009801cb617c$763d7810$62b86830$@teotech.com>
	<AANLkTim1-+x27rZwZ33YRPAvDPtJc3sa7braw58S7nHC@mail.gmail.com>
	<AANLkTikL4HWhw-k7zvkZ2BuGJO9iHhi+v44Muc-0NS4v@mail.gmail.com>
	<A7814BB2-FB31-4ACB-A4A0-56CB3503BEB3@freeswitch.org>
	<AANLkTin4F2=shvUf=yWHUViiaMPde_dUpFa7FCia+w0=@mail.gmail.com>
Message-ID: <AANLkTimRazMC_-KaRjPg3S3NBRdSPrS9rXAs=nPySsfq@mail.gmail.com>

Aww Brian!! give me a break!!! what really sucks it that nobody uses google
...maybe we should put this thing at the wiki faqs:

* HOW TO SEARCH ON THE WIKI: *
On your browser go to:
       www.google.com
On the search box type:
       <write your search terms in here> site:wiki.freeswitch.org

voil?!!! :D

What do you think? ;)

-Milena
On Sat, Oct 2, 2010 at 10:20 AM, Henry Huang <b_ball_henry at hotmail.com>wrote:

> Possibly, the wiki search takes exact terms. One have to be precise and
> know the exact term that they are searching for.
>
> On Sat, Oct 2, 2010 at 10:46 PM, Brian West <brian at freeswitch.org> wrote:
>
>> could just be the wiki search sucks :P
>>
>
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From rupa at rupa.com  Sat Oct  2 10:45:18 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 2 Oct 2010 12:45:18 -0500
Subject: [Freeswitch-users] distributor in LCR
In-Reply-To: <AANLkTin2CnjpVaWfuGhy19tAhrwRQKJs5d25D9umNNK5@mail.gmail.com>
References: <AANLkTin2CnjpVaWfuGhy19tAhrwRQKJs5d25D9umNNK5@mail.gmail.com>
Message-ID: <AANLkTikzJcKXP-+QO0N0X=uz+P=5qLRChQFRNA5DNfHy@mail.gmail.com>

On Fri, Oct 1, 2010 at 6:57 PM, Octavio Duarte <xduvox at gmail.com> wrote:

> Hello
> Here is my problem, i have a several gateways and each one of them gw has a
> different calls' capacity  but the same rate , what i wanna do is to use
> something like mod distributor to be able to distribute call among gw?s
> according it's capacity, i dont want to full each gateway to send calls to
> another empty.
> In LCR module i read that i can use random to get different order, but that
> is not what i need! i see that i can add custom sql but  i wanted to know if
> there is something already done!
>

Since I maintain mod_lcr, I'd suggest that using mod_lcr with custom_sql and
proper use of random is the "right" way to do it.  I haven't looked too
closely at mod_distributor, but maybe you can have lcr return you the list
of "logical gatways" that are each described by a distributor endpoint.
 Then iterate through the list of routes, expanding as needed.  Yes, it is
more work than just using custom_sql or distributor directly but may get you
what you want.

Already done?  FS is a construction kit.  Use the tools to build what you
need.

Maybe if you describe what you need in total we can suggest a reasonable
solution?


> Finally   I dont see the way to use mod limit and distributor together
>
>
Correct.  I have an open ticket on Jira to research it.  So far I haven't
come up with an approach that I'm happy with.  If someone wants to suggest
an approach we can look at it and see if it works and if so implement.
 Patches or just ideas are welcome. Personally, I just suggest using mod_lcr
to solve both the lcr issues and the distribution issues.


> i am really new on FS but i seems to be an amazing project!
>
>
That it is.  FS is very flexible.  That means that some parts can
be overwhelming.   The list is helpful and the IRC channel is *very*
helpful.


> Thank in advance for you help and time!
> Best Regards from Mexico
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From lists at infosecurity.ch  Sat Oct  2 11:22:05 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Sat, 02 Oct 2010 20:22:05 +0200
Subject: [Freeswitch-users] Can FreeSWITCH act as a turn server for ICE?
Message-ID: <4CA7784D.9050904@infosecurity.ch>

I was reading an interesting approach that SER had taken in making the
mediaproxy (rtp proxy) component be announced as a SDP candidate:
http://mediaproxy.ag-projects.com/wiki/ICE

That way the SIP client has to talk ICE and STUN but without the
requirement to run TURN (that's very few supported).

Would be already possible somehow to have a setup like this with
FreeSWITCH having it working also as the role of a TURN server but
without even implementing TURN?

Fabio


From tculjaga at gmail.com  Sat Oct  2 12:18:19 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sat, 2 Oct 2010 21:18:19 +0200
Subject: [Freeswitch-users] FreeSwitch - mod_opal,
	mod_h323 never get installed
In-Reply-To: <AANLkTi=wvXA+mYPdCjSL07Wk=psBSXrYh58b7gkP0LTO@mail.gmail.com>
References: <AANLkTi=9Hw6dGtzdaKwYR+oA2R1y5P0vfuHJqJW8Fncd@mail.gmail.com>
	<AANLkTi=wvXA+mYPdCjSL07Wk=psBSXrYh58b7gkP0LTO@mail.gmail.com>
Message-ID: <AANLkTi=gF1N13NMyGhmgmNw+mJTyhjO06g0o_Pr2Hg9_@mail.gmail.com>

On Fri, Oct 1, 2010 at 3:15 AM, Michael Collins <msc at freeswitch.org> wrote:

> What about mod_h323?
>
>
>
Updated a bit: http://wiki.freeswitch.org/wiki/Mod_h323, you can't do
anything wrong... it simply works.
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From shamun.toha at gmail.com  Sat Oct  2 14:44:54 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Sat, 2 Oct 2010 23:44:54 +0200
Subject: [Freeswitch-users] mod skype legal
In-Reply-To: <AANLkTimCm9h+1FoendyAmUiyG0DehXURip_LCLyghUQZ@mail.gmail.com>
References: <AANLkTi=Eha03JxMxYWZqyShCgvEw+g+AgyFJi3QJnXMz@mail.gmail.com>
	<AANLkTinF+-hUSTa=1FfMzjfodL5P8vowGTCLGiZ2DyOQ@mail.gmail.com>
	<732012231-1285998749-cardhu_decombobulator_blackberry.rim.net-1973678712-@bda034.bisx.prodap.on.blackberry>
	<AANLkTik0AZQrznDshpLVCQrgY1Jpc8Bd4_SMCX5QgQeN@mail.gmail.com>
	<AANLkTimCm9h+1FoendyAmUiyG0DehXURip_LCLyghUQZ@mail.gmail.com>
Message-ID: <AANLkTim+Wqzb1D7uBZwzmkUnt=NGK1muGXkgQyjahWXT@mail.gmail.com>

Hello,

WRONG.  Let me explain it like this.

Argument 1: You and me we are talking in English right? If we use English
protocol for our communication, are we using Illegal protocol?
- Therefore, Skype API is a official protocol from Skype it self, we are
communicating with Skype from FreeSwitch.
- Because we are smart and creative developers, we are using more
ADVANCED/EXPERTS level of Skype usage.

Argument 2: Lets say Skype user id is registered with Skype company switch,
they can still check which version of skype or from where the skype is
making those connections

- Yes, they can block that skypeid so that those id cant login again, but
still its Skype who started the international Protocol/API, and why they
should even block it now? We are still giving them more traffic and
publicity!!

- By the way, still many call centers are using Skype API too!!! to process
calls and solve PBX problems.

- Many hardwares are using those Skype API concept too!!! to process
hardware inputs to skype instance using that same Protocol


In no where, no terms, we exist as illegal implementation of Skype
API!!!!!!!


Thanks & Regards





On Sat, Oct 2, 2010 at 4:32 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:

> Hi,
>
> Recently we saw that skype blocked fring and nimbuzz... could it also be
> the case that they see calls from FS and block them?
>
>
> On Sat, Oct 2, 2010 at 11:11 AM, Shamun toha md <shamun.toha at gmail.com>wrote:
>
>> You are welcome, I am trying to research for my thesis, and increase Skype
>> Instance to 100 or 200 or 300 Channels in 4 socket mother board with 12 core
>> on each. 12 core processor multiply by 4 = 48 core processors.
>>
>> So, please keep growing this mod_skypopen, without any afraid, very soon,
>> if the ALSA driver bugs can be reduced, we can have like 500 Channels on
>> small CPU, because in my analyze i saw ALSA driver is causing many problems,
>> not the Skype API.
>>
>> Thanks & Regards
>>
>>
>>
>>
>> On Sat, Oct 2, 2010 at 7:52 AM, Budi wibowo <bwibowo at gmail.com> wrote:
>>
>>> Thx a lot for the positive feedback
>>>
>>>
>>> Regards
>>>
>>> Budi
>>> -----Original Message-----
>>> From: Shamun toha md <shamun.toha at gmail.com>
>>> Sender: freeswitch-users-bounces at lists.freeswitch.org
>>> Date: Thu, 30 Sep 2010 23:55:41
>>> To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
>>> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Subject: Re: [Freeswitch-users] mod skype legal
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Sat Oct  2 14:55:01 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 2 Oct 2010 16:55:01 -0500
Subject: [Freeswitch-users] Can FreeSWITCH act as a turn server for ICE?
In-Reply-To: <4CA7784D.9050904@infosecurity.ch>
References: <4CA7784D.9050904@infosecurity.ch>
Message-ID: <AD68E753-7870-4FE0-A002-B29249F4084D@freeswitch.org>

Nope.  Its not listed on the Spec sheet we generally don't support it.  The issue with ICE and TURN is their are no implementations that are license compatible that could be used. 

/b

On Oct 2, 2010, at 1:22 PM, Fabio Pietrosanti (naif) wrote:

> I was reading an interesting approach that SER had taken in making the
> mediaproxy (rtp proxy) component be announced as a SDP candidate:
> http://mediaproxy.ag-projects.com/wiki/ICE
> 
> That way the SIP client has to talk ICE and STUN but without the
> requirement to run TURN (that's very few supported).
> 
> Would be already possible somehow to have a setup like this with
> FreeSWITCH having it working also as the role of a TURN server but
> without even implementing TURN?
> 
> Fabio



From Nabble at slickdeals.endjunk.com  Sat Oct  2 14:15:30 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sat, 2 Oct 2010 14:15:30 -0700 (PDT)
Subject: [Freeswitch-users] mod skype legal
In-Reply-To: <AANLkTimCm9h+1FoendyAmUiyG0DehXURip_LCLyghUQZ@mail.gmail.com>
References: <AANLkTi=Eha03JxMxYWZqyShCgvEw+g+AgyFJi3QJnXMz@mail.gmail.com>
	<AANLkTinF+-hUSTa=1FfMzjfodL5P8vowGTCLGiZ2DyOQ@mail.gmail.com>
	<732012231-1285998749-cardhu_decombobulator_blackberry.rim.net-1973678712-@bda034.bisx.prodap.on.blackberry>
	<AANLkTik0AZQrznDshpLVCQrgY1Jpc8Bd4_SMCX5QgQeN@mail.gmail.com>
	<AANLkTimCm9h+1FoendyAmUiyG0DehXURip_LCLyghUQZ@mail.gmail.com>
Message-ID: <1286054130853-5595032.post@n2.nabble.com>



Saeed Ahmed wrote:
> Recently we saw that skype blocked fring and nimbuzz... could it also be
> the case that they see calls from FS and block them?
In the FRing case, Skype is able to block FRing server(s). As with SkypOpen,
Skype will have a hell of time to block individual SkypOpen call which looks
exactly like a real Skype call.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-skype-legal-tp5589549p5595032.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mario_fs at mgtech.com  Sat Oct  2 16:33:33 2010
From: mario_fs at mgtech.com (Mario)
Date: Sat, 02 Oct 2010 16:33:33 -0700
Subject: [Freeswitch-users] error on install on os x 10.6.4
In-Reply-To: <4048EE03-F887-4D0C-A8A9-A72672164CFF@freeswitch.org>
References: <AANLkTikb_MyGHMGet9682meMwbVXiJG_NeB6fQipYSr2@mail.gmail.com>	<AANLkTi=W71voJZJDd=5stGJZX3_S7_d93Gune368+gBa@mail.gmail.com>	<-2648041760610933328@unknownmsgid>	<AANLkTinxB3Rc3TxR0=BvBnhZ-cjXz02b9VxWY2k8D8WV@mail.gmail.com>	<AANLkTi=-0-jJx3qTH+fG4w2ik4hwV1yvzqLGzCEyzoCk@mail.gmail.com>	<B08F4A6E-12DB-40FE-9739-1FAA61B606C5@freeswitch.org>	<-6194582537160853113@unknownmsgid>
	<4048EE03-F887-4D0C-A8A9-A72672164CFF@freeswitch.org>
Message-ID: <4CA7C14D.6050703@mgtech.com>

Peter, did you see my update on your original thread "Struggling with
installing"? I was able to install SPANDSP and prereqs clean. However,
someone mentioned it was convoluted and should not have been needed, it
was what I did to get a clean install of FS. BTW, I only used the
development tools from Apple. and at no time did I use macports.

On 10/02/10 07:40, Brian West wrote:
> That should be fixed.  Installing any of that will destroy your build
> environment.   See this is a community wiki... so someone put that on
> there.  It should be removed and clarified. 
> 
> Default build tools plus http://code.google.com/p/git-osx-installer/ are
> all you need.
> 
> /b
> 
> On Oct 2, 2010, at 1:59 AM, Peter Schrock wrote:
> 
>> And yet it says on this page:
>> http://wiki.freeswitch.org/wiki/Git_Install
>> to "Install MacPorts and run as root" to install git-core (to meet the
>> requirement of having git to build a git tree). As far as I know
>> MacPorts is formerly known as DarwinPorts. 
>> If this is the case, why is it that the wiki is so keen on saying to
>> make sure the prerequisites are met and it isn't mentioned to avoid
>> fink and darwin? Those are suppose to be the equivalent of "yum" for
>> Linux on the OS X, why wouldn't I think to check those sources for any
>> required prerequisites?
>> This is aggravating!
>>
>> On Oct 1, 2010, at 8:49 PM, Brian West <brian at freeswitch.org
>> <mailto:brian at freeswitch.org>> wrote:
>>
>>> I'm going to only guess you have installed darwin ports or fink which
>>> has totally screwed your dev evn.  We only support building with the
>>> utils apple provides NOTHING MORE.
>>>
>>> /b
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From steveayre at gmail.com  Sat Oct  2 16:46:42 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 3 Oct 2010 00:46:42 +0100
Subject: [Freeswitch-users] g729 passthrough mode
In-Reply-To: <COL109-W13224EAAAE107BFB3C60A1F46A0@phx.gbl>
References: <COL109-W13224EAAAE107BFB3C60A1F46A0@phx.gbl>
Message-ID: <AANLkTinmjuB9WoW-S4WuoccjBdjLhPnw6GPyQPWocgtN@mail.gmail.com>

If G729 is at both ends, then no encoding/decoding is required as the
raw data can be sent straight through. This is passthrough mode.
Because there is no encoding/decoding stage, there is no licence
required in this case.

-Steve

On 2 October 2010 13:44, Abid Saleem <abid_freeswitch at live.com> wrote:
> Hi All,
> Could you please tell me what is meant by 9729 codec in passthrough mode in
> the class 4 routing case. For example if Originating provider supports g729
> and is sending call to FS which recieves and routes the call to termination
> provider who also supports g729 codec. So is the license required in this
> case or not?
> Thanks for your time.
> Regards
> --------------
> Abid Saleem
> Sr. Product Manager
> Terminus Technologies
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From peter.schrock at gmail.com  Sun Oct  3 00:55:36 2010
From: peter.schrock at gmail.com (Peter Schrock)
Date: Sun, 3 Oct 2010 00:55:36 -0700
Subject: [Freeswitch-users] error on install on os x 10.6.4
In-Reply-To: <4CA7C14D.6050703@mgtech.com>
References: <AANLkTikb_MyGHMGet9682meMwbVXiJG_NeB6fQipYSr2@mail.gmail.com>
	<AANLkTi=W71voJZJDd=5stGJZX3_S7_d93Gune368+gBa@mail.gmail.com>
	<-2648041760610933328@unknownmsgid>
	<AANLkTinxB3Rc3TxR0=BvBnhZ-cjXz02b9VxWY2k8D8WV@mail.gmail.com>
	<AANLkTi=-0-jJx3qTH+fG4w2ik4hwV1yvzqLGzCEyzoCk@mail.gmail.com>
	<B08F4A6E-12DB-40FE-9739-1FAA61B606C5@freeswitch.org>
	<-6194582537160853113@unknownmsgid>
	<4048EE03-F887-4D0C-A8A9-A72672164CFF@freeswitch.org>
	<4CA7C14D.6050703@mgtech.com>
Message-ID: <AANLkTi=Ec73vF2TzEeMtfyPEgThGKxUb40gsTdKhg-GK@mail.gmail.com>

So, I got a chance to delete macports and fink and tried to reconfigure and
reinstall. I got FS to run once, but now I keep getting this error:

http://pastebin.freeswitch.org/14108

Mario,
I even managed to get everything loaded as you mentioned in the other
thread. At first I tried without spandsp. Now I have it loaded, but I can't
get the thing to run correctly. Can anyone advise how to fix this error? I
tried both suggestions that it gives to no avail.

Peter
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From tculjaga at gmail.com  Sun Oct  3 00:55:41 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sun, 3 Oct 2010 09:55:41 +0200
Subject: [Freeswitch-users] Errors compiling mod_h323
In-Reply-To: <201010011258.41592.justlikeef@gmail.com>
References: <201010011258.41592.justlikeef@gmail.com>
Message-ID: <AANLkTikF2-vsdTG16vn=razRQFjM4k6AF2E74=3GcSPb@mail.gmail.com>

On Fri, Oct 1, 2010 at 6:58 PM, Rob Hutton <justlikeef at gmail.com> wrote:

> I am getting the following error compiling mod_h323 on OpenSuse 11.3
> X86_64.
> I don't even know where to begin on this, but would me more than happy to
> help
> work it out...
>
> Compiling
> /usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp...
> quiet_libtool: compile:  g++ -g -ggdb -I/usr/local/include/ptlib -
> I/usr/local/include/openh323 -I. -DPTRACING=1 -D_REENTRANT -fno-exceptions
> -
> I/usr/src/freeswitch/freeswitch/src/include -
> I/usr/src/freeswitch/freeswitch/src/include -
> I/usr/src/freeswitch/freeswitch/libs/libteletone/src -fPIC
> -fvisibility=hidden
> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -
> DHAVE_CONFIG_H -c
> /usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp
>  -fPIC
> -DPIC -o .libs/mod_h323.o
> In file included from
>
> /usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp:31:0:
> /usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:110:1:
> error: too many initializers for ?const char* const [8]?
>

complains about that :

const char* const
GetAnswerCallResponse[H323Connection::NumAnswerCallResponses+1]={
        "AnswerCallNow",
        "AnswerCallDenied",
        "AnswerCallPending",
        "AnswerCallDeferred",
        "AnswerCallAlertWithMedia",
        "AnswerCallDeferredWithMedia",
        "AnswerCallDeniedByInvalidCID",
        "AnswerCallNowWithAlert",
        "NumAnswerCallResponses"
};

for some reason H323Capability::e_NumMainTypes+1 returnes less than the
array actual size



> /usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:121:1:
> error: too many initializers for ?const char* const [5]?
>

complains about that :

const char* const GetMainTypes[H323Capability::e_NumMainTypes+1] = {
        "Audio",
        "Video",
        "Data",
        "UserInput",
        "ExtendVideo",
        "GenericControl",
        "ConferenceControl",
        "NumMainTypes"
};

for some reason H323Capability::e_NumMainTypes+1 returnes less than the
array actual size





>
> /usr/src/freeswitch/freeswitch/src/mod/endpoints/mod_h323/mod_h323.h:212:40:
> error: expected class-name before ?{? token
>

here it is complaining about something it should not :))) Everything is fine
in that function.






bottom line, i guess you don't have the correct combination of ptlib and
h323plus.

Please check http://wiki.freeswitch.org/wiki/Mod_h323 for detailed
installation process.

Also, note we are focusing on CentOS and Debian distro mainly and its quite
hard to guess what is wrong on different distros.
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From dujinfang at gmail.com  Sun Oct  3 01:36:43 2010
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 3 Oct 2010 16:36:43 +0800
Subject: [Freeswitch-users] error on install on os x 10.6.4
In-Reply-To: <AANLkTi=Ec73vF2TzEeMtfyPEgThGKxUb40gsTdKhg-GK@mail.gmail.com>
References: <AANLkTikb_MyGHMGet9682meMwbVXiJG_NeB6fQipYSr2@mail.gmail.com>
	<AANLkTi=W71voJZJDd=5stGJZX3_S7_d93Gune368+gBa@mail.gmail.com>
	<-2648041760610933328@unknownmsgid>
	<AANLkTinxB3Rc3TxR0=BvBnhZ-cjXz02b9VxWY2k8D8WV@mail.gmail.com>
	<AANLkTi=-0-jJx3qTH+fG4w2ik4hwV1yvzqLGzCEyzoCk@mail.gmail.com>
	<B08F4A6E-12DB-40FE-9739-1FAA61B606C5@freeswitch.org>
	<-6194582537160853113@unknownmsgid>
	<4048EE03-F887-4D0C-A8A9-A72672164CFF@freeswitch.org>
	<4CA7C14D.6050703@mgtech.com>
	<AANLkTi=Ec73vF2TzEeMtfyPEgThGKxUb40gsTdKhg-GK@mail.gmail.com>
Message-ID: <AANLkTikAVRj85_YhnNF5U_B1m0qPE43DfwX_LU_5YEoM@mail.gmail.com>

ignore line 1

how did you run once ?  probably the freeswitch process is still running, try

ps aux|grep freeswitch

On Sun, Oct 3, 2010 at 3:55 PM, Peter Schrock <peter.schrock at gmail.com> wrote:
> So, I got a chance to delete macports and fink and tried to reconfigure and
> reinstall. I got FS to run once, but now I keep getting this error:
> http://pastebin.freeswitch.org/14108
> Mario,
> I even managed to get everything loaded as you mentioned in the other
> thread. At first I tried without spandsp. Now I have it loaded, but I can't
> get the thing to run correctly. Can anyone advise how to fix this error? I
> tried both suggestions that it gives to no avail.
> Peter
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From Nabble at slickdeals.endjunk.com  Sun Oct  3 07:34:29 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 3 Oct 2010 07:34:29 -0700 (PDT)
Subject: [Freeswitch-users] g729 passthrough mode
In-Reply-To: <AANLkTinmjuB9WoW-S4WuoccjBdjLhPnw6GPyQPWocgtN@mail.gmail.com>
References: <COL109-W13224EAAAE107BFB3C60A1F46A0@phx.gbl>
	<AANLkTinmjuB9WoW-S4WuoccjBdjLhPnw6GPyQPWocgtN@mail.gmail.com>
Message-ID: <1286116469745-5596240.post@n2.nabble.com>



Steven Ayre wrote:
> 
> If G729 is at both ends, then no encoding/decoding is required as the
> raw data can be sent straight through. This is passthrough mode.
> Because there is no encoding/decoding stage, there is no licence
> required in this case.
A more appropriate way to say this is no transcoding. Encoding/decoding
processes are still done at each endpoints on the device, i.e IP Phone, ATA,
and/or computer (with softphone), and a G729 license is still required when
the call uses the G729 CoDec (which is mostly paid when one acquired the
device with a built-in G729 CoDec). So, to say no encoding/deconding
involved and no license required are really misleading and definitely
incorrect.

BTW, one can also configure his/her FS system with bypass media option to
ensure the FS only does the SIP with no media handling. The advantage using
this approach is to leave FS alone and free up some of its networking
resounrces to handle the media during a call. There is also some
disadvantages, i.e. no MOH, etc.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/g729-passthrough-mode-tp5594193p5596240.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From peter.schrock at gmail.com  Sun Oct  3 08:34:44 2010
From: peter.schrock at gmail.com (Peter Schrock)
Date: Sun, 3 Oct 2010 08:34:44 -0700
Subject: [Freeswitch-users] error on install on os x 10.6.4
In-Reply-To: <AANLkTikAVRj85_YhnNF5U_B1m0qPE43DfwX_LU_5YEoM@mail.gmail.com>
References: <AANLkTikb_MyGHMGet9682meMwbVXiJG_NeB6fQipYSr2@mail.gmail.com>
	<AANLkTi=W71voJZJDd=5stGJZX3_S7_d93Gune368+gBa@mail.gmail.com>
	<-2648041760610933328@unknownmsgid>
	<AANLkTinxB3Rc3TxR0=BvBnhZ-cjXz02b9VxWY2k8D8WV@mail.gmail.com>
	<AANLkTi=-0-jJx3qTH+fG4w2ik4hwV1yvzqLGzCEyzoCk@mail.gmail.com>
	<B08F4A6E-12DB-40FE-9739-1FAA61B606C5@freeswitch.org>
	<-6194582537160853113@unknownmsgid>
	<4048EE03-F887-4D0C-A8A9-A72672164CFF@freeswitch.org>
	<4CA7C14D.6050703@mgtech.com>
	<AANLkTi=Ec73vF2TzEeMtfyPEgThGKxUb40gsTdKhg-GK@mail.gmail.com>
	<AANLkTikAVRj85_YhnNF5U_B1m0qPE43DfwX_LU_5YEoM@mail.gmail.com>
Message-ID: <-4659146215214473982@unknownmsgid>

The response that is given back is:
"ps: invalid process id: freeswitch"
so, I don't believe it is running, even in background mode.

Peter

On Oct 3, 2010, at 1:42 AM, Seven Du <dujinfang at gmail.com> wrote:

> ignore line 1
>
> how did you run once ?  probably the freeswitch process is still running, try
>
> ps aux|grep freeswitch
>
> On Sun, Oct 3, 2010 at 3:55 PM, Peter Schrock <peter.schrock at gmail.com> wrote:
>> So, I got a chance to delete macports and fink and tried to reconfigure and
>> reinstall. I got FS to run once, but now I keep getting this error:
>> http://pastebin.freeswitch.org/14108
>> Mario,
>> I even managed to get everything loaded as you mentioned in the other
>> thread. At first I tried without spandsp. Now I have it loaded, but I can't
>> get the thing to run correctly. Can anyone advise how to fix this error? I
>> tried both suggestions that it gives to no avail.
>> Peter
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From tculjaga at gmail.com  Sun Oct  3 08:57:58 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sun, 3 Oct 2010 17:57:58 +0200
Subject: [Freeswitch-users] Errors compiling mod_opal
In-Reply-To: <4CA7455D.8080808@puzzled.xs4all.nl>
References: <201010011630.26508.justlikeef@gmail.com>
	<4CA7455D.8080808@puzzled.xs4all.nl>
Message-ID: <AANLkTin=Xi6dVAU4z_VKZV1=mZL2ATJuf-2+qaqhtKy8@mail.gmail.com>

On Sat, Oct 2, 2010 at 4:44 PM, Patrick Lists <
freeswitch-list at puzzled.xs4all.nl> wrote:

> On 10/01/2010 10:30 PM, Rob Hutton wrote:
> > OpenSuse 11.3 x86_64.  No clue where to start, willing to help...
>
> Afaik you need to have the following installed (also the -devel packages):
>
> For mod_opal:
> ptlib version 2.8.1 (or 2.8.2)
> opal  version 3.8.1 (or 3.8.2)
>
> For mod_h323:
> ptlib version 2.8.1 (or 2.8.2)
> h323+ version cvs from 20100525
>
>

check: http://wiki.freeswitch.org/wiki/mod_h323 its quite easy.
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From mario_fs at mgtech.com  Sun Oct  3 09:14:38 2010
From: mario_fs at mgtech.com (Mario)
Date: Sun, 03 Oct 2010 09:14:38 -0700
Subject: [Freeswitch-users] error on install on os x 10.6.4
In-Reply-To: <-4659146215214473982@unknownmsgid>
References: <AANLkTikb_MyGHMGet9682meMwbVXiJG_NeB6fQipYSr2@mail.gmail.com>	<AANLkTi=W71voJZJDd=5stGJZX3_S7_d93Gune368+gBa@mail.gmail.com>	<-2648041760610933328@unknownmsgid>	<AANLkTinxB3Rc3TxR0=BvBnhZ-cjXz02b9VxWY2k8D8WV@mail.gmail.com>	<AANLkTi=-0-jJx3qTH+fG4w2ik4hwV1yvzqLGzCEyzoCk@mail.gmail.com>	<B08F4A6E-12DB-40FE-9739-1FAA61B606C5@freeswitch.org>	<-6194582537160853113@unknownmsgid>	<4048EE03-F887-4D0C-A8A9-A72672164CFF@freeswitch.org>	<4CA7C14D.6050703@mgtech.com>	<AANLkTi=Ec73vF2TzEeMtfyPEgThGKxUb40gsTdKhg-GK@mail.gmail.com>	<AANLkTikAVRj85_YhnNF5U_B1m0qPE43DfwX_LU_5YEoM@mail.gmail.com>
	<-4659146215214473982@unknownmsgid>
Message-ID: <4CA8ABEE.6050508@mgtech.com>

Could not find much except this explanation for the error:
http://developer.apple.com/library/mac/#documentation/Darwin/Reference/ManPages/man3/tcgetpgrp.3.html

On 10/03/2010 08:34 AM, Peter Schrock wrote:
> The response that is given back is:
> "ps: invalid process id: freeswitch"
> so, I don't believe it is running, even in background mode.
> 
> Peter
> 
> On Oct 3, 2010, at 1:42 AM, Seven Du <dujinfang at gmail.com> wrote:
> 
>> ignore line 1
>>
>> how did you run once ?  probably the freeswitch process is still running, try
>>
>> ps aux|grep freeswitch
>>
>> On Sun, Oct 3, 2010 at 3:55 PM, Peter Schrock <peter.schrock at gmail.com> wrote:
>>> So, I got a chance to delete macports and fink and tried to reconfigure and
>>> reinstall. I got FS to run once, but now I keep getting this error:
>>> http://pastebin.freeswitch.org/14108
>>> Mario,
>>> I even managed to get everything loaded as you mentioned in the other
>>> thread. At first I tried without spandsp. Now I have it loaded, but I can't
>>> get the thing to run correctly. Can anyone advise how to fix this error? I
>>> tried both suggestions that it gives to no avail.
>>> Peter
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Blog: http://www.dujinfang.com
>> Proj:  http://www.freeswitch.org.cn
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
*Mario*


From Nabble at slickdeals.endjunk.com  Sun Oct  3 10:52:31 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 3 Oct 2010 10:52:31 -0700 (PDT)
Subject: [Freeswitch-users] Errors compiling mod_opal
In-Reply-To: <201010011630.26508.justlikeef@gmail.com>
References: <201010011630.26508.justlikeef@gmail.com>
Message-ID: <1286128351258-5596700.post@n2.nabble.com>



Rob Hutton wrote:
> 
> OpenSuse 11.3 x86_64.  No clue where to start, willing to help...
Patrick is right that your system needs to have the opal library package
installed to compile mod_opal. If you don't need mod_opal, you can disable
its build by editing the module.conf file under the root directory of FS
source tree and add a # to the front of endpoints/mod_opal line. I don't
need the mod_opal for my FS hosted on a Seagate DockStar. So, the
modules.conf file I have during the build process contains
#endpoints/mod_opal and not endpoints/mod_opal.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Errors-compiling-mod-opal-tp5594220p5596700.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Peter.Hinman at ParcelPool.com  Sun Oct  3 11:36:54 2010
From: Peter.Hinman at ParcelPool.com (Peter Hinman)
Date: Sun, 03 Oct 2010 12:36:54 -0600
Subject: [Freeswitch-users] Alphanumeric case sensitivity
In-Reply-To: <AANLkTim+FOi70+zkeg4ODbz=Dd1TYYW3WFWH3otaShjx@mail.gmail.com>
References: <4CA6578B.6090704@ParcelPool.com>
	<AANLkTim+FOi70+zkeg4ODbz=Dd1TYYW3WFWH3otaShjx@mail.gmail.com>
Message-ID: <4CA8CD46.3060406@ParcelPool.com>

  Hi Michael - thank you for taking a look at the code.

The user_ID for User1 at example.com is lower case "user1", but the user is 
able to successfully register as "User1".

Policy is good, but it's always nice to be able to enforce policy in 
code.  Is there a way to force all registrations to match the case of 
the User_ID? I haven't seen anything in the wiki or the book that seemed 
to do that.

Here is an example of one of our user accounts from the directory - are 
we missing something there?:

<user id="adam.smith" number-alias="1001" mailbox="1001">
<params>
<param name="password" value = "NotARealPassword"/>
<param name="vm-password" value = "****"/>
<param name="vm-email-all-messages" value = "true"/>
<param name="vm-attach-file" value = "true"/>
<param name="vm-keep-local-after-email" value = "true"/>
<param name="vm-mailto" value = "adam.smith at ParcelPool.com"/>
</params>
<variables>
<variable name="toll_allow" value="domestic,international,local"/>
<variable name="accountcode" value="1001"/>
<variable name="user_context" value="default"/>
</variables>
</user>



On 10/1/2010 4:35 PM, Michael Collins wrote:
> What's the user_id of "User1 at example.com <mailto:User1 at example.com>"? 
> Is it "User1" or "user1"? (BTW, this is a classic example of why 
> alphanumeric usernames are a bad idea in a telephony environment.)
>
> I would recommend that you enforce some policy, like always use all 
> lower case. (I've poked around in the source code and I don't see any 
> easy way to make alpha usernames be case insensitive.)
>
> -MC
>
> On Fri, Oct 1, 2010 at 2:50 PM, Peter Hinman 
> <Peter.Hinman at parcelpool.com <mailto:Peter.Hinman at parcelpool.com>> wrote:
>
>      I'm trying to get FS working with alphanumeric UserIDs.  I'm running
>     into a problem when user1 registers using mixed case
>     (User1 at example.com <mailto:User1 at example.com>)
>     and then user2 tries to reach user1 by dialing "user1" instead of
>     "User1".
>
>     FS fails to make the connection between the two extensions and
>     reports:
>     [ERR] switch_ivr_originate.c:2648 Cannot create outgoing channel
>     of type
>     [error] cause: [USER_NOT_REGISTERED]
>
>     The dial plan string for bridging my internal users is:
>     <action application="bridge"
>     data="user/${dialed_extension}@${domain_name}"/>
>
>     Is there a way to use alphanumeric UserIDs and bridge calls between
>     internal extensions when the dialing user doesn't know exactly how
>     (upper, lower, mixed case) the destination user/extension registered
>     with FS?
>
>     --
>     Peter Hinman
>     ParcelPool.com
>     Office: 801-434-7400
>     Cell: 801-830-8292
>
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Peter Hinman
ParcelPool.com
Office: 801-434-7400
Cell: 801-830-8292

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From jonas.gauffin at gmail.com  Sun Oct  3 11:58:09 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Sun, 3 Oct 2010 20:58:09 +0200
Subject: [Freeswitch-users] Intercept Ringing Call
In-Reply-To: <AANLkTimRazMC_-KaRjPg3S3NBRdSPrS9rXAs=nPySsfq@mail.gmail.com>
References: <009801cb617c$763d7810$62b86830$@teotech.com>
	<AANLkTim1-+x27rZwZ33YRPAvDPtJc3sa7braw58S7nHC@mail.gmail.com>
	<AANLkTikL4HWhw-k7zvkZ2BuGJO9iHhi+v44Muc-0NS4v@mail.gmail.com>
	<A7814BB2-FB31-4ACB-A4A0-56CB3503BEB3@freeswitch.org>
	<AANLkTin4F2=shvUf=yWHUViiaMPde_dUpFa7FCia+w0=@mail.gmail.com>
	<AANLkTimRazMC_-KaRjPg3S3NBRdSPrS9rXAs=nPySsfq@mail.gmail.com>
Message-ID: <AANLkTimc1UFXa_edzoTuF8FT_sB7Ojgc40cNB2rhT46f@mail.gmail.com>

Or change so that the search form uses google by default

On Sat, Oct 2, 2010 at 5:56 PM, Milena <testeador01 at gmail.com> wrote:

>
> Aww Brian!! give me a break!!! what really sucks it that nobody uses google
> ...maybe we should put this thing at the wiki faqs:
>
> * HOW TO SEARCH ON THE WIKI: *
> On your browser go to:
>        www.google.com
> On the search box type:
>        <write your search terms in here> site:wiki.freeswitch.org
>
> voil?!!! :D
>
> What do you think? ;)
>
> -Milena
> On Sat, Oct 2, 2010 at 10:20 AM, Henry Huang <b_ball_henry at hotmail.com>wrote:
>
>> Possibly, the wiki search takes exact terms. One have to be precise and
>> know the exact term that they are searching for.
>>
>> On Sat, Oct 2, 2010 at 10:46 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> could just be the wiki search sucks :P
>>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From shamun.toha at gmail.com  Sun Oct  3 12:49:13 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Sun, 3 Oct 2010 21:49:13 +0200
Subject: [Freeswitch-users] FreeSwitch - mod_opal,
	mod_h323 never get installed
In-Reply-To: <AANLkTi=gF1N13NMyGhmgmNw+mJTyhjO06g0o_Pr2Hg9_@mail.gmail.com>
References: <AANLkTi=9Hw6dGtzdaKwYR+oA2R1y5P0vfuHJqJW8Fncd@mail.gmail.com>
	<AANLkTi=wvXA+mYPdCjSL07Wk=psBSXrYh58b7gkP0LTO@mail.gmail.com>
	<AANLkTi=gF1N13NMyGhmgmNw+mJTyhjO06g0o_Pr2Hg9_@mail.gmail.com>
Message-ID: <AANLkTi=vc53L4gi3WDJYww34G+85XaOCunJ_YpUAvvUb@mail.gmail.com>

Hello,

I installed the mod_opal using the existing script called buildopal.sh and
then it updated all the ptlib and opal (otherwise it never works), after
that i can do like

1. load mod_opal
2. originate opal/h323:sayhello at myworld 5000

But Ekiga never works in Fedora, so call was failed, not sure i made this
working till now.


[root at example build]# ls
buildlib.sh         fixautoconf.sh             freeswitch.ld.so.conf
getlib.sh          Makefile.am       sounds_version.txt
buildmrcpserver.sh  fixautomake.sh             freeswitch.monitrc
getlib.sh.in       Makefile.in       strip.pl
buildopal.sh        fixlibtool.sh              freeswitch.pc
getsounds.sh       modmake.rules     swigall.sh
buildzrtp.sh        freeswitch.build           freeswitch.pc.in
getsounds.sh.in    modmake.rulesam   turbo_build.sh
config              freeswitch.init.archlinux  freeswitch.sysconfig
ignore_helper.pl   modmake.rules.in
config.layout       freeswitch.init.redhat     fs_ivrd.init.redhat
indent_options.sh  modules.conf.in
curses.patch        freeswitch.init.suse       gen_indent.pl
Makefile           moh_version.txt
[root at example build]#


What is the difference between mod_opal and mod_h323 then? Do i have to
uninstall mod_opal to get mod_h323? By the way i did it in Fedora 12. Will
it be different in CentOS ?


Thanks & Best regards





On Sat, Oct 2, 2010 at 9:18 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

>
>
> On Fri, Oct 1, 2010 at 3:15 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> What about mod_h323?
>>
>>
>>
> Updated a bit: http://wiki.freeswitch.org/wiki/Mod_h323, you can't do
> anything wrong... it simply works.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Sun Oct  3 13:06:56 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sun, 3 Oct 2010 22:06:56 +0200
Subject: [Freeswitch-users] g729 passthrough mode
In-Reply-To: <1286116469745-5596240.post@n2.nabble.com>
References: <COL109-W13224EAAAE107BFB3C60A1F46A0@phx.gbl>
	<AANLkTinmjuB9WoW-S4WuoccjBdjLhPnw6GPyQPWocgtN@mail.gmail.com>
	<1286116469745-5596240.post@n2.nabble.com>
Message-ID: <8F6FC0E5-B0A2-4AAF-8489-41EA35A2C60F@ipeva.fr>

Mazilo,

I think Steven meant that no transcoding was required at the FS level,  
which was the point of Abid's question.

And the main disadvantage of using bypass media is probably the NAT  
handling, and the lack of topology hiding.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 03/10/2010 ? 16:34, mazilo a ?crit :

>
>
> Steven Ayre wrote:
>>
>> If G729 is at both ends, then no encoding/decoding is required as the
>> raw data can be sent straight through. This is passthrough mode.
>> Because there is no encoding/decoding stage, there is no licence
>> required in this case.
> A more appropriate way to say this is no transcoding. Encoding/ 
> decoding
> processes are still done at each endpoints on the device, i.e IP  
> Phone, ATA,
> and/or computer (with softphone), and a G729 license is still  
> required when
> the call uses the G729 CoDec (which is mostly paid when one acquired  
> the
> device with a built-in G729 CoDec). So, to say no encoding/deconding
> involved and no license required are really misleading and definitely
> incorrect.
>
> BTW, one can also configure his/her FS system with bypass media  
> option to
> ensure the FS only does the SIP with no media handling. The  
> advantage using
> this approach is to leave FS alone and free up some of its networking
> resounrces to handle the media during a call. There is also some
> disadvantages, i.e. no MOH, etc.
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive  
> to men,
> but not when used together.
> -- 
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/g729-passthrough-mode-tp5594193p5596240.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From adminjew at gmail.com  Sun Oct  3 13:32:14 2010
From: adminjew at gmail.com (Yitzchok)
Date: Sun, 3 Oct 2010 16:32:14 -0400
Subject: [Freeswitch-users] mod_managed on linux centos
In-Reply-To: <4C838C3B.3070202@puzzled.xs4all.nl>
References: <AANLkTikg4bxav+e04fAmJ_dKTHKERAPk715h9oHeYMRh@mail.gmail.com>
	<1283347083542-5486890.post@n2.nabble.com>
	<AANLkTi=xt9CZBPZLnBX56e2Rjdc_FRf_=er6Yy7w7ZuN@mail.gmail.com>
	<AANLkTi=S=tpzczDM8WSm6N8AeTkN3XNTPda6048sRoJ9@mail.gmail.com>
	<4C801B6E.4050103@puzzled.xs4all.nl>
	<AANLkTimzjbLHbObgsQKqX3t-hfaZWr95UGMi17YxZFHS@mail.gmail.com>
	<4C838C3B.3070202@puzzled.xs4all.nl>
Message-ID: <AANLkTinP+UfvkM0dmBycy5GqivAXNHGF9RnTBNvV0gV4@mail.gmail.com>

I still didn't get this to work (I tried in centos and opensuse) both return
the same error.

Can someone with mod_managed installed on linux (after copying Demo.csx to
the mod/Managed/ directory) try calling "*managed ApiDemo test*
" and let me know if you get an error


freeswitch at internal> *managed ApiDemo test*
freeswitch at internal> *managed ApiDemo test*



Yitzchok


On Sun, Sep 5, 2010 at 8:25 AM, Patrick Lists <
freeswitch-list at puzzled.xs4all.nl> wrote:

> On 09/05/2010 06:28 AM, Yitzchok wrote:
> > Anyone have any idea why it just works on the first call to the
> > "managed" command and then all future calls to the "managed" command
> > gets me the exception above until I restart FreeSWITCH (and that doesn't
> > fix the problem).
>
> Sorry I have no idea. The only vague inverse similarity I have seen is
> mod_java not loading during startup but it will load fine manually once
> FS is started. Guess you need to do some more debugging.
>
> Regards,
> Patrick
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From adminjew at gmail.com  Sun Oct  3 13:36:46 2010
From: adminjew at gmail.com (Yitzchok)
Date: Sun, 3 Oct 2010 16:36:46 -0400
Subject: [Freeswitch-users] mod_managed on linux centos
In-Reply-To: <AANLkTinP+UfvkM0dmBycy5GqivAXNHGF9RnTBNvV0gV4@mail.gmail.com>
References: <AANLkTikg4bxav+e04fAmJ_dKTHKERAPk715h9oHeYMRh@mail.gmail.com>
	<1283347083542-5486890.post@n2.nabble.com>
	<AANLkTi=xt9CZBPZLnBX56e2Rjdc_FRf_=er6Yy7w7ZuN@mail.gmail.com>
	<AANLkTi=S=tpzczDM8WSm6N8AeTkN3XNTPda6048sRoJ9@mail.gmail.com>
	<4C801B6E.4050103@puzzled.xs4all.nl>
	<AANLkTimzjbLHbObgsQKqX3t-hfaZWr95UGMi17YxZFHS@mail.gmail.com>
	<4C838C3B.3070202@puzzled.xs4all.nl>
	<AANLkTinP+UfvkM0dmBycy5GqivAXNHGF9RnTBNvV0gV4@mail.gmail.com>
Message-ID: <AANLkTik1CY5-6DUvsLg512bGRr66xLF5H8Gi7WYRenHx@mail.gmail.com>

This is interesting if I enter fs_cli and call *managed ApiDemo test *and
then* /exit *and then *./fs_cli* and *managed ApiDemo test *it works again
for one call to managed.
*
*Yitzchok


On Sun, Oct 3, 2010 at 4:32 PM, Yitzchok <adminjew at gmail.com> wrote:

> I still didn't get this to work (I tried in centos and opensuse) both
> return the same error.
>
> Can someone with mod_managed installed on linux (after copying Demo.csx to
> the mod/Managed/ directory) try calling "*managed ApiDemo test*
> " and let me know if you get an error
>
>
> freeswitch at internal> *managed ApiDemo test*
> freeswitch at internal> *managed ApiDemo test*
>
>
>
> Yitzchok
>
>
>
> On Sun, Sep 5, 2010 at 8:25 AM, Patrick Lists <
> freeswitch-list at puzzled.xs4all.nl> wrote:
>
>> On 09/05/2010 06:28 AM, Yitzchok wrote:
>> > Anyone have any idea why it just works on the first call to the
>> > "managed" command and then all future calls to the "managed" command
>> > gets me the exception above until I restart FreeSWITCH (and that doesn't
>> > fix the problem).
>>
>> Sorry I have no idea. The only vague inverse similarity I have seen is
>> mod_java not loading during startup but it will load fine manually once
>> FS is started. Guess you need to do some more debugging.
>>
>> Regards,
>> Patrick
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From peter.schrock at gmail.com  Sun Oct  3 14:07:34 2010
From: peter.schrock at gmail.com (Peter Schrock)
Date: Sun, 3 Oct 2010 14:07:34 -0700
Subject: [Freeswitch-users] pid file can't open
Message-ID: <AANLkTinCfw5N2P7gXsKVpd_m0GZTfDROJXpY9TSzr2qo@mail.gmail.com>

I loaded fink and macports to be able to access binaries to help with my use
of freeswitch. come to find out, fink and macports are not supported. I
tried uninstalling fink and macports and now I am having trouble with
freeswitch and opening the freeswitch.pid file. Does anyone have any
suggestions on what I should do? Here is the error message I keep getting:

Error: stacksize 65532 is too large: run ulimit -s 240 or run
/usr/local/freeswitch/bin/freeswitch -waste.
auto-adjusting stack size for optimal performance...
Cannot open pid file /usr/local/freeswitch/run/freeswitch.pid.

If anyone can help, that would be truly appreciated.

PeterS
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From sos at sokhapkin.dyndns.org  Sun Oct  3 14:16:45 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sun, 3 Oct 2010 17:16:45 -0400
Subject: [Freeswitch-users] pid file can't open
In-Reply-To: <AANLkTinCfw5N2P7gXsKVpd_m0GZTfDROJXpY9TSzr2qo@mail.gmail.com>
References: <AANLkTinCfw5N2P7gXsKVpd_m0GZTfDROJXpY9TSzr2qo@mail.gmail.com>
Message-ID: <201010031716.45863.sos@sokhapkin.dyndns.org>

Either directory /usr/local/freeswitch/run doesn't exists, or FS is started 
with non-root privileges and has no write access to the directory or already 
existing freeswitch.pid file.

On Sunday 03 October 2010, Peter Schrock wrote:
> I loaded fink and macports to be able to access binaries to help with my
>  use of freeswitch. come to find out, fink and macports are not supported.
>  I tried uninstalling fink and macports and now I am having trouble with
>  freeswitch and opening the freeswitch.pid file. Does anyone have any
>  suggestions on what I should do? Here is the error message I keep getting:
> 
> Error: stacksize 65532 is too large: run ulimit -s 240 or run
> /usr/local/freeswitch/bin/freeswitch -waste.
> auto-adjusting stack size for optimal performance...
> Cannot open pid file /usr/local/freeswitch/run/freeswitch.pid.
> 
> If anyone can help, that would be truly appreciated.
> 
> PeterS
> 



From emss.mail at gmail.com  Sun Oct  3 11:55:24 2010
From: emss.mail at gmail.com (Eric Masson)
Date: Sun, 03 Oct 2010 20:55:24 +0200
Subject: [Freeswitch-users] sip to spa3102 fallback
Message-ID: <i8ajiu$2ce$1@dough.gmane.org>

Hello,

I have two sip providers configuration files in 
$(fs_conf}/conf/diaplan/default :
- 00_keyyo.net.xml
<include>
   <extension name="keyyo" continue="false">
     <condition field="destination_number" expression = "^(.*)$">
       <action application="set" data="effective_caller_id_name=**MyID**"/>
       <action application="set" 
data="effective_caller_id_number=**Myphonenumber"/>
       <action application="bridge" data="sofia/gateway/keyyo.net/$1"/>
     </condition>
   </extension>
</include>

- 00_spa3102.xml
<include>
   <extension name="To PSTN">
     <condition field="destination_number" expression="^(.*)$">
       <action application="bridge" 
data="sofia/internal/${destination_number}@spafenmsf:5061" />
     </condition>
   </extension>
</include>

I'd like to implement the following scenario for outbound calls :
- dial using "keyyo" if available
- fallback to "To PSTN" if keyyo isn't available (not registered, 
already used, for example)

I've googled for freeswitch failover setups but didn't find anything 
that could have enlightened me (I'm kind of new in fs).

Where could I find docs to achieve the exposed goal please ?

Kind Regards

Eric Masson



From steveayre at gmail.com  Sun Oct  3 15:06:50 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 3 Oct 2010 23:06:50 +0100
Subject: [Freeswitch-users] g729 passthrough mode
In-Reply-To: <8F6FC0E5-B0A2-4AAF-8489-41EA35A2C60F@ipeva.fr>
References: <COL109-W13224EAAAE107BFB3C60A1F46A0@phx.gbl>
	<AANLkTinmjuB9WoW-S4WuoccjBdjLhPnw6GPyQPWocgtN@mail.gmail.com>
	<1286116469745-5596240.post@n2.nabble.com>
	<8F6FC0E5-B0A2-4AAF-8489-41EA35A2C60F@ipeva.fr>
Message-ID: <AANLkTikYyehAJbJfMAoE6Op7WJMUHGJTPNzgy10B=Z2n@mail.gmail.com>

"I think Steven meant that no transcoding was required at the FS
level, which was the point of Abid's question."

Yes, to show it's specifically the lack of encoder/decoder that means
no license is required and that it's the raw data from the client
that's sent straight through in passthrough. Which isn't quite how it
works with other codecs where you can attach media bugs etc which you
can't do with a passthrough codec.


"And the main disadvantage of using bypass media is probably the NAT
handling, and the lack of topology hiding."

Plus in a lot of cases firewalls may not allow it.

Example 1:
We have customers -> FS -> gateways. Gateways are reachable by FS but
not from customers (due to IP ACL on gateway firewalls).

Example 2:
We have also have customers -> FS -> gateways. Gateways are on LAN, so
reachable from FS but not from customers.

In both those cases bypass media wouldn't work, since there's no way
for traffic to go between the gateways, but FS can't know that so the
call would be setup but with no audio.

-Steve




On 3 October 2010 21:06, David Ponzone <david.ponzone at ipeva.fr> wrote:
> Mazilo,
> I think Steven meant that no transcoding was required at the FS level, which
> was the point of Abid's question.
> And the main disadvantage of using bypass media is probably the NAT
> handling, and the lack of topology hiding.
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 03/10/2010 ? 16:34, mazilo a ?crit :
>
>
> Steven Ayre wrote:
>
> If G729 is at both ends, then no encoding/decoding is required as the
>
> raw data can be sent straight through. This is passthrough mode.
>
> Because there is no encoding/decoding stage, there is no licence
>
> required in this case.
>
> A more appropriate way to say this is no transcoding. Encoding/decoding
> processes are still done at each endpoints on the device, i.e IP Phone, ATA,
> and/or computer (with softphone), and a G729 license is still required when
> the call uses the G729 CoDec (which is mostly paid when one acquired the
> device with a built-in G729 CoDec). So, to say no encoding/deconding
> involved and no license required are really misleading and definitely
> incorrect.
>
> BTW, one can also configure his/her FS system with bypass media option to
> ensure the FS only does the SIP with no media handling. The advantage using
> this approach is to leave FS alone and free up some of its networking
> resounrces to handle the media during a call. There is also some
> disadvantages, i.e. no MOH, etc.
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/g729-passthrough-mode-tp5594193p5596240.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From peter.schrock at gmail.com  Sun Oct  3 16:00:38 2010
From: peter.schrock at gmail.com (Peter Schrock)
Date: Sun, 3 Oct 2010 16:00:38 -0700
Subject: [Freeswitch-users] pid file can't open
In-Reply-To: <201010031716.45863.sos@sokhapkin.dyndns.org>
References: <AANLkTinCfw5N2P7gXsKVpd_m0GZTfDROJXpY9TSzr2qo@mail.gmail.com>
	<201010031716.45863.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTin+FtKcf6x=b-8Ntcxj4iOiqzZVefYZMNdZFqkk@mail.gmail.com>

actually, neither is the case. I have found on the file "freeswitch.pid" and
I get the error message in root.

Peter

On Sun, Oct 3, 2010 at 2:16 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org>wrote:

> Either directory /usr/local/freeswitch/run doesn't exists, or FS is started
> with non-root privileges and has no write access to the directory or
> already
> existing freeswitch.pid file.
>
> On Sunday 03 October 2010, Peter Schrock wrote:
> > I loaded fink and macports to be able to access binaries to help with my
> >  use of freeswitch. come to find out, fink and macports are not
> supported.
> >  I tried uninstalling fink and macports and now I am having trouble with
> >  freeswitch and opening the freeswitch.pid file. Does anyone have any
> >  suggestions on what I should do? Here is the error message I keep
> getting:
> >
> > Error: stacksize 65532 is too large: run ulimit -s 240 or run
> > /usr/local/freeswitch/bin/freeswitch -waste.
> > auto-adjusting stack size for optimal performance...
> > Cannot open pid file /usr/local/freeswitch/run/freeswitch.pid.
> >
> > If anyone can help, that would be truly appreciated.
> >
> > PeterS
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From peter.schrock at gmail.com  Sun Oct  3 16:03:42 2010
From: peter.schrock at gmail.com (Peter Schrock)
Date: Sun, 3 Oct 2010 16:03:42 -0700
Subject: [Freeswitch-users] pid file can't open
In-Reply-To: <AANLkTin+FtKcf6x=b-8Ntcxj4iOiqzZVefYZMNdZFqkk@mail.gmail.com>
References: <AANLkTinCfw5N2P7gXsKVpd_m0GZTfDROJXpY9TSzr2qo@mail.gmail.com>
	<201010031716.45863.sos@sokhapkin.dyndns.org>
	<AANLkTin+FtKcf6x=b-8Ntcxj4iOiqzZVefYZMNdZFqkk@mail.gmail.com>
Message-ID: <AANLkTinV9Ui0VAXyG7fw2vDZy0JrXhBd3GK-rp1N4fjo@mail.gmail.com>

to be more accurate, this the error I get when trying to access fs_cli.

peter

On Sun, Oct 3, 2010 at 4:00 PM, Peter Schrock <peter.schrock at gmail.com>wrote:

> actually, neither is the case. I have found on the file "freeswitch.pid"
> and I get the error message in root.
>
> Peter
>
>
> On Sun, Oct 3, 2010 at 2:16 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org>wrote:
>
>> Either directory /usr/local/freeswitch/run doesn't exists, or FS is
>> started
>> with non-root privileges and has no write access to the directory or
>> already
>> existing freeswitch.pid file.
>>
>> On Sunday 03 October 2010, Peter Schrock wrote:
>> > I loaded fink and macports to be able to access binaries to help with my
>> >  use of freeswitch. come to find out, fink and macports are not
>> supported.
>> >  I tried uninstalling fink and macports and now I am having trouble with
>> >  freeswitch and opening the freeswitch.pid file. Does anyone have any
>> >  suggestions on what I should do? Here is the error message I keep
>> getting:
>> >
>> > Error: stacksize 65532 is too large: run ulimit -s 240 or run
>> > /usr/local/freeswitch/bin/freeswitch -waste.
>> > auto-adjusting stack size for optimal performance...
>> > Cannot open pid file /usr/local/freeswitch/run/freeswitch.pid.
>> >
>> > If anyone can help, that would be truly appreciated.
>> >
>> > PeterS
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From peter.schrock at gmail.com  Sun Oct  3 16:09:51 2010
From: peter.schrock at gmail.com (Peter Schrock)
Date: Sun, 3 Oct 2010 16:09:51 -0700
Subject: [Freeswitch-users] pid file can't open
In-Reply-To: <AANLkTinV9Ui0VAXyG7fw2vDZy0JrXhBd3GK-rp1N4fjo@mail.gmail.com>
References: <AANLkTinCfw5N2P7gXsKVpd_m0GZTfDROJXpY9TSzr2qo@mail.gmail.com>
	<201010031716.45863.sos@sokhapkin.dyndns.org>
	<AANLkTin+FtKcf6x=b-8Ntcxj4iOiqzZVefYZMNdZFqkk@mail.gmail.com>
	<AANLkTinV9Ui0VAXyG7fw2vDZy0JrXhBd3GK-rp1N4fjo@mail.gmail.com>
Message-ID: <AANLkTikYTtmQTHRUqv2HRT4=mddDU3SnVFoMPuVGG-9m@mail.gmail.com>

okay, so I figured out the last error, but now, when I get fs running in the
background and try to access fs_cli, this is what I get:

dyld: Library not loaded: /opt/local/lib/libssl.1.0.0.dylib
  Referenced from: /usr/local/bin/fs_cli
  Reason: image not found
Trace/BPT trap

Any thoughts?

PeterS

On Sun, Oct 3, 2010 at 4:03 PM, Peter Schrock <peter.schrock at gmail.com>wrote:

> to be more accurate, this the error I get when trying to access fs_cli.
>
> peter
>
>
> On Sun, Oct 3, 2010 at 4:00 PM, Peter Schrock <peter.schrock at gmail.com>wrote:
>
>> actually, neither is the case. I have found on the file "freeswitch.pid"
>> and I get the error message in root.
>>
>> Peter
>>
>>
>> On Sun, Oct 3, 2010 at 2:16 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org
>> > wrote:
>>
>>> Either directory /usr/local/freeswitch/run doesn't exists, or FS is
>>> started
>>> with non-root privileges and has no write access to the directory or
>>> already
>>> existing freeswitch.pid file.
>>>
>>> On Sunday 03 October 2010, Peter Schrock wrote:
>>> > I loaded fink and macports to be able to access binaries to help with
>>> my
>>> >  use of freeswitch. come to find out, fink and macports are not
>>> supported.
>>> >  I tried uninstalling fink and macports and now I am having trouble
>>> with
>>> >  freeswitch and opening the freeswitch.pid file. Does anyone have any
>>> >  suggestions on what I should do? Here is the error message I keep
>>> getting:
>>> >
>>> > Error: stacksize 65532 is too large: run ulimit -s 240 or run
>>> > /usr/local/freeswitch/bin/freeswitch -waste.
>>> > auto-adjusting stack size for optimal performance...
>>> > Cannot open pid file /usr/local/freeswitch/run/freeswitch.pid.
>>> >
>>> > If anyone can help, that would be truly appreciated.
>>> >
>>> > PeterS
>>> >
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
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From mario_fs at mgtech.com  Sun Oct  3 16:21:45 2010
From: mario_fs at mgtech.com (Mario)
Date: Sun, 03 Oct 2010 16:21:45 -0700
Subject: [Freeswitch-users] Email fails due to FROM not set as needed
Message-ID: <4CA91009.4020203@mgtech.com>

Email from FreeSwitch does not work because the relay server does not
recognize the FROM: 1000 at 1.2.3.4 address. The Postfix log shows the FS
email going out but rejected due to a bad FROM address. I could not find
anything on this so does anyone know how to get FS to use a preset FROM
email address?

I have Postfix (Sendmail look alike) running and tested. FreeSwitch is
set to handle emails in switch.conf, and the correct parms are in the
user settings. I need to control the FROM address FS uses. Thanks.
Mario


From david.ponzone at ipeva.fr  Sun Oct  3 16:57:01 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 4 Oct 2010 01:57:01 +0200
Subject: [Freeswitch-users] Email fails due to FROM not set as needed
In-Reply-To: <4CA91009.4020203@mgtech.com>
References: <4CA91009.4020203@mgtech.com>
Message-ID: <5CD4155D-45A2-43E8-86A6-77CD74F80A06@ipeva.fr>

Mario,

i think you need to read the wiki a little, to read ALL FS's default  
config files, and (sorry, I gonna be sarcastic) to learn how to use  
grep :)

grep mail conf/* and grep mail conf/autoload_configs/* may give you  
some answers:
conf/notify-voicemail.tpl
conf/autoload_configs/voicemail.conf.xml

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 04/10/2010 ? 01:21, Mario a ?crit :

> Email from FreeSwitch does not work because the relay server does not
> recognize the FROM: 1000 at 1.2.3.4 address. The Postfix log shows the FS
> email going out but rejected due to a bad FROM address. I could not  
> find
> anything on this so does anyone know how to get FS to use a preset  
> FROM
> email address?
>
> I have Postfix (Sendmail look alike) running and tested. FreeSwitch is
> set to handle emails in switch.conf, and the correct parms are in the
> user settings. I need to control the FROM address FS uses. Thanks.
> Mario
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From peder at networkoblivion.com  Sun Oct  3 17:37:16 2010
From: peder at networkoblivion.com (Peder)
Date: Sun, 3 Oct 2010 19:37:16 -0500
Subject: [Freeswitch-users] Email fails due to FROM not set as needed
In-Reply-To: <4CA91009.4020203@mgtech.com>
References: <4CA91009.4020203@mgtech.com>
Message-ID: <04a001cb635c$4b8c7b70$e2a57250$@com>

Edit voicemail.conf.xml.

Under the <email> section, change "email-from".

<param name="email-from" value="blah at blah.com"/>

I changed the email template and it didn't work, so I changed it in
voicemail.conf.xml and that fixed it.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mario
Sent: Sunday, October 03, 2010 6:22 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Email fails due to FROM not set as needed

Email from FreeSwitch does not work because the relay server does not
recognize the FROM: 1000 at 1.2.3.4 address. The Postfix log shows the FS
email going out but rejected due to a bad FROM address. I could not find
anything on this so does anyone know how to get FS to use a preset FROM
email address?

I have Postfix (Sendmail look alike) running and tested. FreeSwitch is
set to handle emails in switch.conf, and the correct parms are in the
user settings. I need to control the FROM address FS uses. Thanks.
Mario

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



From mario_fs at mgtech.com  Sun Oct  3 17:48:01 2010
From: mario_fs at mgtech.com (Mario)
Date: Sun, 03 Oct 2010 17:48:01 -0700
Subject: [Freeswitch-users] Email fails due to FROM not set as needed
In-Reply-To: <5CD4155D-45A2-43E8-86A6-77CD74F80A06@ipeva.fr>
References: <4CA91009.4020203@mgtech.com>
	<5CD4155D-45A2-43E8-86A6-77CD74F80A06@ipeva.fr>
Message-ID: <4CA92441.30009@mgtech.com>

Thanks a lot for the grep, I did work for 3 hours before posting. I
found these 3 areas, item 1 does not work (would have been my preference
since I am sending due to different reasons and would like different
user ids in from). Item 2 had no effect. Item 3 finally worked but now I
am blocked due "550 Administrative prohibition (in reply to end of DATA
command): which is not an FS issue that looks like I can't get past it
due to the the hosting company (bluehost). Thanks a lot for your quick help!

1. notify-voicemail.tpl:
From: "${voicemail_caller_id_name}"
<${voicemail_caller_id_number}@mydomain.com

2. voicemail.tpl:
From: "${voicemail_caller_id_name}"
<${voicemail_caller_id_number}@mydomain.com

3.voicemail.xml
 <param name="email-from" value="phones at mydomain.com"/

On 10/03/10 16:57, David Ponzone wrote:
> Mario,
> 
> i think you need to read the wiki a little, to read ALL FS's default
> config files, and (sorry, I gonna be sarcastic) to learn how to use grep :)
> 
> grep mail conf/* and grep mail conf/autoload_configs/* may give you some
> answers:
> conf/notify-voicemail.tpl
> conf/autoload_configs/voicemail.conf.xml
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com
> <BLOCKED::http://www.ipeva-studio.com/>
> 
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion non autoris?e est interdite. Tout message ?lectronique est
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous
> n'?tes pas destinataire de ce message, merci de le d?truire
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
> 
> 
> 
> Le 04/10/2010 ? 01:21, Mario a ?crit :
> 
>> Email from FreeSwitch does not work because the relay server does not
>> recognize the FROM: 1000 at 1.2.3.4 <mailto:1000 at 1.2.3.4> address. The
>> Postfix log shows the FS
>> email going out but rejected due to a bad FROM address. I could not find
>> anything on this so does anyone know how to get FS to use a preset FROM
>> email address?
>>
>> I have Postfix (Sendmail look alike) running and tested. FreeSwitch is
>> set to handle emails in switch.conf, and the correct parms are in the
>> user settings. I need to control the FROM address FS uses. Thanks.
>> Mario
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From mario_fs at mgtech.com  Sun Oct  3 17:48:34 2010
From: mario_fs at mgtech.com (Mario)
Date: Sun, 03 Oct 2010 17:48:34 -0700
Subject: [Freeswitch-users] Email fails due to FROM not set as needed
In-Reply-To: <04a001cb635c$4b8c7b70$e2a57250$@com>
References: <4CA91009.4020203@mgtech.com> <04a001cb635c$4b8c7b70$e2a57250$@com>
Message-ID: <4CA92462.3070300@mgtech.com>

Thanks a lot, see my previous post.

On 10/03/10 17:37, Peder wrote:
> Edit voicemail.conf.xml.
> 
> Under the <email> section, change "email-from".
> 
> <param name="email-from" value="blah at blah.com"/>
> 
> I changed the email template and it didn't work, so I changed it in
> voicemail.conf.xml and that fixed it.
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mario
> Sent: Sunday, October 03, 2010 6:22 PM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] Email fails due to FROM not set as needed
> 
> Email from FreeSwitch does not work because the relay server does not
> recognize the FROM: 1000 at 1.2.3.4 address. The Postfix log shows the FS
> email going out but rejected due to a bad FROM address. I could not find
> anything on this so does anyone know how to get FS to use a preset FROM
> email address?
> 
> I have Postfix (Sendmail look alike) running and tested. FreeSwitch is
> set to handle emails in switch.conf, and the correct parms are in the
> user settings. I need to control the FROM address FS uses. Thanks.
> Mario
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From mario_fs at mgtech.com  Sun Oct  3 18:05:22 2010
From: mario_fs at mgtech.com (Mario)
Date: Sun, 03 Oct 2010 18:05:22 -0700
Subject: [Freeswitch-users] Email fails due to FROM not set as needed
In-Reply-To: <4CA92441.30009@mgtech.com>
References: <4CA91009.4020203@mgtech.com>	<5CD4155D-45A2-43E8-86A6-77CD74F80A06@ipeva.fr>
	<4CA92441.30009@mgtech.com>
Message-ID: <4CA92852.3090701@mgtech.com>

Whoops made an error and left an item out to here is udpate in case
others can use it. I got the right FROM but I think the RCPT is bad
since I am blocked due "550 Administrative prohibition (in reply to end
of DATA command)" Not sure if it's an FS issue, I can't get past it due
to the the hosting company (bluehost).

Thanks a lot for the grep, I did work for 3 hours before posting. I
found these 4 areas, item 1 does not work (would have been my preference
since I am sending due to different reasons and would like different
user ids in from). Item 2 and 3 had no effect. Item 4 finally worked but
now I  Thanks a lot for your quick help!

1. userid.xml (found in wiki)
 <variable name="voicemail_domain" value="mydomain.com"/>

2. notify-voicemail.tpl:
From: "${voicemail_caller_id_name}"
<${voicemail_caller_id_number}@mydomain.com

3. voicemail.tpl:
From: "${voicemail_caller_id_name}"
<${voicemail_caller_id_number}@mydomain.com

4. voicemail.xml
 <param name="email-from" value="phones at mydomain.com"/



On 10/03/10 17:48, Mario wrote:
> Thanks a lot for the grep, I did work for 3 hours before posting. I
> found these 3 areas, item 1 does not work (would have been my preference
> since I am sending due to different reasons and would like different
> user ids in from). Item 2 had no effect. Item 3 finally worked but now I
> am blocked due "550 Administrative prohibition (in reply to end of DATA
> command): which is not an FS issue that looks like I can't get past it
> due to the the hosting company (bluehost). Thanks a lot for your quick help!
> 
> 1. notify-voicemail.tpl:
> From: "${voicemail_caller_id_name}"
> <${voicemail_caller_id_number}@mydomain.com
> 
> 2. voicemail.tpl:
> From: "${voicemail_caller_id_name}"
> <${voicemail_caller_id_number}@mydomain.com
> 
> 3.voicemail.xml
>  <param name="email-from" value="phones at mydomain.com"/
> 
> On 10/03/10 16:57, David Ponzone wrote:
>> Mario,
>>
>> i think you need to read the wiki a little, to read ALL FS's default
>> config files, and (sorry, I gonna be sarcastic) to learn how to use grep :)
>>
>> grep mail conf/* and grep mail conf/autoload_configs/* may give you some
>> answers:
>> conf/notify-voicemail.tpl
>> conf/autoload_configs/voicemail.conf.xml
>>
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com
>> <BLOCKED::http://www.ipeva-studio.com/>
>>
>> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou
>> diffusion non autoris?e est interdite. Tout message ?lectronique est
>> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au
>> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous
>> n'?tes pas destinataire de ce message, merci de le d?truire
>> imm?diatement et d'avertir l'exp?diteur./
>> /
>> /
>>
>>
>>
>> Le 04/10/2010 ? 01:21, Mario a ?crit :
>>
>>> Email from FreeSwitch does not work because the relay server does not
>>> recognize the FROM: 1000 at 1.2.3.4 <mailto:1000 at 1.2.3.4> address. The
>>> Postfix log shows the FS
>>> email going out but rejected due to a bad FROM address. I could not find
>>> anything on this so does anyone know how to get FS to use a preset FROM
>>> email address?
>>>
>>> I have Postfix (Sendmail look alike) running and tested. FreeSwitch is
>>> set to handle emails in switch.conf, and the correct parms are in the
>>> user settings. I need to control the FROM address FS uses. Thanks.
>>> Mario
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From frnkblk at iname.com  Sun Oct  3 20:05:57 2010
From: frnkblk at iname.com (Frank Bulk - iName.com)
Date: Sun, 3 Oct 2010 22:05:57 -0500
Subject: [Freeswitch-users] Anyone know a LRN provider?
In-Reply-To: <201009281110.29649.sos@sokhapkin.dyndns.org>
References: <AANLkTi=6HnKSLt4_n=bjKCTKwC48sFYFDFy_4kVMBg74@mail.gmail.com>
	<201009281110.29649.sos@sokhapkin.dyndns.org>
Message-ID: <!&!AAAAAAAAAAAuAAAAAAAAAKTyXRN5/+lGvU59a+P7CFMBAN6gY+ZG84BMpVQcAbDh1IQAAAATbSgAABAAAAAsbjxtu0R+Srydj1U8cvlDAQAAAAA=@iname.com>

tnid.org may be worth looking at, for incidental non-API use.

Frank

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey
Okhapkin
Sent: Tuesday, September 28, 2010 10:10 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Anyone know a LRN provider?

http://www.callwithus.com/API#lrn

On Tuesday 28 September 2010, Essobi wrote:
> Anyone know a LRN provider?  Trying to find someone who provides access to
> the Local Number Portability lookup database to see if a number is ported,
> and if so, what LRN to use.
> 


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



From freedomyug at gmail.com  Sun Oct  3 21:45:59 2010
From: freedomyug at gmail.com (Gourav Shah)
Date: Mon, 4 Oct 2010 10:15:59 +0530
Subject: [Freeswitch-users] Announcing FreeSWITCHBOX 2.0.2
Message-ID: <AANLkTikzbxxeusit2SpQKckAM76RXMs6HPZa_=BSvJYm@mail.gmail.com>

Announcement Page:
http://www.initcron.org/voip-2/announcing-freeswitchbox-v2-0-2/


Initcron Labs announces  freeSWITCHBOX version 2.0.2 , a VoIP in a box
appliance.  This release comes in a quick succession  after a week of  its
first release 1.0.1.  Reason for  bumping the release version to 2* is the
major changes in its design and packages included.  The features included in
this version are,

   1.
      1.
         1. Added  blue.box web interface from
2600Hz<http://www.2600hz.org/> project.
         Team at 2600Hz has done a excellent work designing and
developing this web
         management interface for freeswitch.  We still have  freePBX
v3  included
         with the appliance, however strongly recommend using blue.box
as it is more
         stable.
         2. Added automated skype installer.  Start appliance, go to the web
         interface at http://HOSTNAME  and you?ll see a link to install
         skype.  Once you read and accept skype EULA, you could
install skype in one
         click.  However, be patient while the script downloads skype
and sets it up.
          As off now its a barebone UI that we have written and will
be enhanced in
         future.  To make skype work with freeswitch, you still need
to follow the
         steps from theFreeSWITCH mod_skypeopen
Wiki<http://wiki.freeswitch.org/wiki/Mod_skypopen#Configuration_and_Operation>
and
         configure accordingly.  We aim to automate this as well in near future.
         3. Updated cherokee server timeout to 1500 seconds  to allow skype
         installation
         4. Enabled shell login for wwwrun user which owns freeswitch files
         and web management files
         5. Included FreeSWITCH sound files  (sounds-install, moh-install).

This appliance is been created with Suse
Studio<http://www.initcron.org/sysadmin/build-custom-linux-in-minutes-suse-studio/>.
To download we provide  two options,
1. Suse Gallery Page  :
http://susegallery.com/a/Kr7Ayv/freeswitchbox<http://susegallery.com/a/Kr7Ayv/freeswitchbox>
2. Sourceforge Page  :
http://sourceforge.net/projects/freeswitchbox/files/<http://sourceforge.net/projects/freeswitchbox/files/>


Thanks
GS
Check out   freeSWITCHBOX <http://www.initcron.org/freeswitchbox/> description
page to know more about this appliance.
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From tekkye at gmail.com  Sun Oct  3 23:51:09 2010
From: tekkye at gmail.com (Tech Man)
Date: Mon, 4 Oct 2010 01:51:09 -0500
Subject: [Freeswitch-users] Call Rejected Issue
In-Reply-To: <69B2390F-D94F-4490-B4B4-B461189E3FA8@freeswitch.org>
References: <AANLkTikBSri2YDAY8FBVR4oiP5qVQcN_KeqwCpCVCdPH@mail.gmail.com>
	<69B2390F-D94F-4490-B4B4-B461189E3FA8@freeswitch.org>
Message-ID: <AANLkTinMudz_S_X5THNK5PzUoJpfz==Ws3kvF-ohRorN@mail.gmail.com>

Thanks.  It seems that outbound call to PSTN is not supported by
iptel/ipkall.  Inbound is working though.  So I'm now trying to configure
the outbound call through google voice but it's not connecting. The source
code was downloaded yesterday via git:
git clone git://git.freeswitch.org/freeswitch.git

I pulled some other information below but it didn't help me to find the
cause.  Any suggestions for next steps?  thanks

Sam

*****************************************************************************

--DingaLing status--
login    |    connected
user at gmail.com/talk    |    UNCONNECTED

*****************************************************************************

conf/jingle_profiles/client.xml

<include>
  <!-- Client Profile (Original mode) -->
  <!-- to use this profile take the x- away from the open and close tags so
its <profile> and </profile> -->
  <profile type="client">
    <param name="name" value="$${xmpp_client_profile}"/>
    <param name="login" value="user at gmail.com/talk"/>
    <param name="password" value="password"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="message" value="Jingle all the way"/>
    <param name="rtp-ip" value="$${bind_server_ip}"/>
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="auto-login" value="true"/>
    <!-- SASL "plain" or "md5" -->
    <param name="sasl" value="plain"/>
    <!-- if the server where the jabber is hosted is not the same as the one
in the jid -->
    <param name="server" value="talk.google.com"/>
    <!-- Enable TLS or not -->
    <param name="tls" value="true"/>
    <!-- disable to trade async for more calls -->
    <param name="use-rtp-timer" value="true"/>
    <!-- default extension (if one cannot be determined) -->
    <param name="exten" value="888"/>
    <!-- VAD choose one -->
    <!-- <param name="vad" value="in"/> -->
    <!-- <param name="vad" value="out"/> -->
    <param name="vad" value="both"/>
    <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
    <!--<param name="candidate-acl" value="wan.auto"/>-->
    <param name="local-network-acl" value="localnet.auto"/>
  </profile>
</include>

*****************************************************************************

Enabling debug on dingaling.conf.xml:

2010-10-04 00:59:22.243782 [NOTICE] libdingaling.c:1368 SEND:
-------------------------------------------------------------------------------
<stream:stream xmlns:stream="http://etherx.jabber.org/streams"
xmlns="jabber:client" to="gmail.com" version="1.0"></stream:stream>

2010-10-04 00:59:22.277462 [INFO] libdingaling.c:1366 RECV:
-------------------------------------------------------------------------------
<stream:stream from="gmail.com" id="9EAEB6231C3BDA1A" version="1.0"
xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:client">
  <stream:features>
    <starttls xmlns="urn:ietf:params:xml:ns:xmpp-tls">
      <required></required>
    </starttls>
    <mechanisms xmlns="urn:ietf:params:xml:ns:xmpp-sasl">
      <mechanism>X-GOOGLE-TOKEN</mechanism>
    </mechanisms>
  </stream:features>
</stream:stream>



On Fri, Oct 1, 2010 at 9:11 AM, Brian West <brian at freeswitch.org> wrote:

> The far side rejected the call turn on the sip trace and watch it... It'll
> be rather clear what is going on if you run "sofia profile external siptrace
> on"
>
> /b
>
> On Oct 1, 2010, at 1:12 AM, Tech Man wrote:
>
> > I'm trying to configure an iptel SIP account but everytime I call an
> external number, it gets rejected.  I'm running FreeSwitch 1.0.6 on a Mac OS
> X 10.5.8.  Any help would be greatly appreciated.  Thanks
> >
> > Sam
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From elihayunfs at gmail.com  Mon Oct  4 01:36:13 2010
From: elihayunfs at gmail.com (Eli Hayun)
Date: Mon, 04 Oct 2010 10:36:13 +0200
Subject: [Freeswitch-users] No autoanswer with Originate call
Message-ID: <1286181373.20572.5.camel@localhost>

Hi
When I am dialing from A phone to B phone and I do :

      <action application="set" data="sip_auto_answer=true" />
       <action application="set" data="sip_h_Call-Info=<sip:
${domain_name}>;answer-after=2" />

everything is OK

When I try from lua script 
api:executeString("originate [originate string] AUTO_ANS xml default")

and in the dialplan I put the set autoanswer=true it is not working.

any help?

Eli
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From tjardick at vanderkraan.net  Mon Oct  4 02:46:28 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Mon, 4 Oct 2010 11:46:28 +0200
Subject: [Freeswitch-users] In dialogue SIP Message
Message-ID: <B8B7202C-12C4-42C2-9A7C-11EF1086803E@vanderkraan.net>

Hi,

I'm looking for a way to send SIP MESSAGE but within the current call dialogue (call-id).

I know that via the chat API I can send messages but as they are without the context of the current call this doesn't work for me.

So like the SNOM phones being able to send record: on within a established call i'd like to send SIP Messages from freeswitch within the current call to the remote destination.

Can this be done and could anyone provide me with ESL/API command to perform this action ?

Thanks in advance.

Regards,

Tjardick



From riedinger at sns.eu  Mon Oct  4 03:58:20 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Mon, 04 Oct 2010 12:58:20 +0200
Subject: [Freeswitch-users] Substitution of mod_g729 by mod_com_g729 causes
	core dump
Message-ID: <4CA9B34C.5070400@sns.eu>

  I just tried to subsitute mod_g729 by mod_com_g729 by using an 
appropriate moduls.conf.xml. After a restart of Freeswitch, it crahses, 
when the first inbound call gets processed:

freeswitch at sns-fs1.sns.eu> 2010-10-04 12:32:27.064152 [NOTICE] 
switch_channel.c:779 New Channel sofia/external/xyz at a.b.c.d 
[8e94789e-5a8d-4088-8ad1-cd5fb2be0c4f]
freeswitch: src/switch_core_memory.c:448: switch_core_perform_alloc: 
Assertion `pool != ((void *)0)' failed.
Aborted (core dumped)

There seems to be no error, when mod_com_g729 gets loaded:
2010-10-04 12:32:21.856829 [INFO] mod_com_g729.c:243 Permitted G.729AB 
channels: 20
2010-10-04 12:32:21.856920 [NOTICE] switch_loadable_module.c:185 Adding 
Codec G729 18 G.729 8000hz 10ms 8000bps
2010-10-04 12:32:21.856925 [NOTICE] switch_loadable_module.c:185 Adding 
Codec G729 18 G.729 8000hz 20ms 8000bps
2010-10-04 12:32:21.856928 [NOTICE] switch_loadable_module.c:185 Adding 
Codec G729 18 G.729 8000hz 30ms 8000bps
2010-10-04 12:32:21.856933 [NOTICE] switch_loadable_module.c:185 Adding 
Codec G729 18 G.729 8000hz 40ms 8000bps
2010-10-04 12:32:21.856937 [NOTICE] switch_loadable_module.c:185 Adding 
Codec G729 18 G.729 8000hz 50ms 8000bps
2010-10-04 12:32:21.856943 [NOTICE] switch_loadable_module.c:185 Adding 
Codec G729 18 G.729 8000hz 60ms 8000bps
2010-10-04 12:32:21.856947 [NOTICE] switch_loadable_module.c:185 Adding 
Codec G729 18 G.729 8000hz 70ms 8000bps
2010-10-04 12:32:21.856974 [NOTICE] switch_loadable_module.c:185 Adding 
Codec G729 18 G.729 8000hz 80ms 8000bps
2010-10-04 12:32:21.856978 [NOTICE] switch_loadable_module.c:185 Adding 
Codec G729 18 G.729 8000hz 90ms 8000bps
2010-10-04 12:32:21.856983 [NOTICE] switch_loadable_module.c:185 Adding 
Codec G729 18 G.729 8000hz 100ms 8000bps
2010-10-04 12:32:21.856988 [NOTICE] switch_loadable_module.c:185 Adding 
Codec G729 18 G.729 8000hz 110ms 8000bps
2010-10-04 12:32:21.856993 [NOTICE] switch_loadable_module.c:185 Adding 
Codec G729 18 G.729 8000hz 120ms 8000bps
2010-10-04 12:32:21.857144 [NOTICE] switch_loadable_module.c:274 Adding 
API Function 'g729_count'
2010-10-04 12:32:21.857175 [NOTICE] switch_loadable_module.c:274 Adding 
API Function 'g729_used'
2010-10-04 12:32:21.857199 [NOTICE] switch_loadable_module.c:274 Adding 
API Function 'g729_available'
2010-10-04 12:32:21.857229 [NOTICE] switch_loadable_module.c:274 Adding 
API Function 'g729_info'



I'm using
FreeSWITCH Version 1.0.head (git-758f383 2010-09-30 13-30-36 -0700)

If I use mod_g729 instead of mod_com_g729 Freeswitch is running stable.

Does anybody have any ideas how to debug this issue?

Thank you in advance
     Jan


-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                         E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973

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From david.ponzone at ipeva.fr  Mon Oct  4 04:33:12 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 4 Oct 2010 13:33:12 +0200
Subject: [Freeswitch-users] Substitution of mod_g729 by mod_com_g729
	causes core dump
In-Reply-To: <4CA9B34C.5070400@sns.eu>
References: <4CA9B34C.5070400@sns.eu>
Message-ID: <1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>

Jan,

which version of mod_com_g729 and FS are you using ?
There were very recent updates to the codec, that may require to be  
used with the latest git version of FS.

Brian West should able to confirm that later in the day.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 04/10/2010 ? 12:58, Jan Riedinger a ?crit :

> I just tried to subsitute mod_g729 by mod_com_g729 by using an  
> appropriate moduls.conf.xml. After a restart of Freeswitch, it  
> crahses, when the first inbound call gets processed:
>
> freeswitch at sns-fs1.sns.eu> 2010-10-04 12:32:27.064152 [NOTICE]  
> switch_channel.c:779 New Channel sofia/external/xyz at a.b.c.d  
> [8e94789e-5a8d-4088-8ad1-cd5fb2be0c4f]
> freeswitch: src/switch_core_memory.c:448: switch_core_perform_alloc:  
> Assertion `pool != ((void *)0)' failed.
> Aborted (core dumped)
>
> There seems to be no error, when mod_com_g729 gets loaded:
> 2010-10-04 12:32:21.856829 [INFO] mod_com_g729.c:243 Permitted G. 
> 729AB channels: 20
> 2010-10-04 12:32:21.856920 [NOTICE] switch_loadable_module.c:185  
> Adding Codec G729 18 G.729 8000hz 10ms 8000bps
> 2010-10-04 12:32:21.856925 [NOTICE] switch_loadable_module.c:185  
> Adding Codec G729 18 G.729 8000hz 20ms 8000bps
> 2010-10-04 12:32:21.856928 [NOTICE] switch_loadable_module.c:185  
> Adding Codec G729 18 G.729 8000hz 30ms 8000bps
> 2010-10-04 12:32:21.856933 [NOTICE] switch_loadable_module.c:185  
> Adding Codec G729 18 G.729 8000hz 40ms 8000bps
> 2010-10-04 12:32:21.856937 [NOTICE] switch_loadable_module.c:185  
> Adding Codec G729 18 G.729 8000hz 50ms 8000bps
> 2010-10-04 12:32:21.856943 [NOTICE] switch_loadable_module.c:185  
> Adding Codec G729 18 G.729 8000hz 60ms 8000bps
> 2010-10-04 12:32:21.856947 [NOTICE] switch_loadable_module.c:185  
> Adding Codec G729 18 G.729 8000hz 70ms 8000bps
> 2010-10-04 12:32:21.856974 [NOTICE] switch_loadable_module.c:185  
> Adding Codec G729 18 G.729 8000hz 80ms 8000bps
> 2010-10-04 12:32:21.856978 [NOTICE] switch_loadable_module.c:185  
> Adding Codec G729 18 G.729 8000hz 90ms 8000bps
> 2010-10-04 12:32:21.856983 [NOTICE] switch_loadable_module.c:185  
> Adding Codec G729 18 G.729 8000hz 100ms 8000bps
> 2010-10-04 12:32:21.856988 [NOTICE] switch_loadable_module.c:185  
> Adding Codec G729 18 G.729 8000hz 110ms 8000bps
> 2010-10-04 12:32:21.856993 [NOTICE] switch_loadable_module.c:185  
> Adding Codec G729 18 G.729 8000hz 120ms 8000bps
> 2010-10-04 12:32:21.857144 [NOTICE] switch_loadable_module.c:274  
> Adding API Function 'g729_count'
> 2010-10-04 12:32:21.857175 [NOTICE] switch_loadable_module.c:274  
> Adding API Function 'g729_used'
> 2010-10-04 12:32:21.857199 [NOTICE] switch_loadable_module.c:274  
> Adding API Function 'g729_available'
> 2010-10-04 12:32:21.857229 [NOTICE] switch_loadable_module.c:274  
> Adding API Function 'g729_info'
>
>
>
> I'm using
> FreeSWITCH Version 1.0.head (git-758f383 2010-09-30 13-30-36 -0700)
>
> If I use mod_g729 instead of mod_com_g729 Freeswitch is running  
> stable.
>
> Does anybody have any ideas how to debug this issue?
>
> Thank you in advance
>     Jan
>
>
> -- 
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                         E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>
> AG Charlottenburg - HRB 71973
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From leon at scarlet-internet.nl  Mon Oct  4 04:57:53 2010
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Mon, 4 Oct 2010 13:57:53 +0200
Subject: [Freeswitch-users] High Availability and hostname used by Sofia
Message-ID: <EA736C6E-2651-43A6-BE6B-D202DC951F38@scarlet-internet.nl>

Hello List,

This is going to be a long mail, please bear with me :-)

I'd like to configure High Availability for a FreeSWITCH installation  
by using a failover server. I already discussed it a bit with Sekil on  
IRC, but I thought it'd be useful to discuss it here as well.


The idea is as follows:

- I have 2 servers both with an identical FreeSWITCH installation, FS1  
and FS2.
- Both servers have different static IP's but also one floating IP by  
means of heartbeat / haresources.
- Both FS instances have one Sofia SIP profile that use the floating IP.
- Both FS instances are always started (so the backup server will fail  
to load its SIP profile because the IP is not configured - which is  
expected behaviour)
- There'll be a /etc/ha.d/resource.d "resource script" that starts the  
SIP profile (through fs_cli -x) the moment it becomes primary
- The resource script will also send a "sofia recover" after the SIP  
profile has been brought up


This way:

- There'll be one logical switch.
- The moment FS2 becomes primary, users will be reachable through the  
entry in 'sip_registrations' table in the ODBC db, even though they  
may have registered on FS1
- FS2 will start to send SIP options to NAT'ed useragents (keepalive /  
udp hole punching) the moment the SIP profile is started.
- No client useragent involvement is necessary when failing over (like  
re-register)
- Running calls will be continued because of the "sofia recover"


Only problem here is that when a useragent registers to us, then FS  
also stores its own hostname in "sip_registrations" table in the  
columns "hostname" and "orig_hostname" (from  
mod_sofia_globals.hostname). This column is also used in the SELECT  
query when bridging to a user - so that means if the user registered  
to FS1, then a failover happens, then if a call is bridged towards  
that user on FS2, it won't find it because it will search with FS2's  
hostname in the SELECT query, right ?

If everything I'm writing so far is correct, then would it be alright  
to override the hostname in sofia.conf ? I think that would solve my  
problem. I already looked it up - it is first set in mod_sofia.c at  
line 4565 through gethostname(). It can be overriden in sofia.c  
(function config_sofia) around line 2845. It would probably also work  
to give both hosts an identical hostname in the OS, but I don't like  
that (because of rsyslog and more).


I'm not convinced yet on how to proceed, please let me know if I'm  
going the wrong direction. If I'm successful on getting this running,  
I'll document everything on the wiki - also the haresources script I  
already wrote I'll put in my contrib dir (or elsewhere in main tree if  
preferred).


Thanks for your time,

Leon


From ovvenkatesan at gmail.com  Mon Oct  4 05:07:58 2010
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Mon, 4 Oct 2010 17:37:58 +0530
Subject: [Freeswitch-users] openzap error "RECOVERY_ON_TIMER_EXPIRE"
Message-ID: <AANLkTimWj6z1tC1JAFjxhyCXMQhf0i33FRWUcapiGZxy@mail.gmail.com>

Hi to all,

I am using Sangoma A101 card and openzap to make outbound call.
When I am trying to originate call, getting blow error

2010-10-04 17:34:16.606653 [ERR] switch_ivr_originate.c:2648 Cannot create
outgoing channel of type [openzap] cause: [RECOVERY_ON_TIMER_EXPIRE]
2010-10-04 17:34:16.606653 [DEBUG] switch_ivr_originate.c:3456 Originate
Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE]


Any one come across this problem, what is this mean and how to rectify the
same ?

Thanks in advance.

Regards,
Venkat.

-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From riedinger at sns.eu  Mon Oct  4 05:40:08 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Mon, 04 Oct 2010 14:40:08 +0200
Subject: [Freeswitch-users] Substitution of mod_g729 by
 mod_com_g729	causes core dump
In-Reply-To: <1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>
References: <4CA9B34C.5070400@sns.eu>
	<1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>
Message-ID: <4CA9CB28.8010101@sns.eu>

  It seems that David is right. I updated to the current git tree 
(git-c701d41 2010-10-03 20-00-32 -0400). Now I find in the log:

2010-10-04 14:24:00.542096 [CRIT] switch_loadable_module.c:928 Error 
Loading module /usr/local/freeswitch/mod/mod_com_g729.so
**Trying to load an out of date module, please rebuild the module.**

How can I update mod_com_g729.so? There aren't any instructions in the 
install.txt for updating it. Shall I rerun the fsg729-153-installer? 
Will the installer pull the right version for my current Freeswitch 
installation?

BR
     Jan

Am 04.10.2010 13:33, schrieb David Ponzone:
> Jan,
>
> which version of mod_com_g729 and FS are you using ?
> There were very recent updates to the codec, that may require to be 
> used with the latest git version of FS.
>
> Brian West should able to confirm that later in the day.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IP eva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 04/10/2010 ? 12:58, Jan Riedinger a ?crit :
>
>> I just tried to subsitute mod_g729 by mod_com_g729 by using an 
>> appropriate moduls.conf.xml. After a restart of Freeswitch, it 
>> crahses, when the first inbound call gets processed:
>>
>> freeswitch at sns-fs1.sns.eu <mailto:freeswitch at sns-fs1.sns.eu>> 
>> 2010-10-04 12:32:27.064152 [NOTICE] switch_channel.c:779 New Channel 
>> sofia/external/xyz at a.b.c.d <mailto:sofia/external/xyz at a.b.c.d> 
>> [8e94789e-5a8d-4088-8ad1-cd5fb2be0c4f]
>> freeswitch: src/switch_core_memory.c:448: switch_core_perform_alloc: 
>> Assertion `pool != ((void *)0)' failed.
>> Aborted (core dumped)
>>
>> There seems to be no error, when mod_com_g729 gets loaded:
>> 2010-10-04 12:32:21.856829 [INFO] mod_com_g729.c:243 Permitted 
>> G.729AB channels: 20
>> 2010-10-04 12:32:21.856920 [NOTICE] switch_loadable_module.c:185 
>> Adding Codec G729 18 G.729 8000hz 10ms 8000bps
>> 2010-10-04 12:32:21.856925 [NOTICE] switch_loadable_module.c:185 
>> Adding Codec G729 18 G.729 8000hz 20ms 8000bps
>> 2010-10-04 12:32:21.856928 [NOTICE] switch_loadable_module.c:185 
>> Adding Codec G729 18 G.729 8000hz 30ms 8000bps
>> 2010-10-04 12:32:21.856933 [NOTICE] switch_loadable_module.c:185 
>> Adding Codec G729 18 G.729 8000hz 40ms 8000bps
>> 2010-10-04 12:32:21.856937 [NOTICE] switch_loadable_module.c:185 
>> Adding Codec G729 18 G.729 8000hz 50ms 8000bps
>> 2010-10-04 12:32:21.856943 [NOTICE] switch_loadable_module.c:185 
>> Adding Codec G729 18 G.729 8000hz 60ms 8000bps
>> 2010-10-04 12:32:21.856947 [NOTICE] switch_loadable_module.c:185 
>> Adding Codec G729 18 G.729 8000hz 70ms 8000bps
>> 2010-10-04 12:32:21.856974 [NOTICE] switch_loadable_module.c:185 
>> Adding Codec G729 18 G.729 8000hz 80ms 8000bps
>> 2010-10-04 12:32:21.856978 [NOTICE] switch_loadable_module.c:185 
>> Adding Codec G729 18 G.729 8000hz 90ms 8000bps
>> 2010-10-04 12:32:21.856983 [NOTICE] switch_loadable_module.c:185 
>> Adding Codec G729 18 G.729 8000hz 100ms 8000bps
>> 2010-10-04 12:32:21.856988 [NOTICE] switch_loadable_module.c:185 
>> Adding Codec G729 18 G.729 8000hz 110ms 8000bps
>> 2010-10-04 12:32:21.856993 [NOTICE] switch_loadable_module.c:185 
>> Adding Codec G729 18 G.729 8000hz 120ms 8000bps
>> 2010-10-04 12:32:21.857144 [NOTICE] switch_loadable_module.c:274 
>> Adding API Function 'g729_count'
>> 2010-10-04 12:32:21.857175 [NOTICE] switch_loadable_module.c:274 
>> Adding API Function 'g729_used'
>> 2010-10-04 12:32:21.857199 [NOTICE] switch_loadable_module.c:274 
>> Adding API Function 'g729_available'
>> 2010-10-04 12:32:21.857229 [NOTICE] switch_loadable_module.c:274 
>> Adding API Function 'g729_info'
>>
>>
>>
>> I'm using
>> FreeSWITCH Version 1.0.head (git-758f383 2010-09-30 13-30-36 -0700)
>>
>> If I use mod_g729 instead of mod_com_g729 Freeswitch is running stable.
>>
>> Does anybody have any ideas how to debug this issue?
>>
>> Thank you in advance
>>     Jan
>>
>>
>> -- 
>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>                                          E-Mail:riedinger at sns.eu  <mailto:riedinger at sns.eu>
>> SNS Consult GmbH                        ICQ   :  163-237-041
>> S?dwestkorso 49a                        MSN   :jan at sns-consult.de  <mailto:jan at sns-consult.de>
>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>
>> AG Charlottenburg - HRB 71973
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                         E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973

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From gavin.henry at gmail.com  Mon Oct  4 05:52:08 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 4 Oct 2010 13:52:08 +0100
Subject: [Freeswitch-users] Setting a domain name in the Contact header of
	an INVITE
Message-ID: <AANLkTikC_51mHrYE=L6T--A-GQWphLGJW1hpn=O0Kxmn@mail.gmail.com>

Hi,

Is this possible?

Contact: <sip:gw+test at test.domain.co.uk:5060;transport=udp;gw=text>

instead of:

Contact: <sip:gw+test at FS_PUBLIC_IP:5060;transport=udp;gw=test>



I've tried exporting sip_contact_host and domain_name etc.

sip_contact_params only adds string literals to this header.

Cheers.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.surevoip.co.uk


From david.ponzone at ipeva.fr  Mon Oct  4 06:09:28 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 4 Oct 2010 15:09:28 +0200
Subject: [Freeswitch-users] Substitution of mod_g729 by mod_com_g729
	causes core dump
In-Reply-To: <4CA9CB28.8010101@sns.eu>
References: <4CA9B34C.5070400@sns.eu>
	<1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>
	<4CA9CB28.8010101@sns.eu>
Message-ID: <D76D3E6E-7576-4299-BF16-9A7B29BFC37F@ipeva.fr>

Nope, you have to install them manually.
Current is v158.
You can download it from the same place than 153, but via direct URL  
only.
It's still Beta I think.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 04/10/2010 ? 14:40, Jan Riedinger a ?crit :

> It seems that David is right. I updated to the current git tree (git- 
> c701d41 2010-10-03 20-00-32 -0400). Now I find in the log:
>
> 2010-10-04 14:24:00.542096 [CRIT] switch_loadable_module.c:928 Error  
> Loading module /usr/local/freeswitch/mod/mod_com_g729.so
> **Trying to load an out of date module, please rebuild the module.**
>
> How can I update mod_com_g729.so? There aren't any instructions in  
> the install.txt for updating it. Shall I rerun the fsg729-153- 
> installer? Will the installer pull the right version for my current  
> Freeswitch installation?
>
> BR
>     Jan
>
> Am 04.10.2010 13:33, schrieb David Ponzone:
>>
>> Jan,
>>
>> which version of mod_com_g729 and FS are you using ?
>> There were very recent updates to the codec, that may require to be  
>> used with the latest git version of FS.
>>
>> Brian West should able to confirm that later in the day.
>>
>> David Ponzone   Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client  IP eva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ? l'intention exclusive de ses destinataires. Toute  
>> utilisation ou diffusion non autoris?e est interdite. Tout message  
>> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
>> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
>> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>>
>> Le 04/10/2010 ? 12:58, Jan Riedinger a ?crit :
>>
>>> I just tried to subsitute mod_g729 by mod_com_g729 by using an  
>>> appropriate moduls.conf.xml. After a restart of Freeswitch, it  
>>> crahses, when the first inbound call gets processed:
>>>
>>> freeswitch at sns-fs1.sns.eu> 2010-10-04 12:32:27.064152 [NOTICE]  
>>> switch_channel.c:779 New Channel sofia/external/xyz at a.b.c.d  
>>> [8e94789e-5a8d-4088-8ad1-cd5fb2be0c4f]
>>> freeswitch: src/switch_core_memory.c:448:  
>>> switch_core_perform_alloc: Assertion `pool != ((void *)0)' failed.
>>> Aborted (core dumped)
>>>
>>> There seems to be no error, when mod_com_g729 gets loaded:
>>> 2010-10-04 12:32:21.856829 [INFO] mod_com_g729.c:243 Permitted G. 
>>> 729AB channels: 20
>>> 2010-10-04 12:32:21.856920 [NOTICE] switch_loadable_module.c:185  
>>> Adding Codec G729 18 G.729 8000hz 10ms 8000bps
>>> 2010-10-04 12:32:21.856925 [NOTICE] switch_loadable_module.c:185  
>>> Adding Codec G729 18 G.729 8000hz 20ms 8000bps
>>> 2010-10-04 12:32:21.856928 [NOTICE] switch_loadable_module.c:185  
>>> Adding Codec G729 18 G.729 8000hz 30ms 8000bps
>>> 2010-10-04 12:32:21.856933 [NOTICE] switch_loadable_module.c:185  
>>> Adding Codec G729 18 G.729 8000hz 40ms 8000bps
>>> 2010-10-04 12:32:21.856937 [NOTICE] switch_loadable_module.c:185  
>>> Adding Codec G729 18 G.729 8000hz 50ms 8000bps
>>> 2010-10-04 12:32:21.856943 [NOTICE] switch_loadable_module.c:185  
>>> Adding Codec G729 18 G.729 8000hz 60ms 8000bps
>>> 2010-10-04 12:32:21.856947 [NOTICE] switch_loadable_module.c:185  
>>> Adding Codec G729 18 G.729 8000hz 70ms 8000bps
>>> 2010-10-04 12:32:21.856974 [NOTICE] switch_loadable_module.c:185  
>>> Adding Codec G729 18 G.729 8000hz 80ms 8000bps
>>> 2010-10-04 12:32:21.856978 [NOTICE] switch_loadable_module.c:185  
>>> Adding Codec G729 18 G.729 8000hz 90ms 8000bps
>>> 2010-10-04 12:32:21.856983 [NOTICE] switch_loadable_module.c:185  
>>> Adding Codec G729 18 G.729 8000hz 100ms 8000bps
>>> 2010-10-04 12:32:21.856988 [NOTICE] switch_loadable_module.c:185  
>>> Adding Codec G729 18 G.729 8000hz 110ms 8000bps
>>> 2010-10-04 12:32:21.856993 [NOTICE] switch_loadable_module.c:185  
>>> Adding Codec G729 18 G.729 8000hz 120ms 8000bps
>>> 2010-10-04 12:32:21.857144 [NOTICE] switch_loadable_module.c:274  
>>> Adding API Function 'g729_count'
>>> 2010-10-04 12:32:21.857175 [NOTICE] switch_loadable_module.c:274  
>>> Adding API Function 'g729_used'
>>> 2010-10-04 12:32:21.857199 [NOTICE] switch_loadable_module.c:274  
>>> Adding API Function 'g729_available'
>>> 2010-10-04 12:32:21.857229 [NOTICE] switch_loadable_module.c:274  
>>> Adding API Function 'g729_info'
>>>
>>>
>>>
>>> I'm using
>>> FreeSWITCH Version 1.0.head (git-758f383 2010-09-30 13-30-36 -0700)
>>>
>>> If I use mod_g729 instead of mod_com_g729 Freeswitch is running  
>>> stable.
>>>
>>> Does anybody have any ideas how to debug this issue?
>>>
>>> Thank you in advance
>>>     Jan
>>>
>>>
>>> -- 
>>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>>                                         E-Mail:  riedinger at sns.eu
>>> SNS Consult GmbH                        ICQ   :  163-237-041
>>> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
>>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>>
>>> AG Charlottenburg - HRB 71973
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> -- 
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                         E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>
> AG Charlottenburg - HRB 71973
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From gavin.henry at gmail.com  Mon Oct  4 06:10:39 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 4 Oct 2010 14:10:39 +0100
Subject: [Freeswitch-users] Setting a domain name in the Contact header
	of an INVITE
In-Reply-To: <AANLkTikC_51mHrYE=L6T--A-GQWphLGJW1hpn=O0Kxmn@mail.gmail.com>
References: <AANLkTikC_51mHrYE=L6T--A-GQWphLGJW1hpn=O0Kxmn@mail.gmail.com>
Message-ID: <AANLkTiktk53m4BxMTx1v6GOvnOTw=XpDnFa8SHgaT3aU@mail.gmail.com>

On 4 October 2010 13:52, Gavin Henry <gavin.henry at gmail.com> wrote:
> Hi,
>
> Is this possible?
>
> Contact: <sip:gw+test at test.domain.co.uk:5060;transport=udp;gw=text>
>
> instead of:
>
> Contact: <sip:gw+test at FS_PUBLIC_IP:5060;transport=udp;gw=test>

I didn't try hard enough!!!

                <action application="export"
data="sip_contact_host=test.domain.co.uk"/>

I had set instead of export and you can also do it in your gw profile like so:



        <gateway name="bt">
                <param name="realm" value="test.domain.co.uk"/>
                <param name="from-domain" value="test.domain.co.uk" />
                <param name="contact-host" value="test.domain.co.uk" />
                <param name="proxy" value="blah.co.uk"/>
                <param name="outbound-proxy" value="blah_IP"/>
                <param name="register" value="false"/>
                <param name="username" value="+441234567890"/>
                <param name="password" value="blank"/>
                <param name="caller-id-in-from" value="true"/>
        </gateway>


-- 
http://www.suretecsystems.com/services/openldap/
http://www.surevoip.co.uk


From brian at freeswitch.org  Mon Oct  4 06:15:07 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 4 Oct 2010 08:15:07 -0500
Subject: [Freeswitch-users] Substitution of mod_g729 by mod_com_g729
	causes core dump
In-Reply-To: <D76D3E6E-7576-4299-BF16-9A7B29BFC37F@ipeva.fr>
References: <4CA9B34C.5070400@sns.eu>
	<1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>
	<4CA9CB28.8010101@sns.eu>
	<D76D3E6E-7576-4299-BF16-9A7B29BFC37F@ipeva.fr>
Message-ID: <7CAB482F-5FFA-491F-8BF6-EAE4E376C6F1@freeswitch.org>

http://files.freeswitch.org/g729/fsg729-158-installer

Just the CDN doesn't have the index updated yet.  That will fix your issue.

/b

On Oct 4, 2010, at 8:09 AM, David Ponzone wrote:

> Nope, you have to install them manually.
> Current is v158.
> You can download it from the same place than 153, but via direct URL only.
> It's still Beta I think.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 



From yehavi.bourvine at gmail.com  Mon Oct  4 06:25:26 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 4 Oct 2010 15:25:26 +0200
Subject: [Freeswitch-users] Some weird problems with Polycom phones.
Message-ID: <AANLkTimQy1k+OhAt2QkwwEMATjj4ke+zrzi-teWwYRh9@mail.gmail.com>

Hello,

  I have a few oddities with Polycom phones which I just have documented at
the Wiki. I append them here also for those who might tackle them.


The following issues have opened tickets with Polycom's support:

   - *Dial tone on a shared line*: Suppose you have an SLA extension defined
   on more than one phone. You answer on one phone, put it on hold, go to the
   other phone and press the blinking line in order to have the call there. If
   you press this line and within 1 second pull the handset - you hear a dial
   tone at the background of the session. If you wait more than 1 second or
   first pull the handset (and then press the line button) - all is ok.
   - *Dial using the second line*: If you have a phone with more than one
   extension defined on it, it will will use the first line for outgoing calls
   by default (unless of course you select a specific line). However, there is
   one special case: you dial on-hook and then press the speaker button:
      - The call is initiated on the first line.
      - Immediately after getting "Trying" the phone sends CANCEL.
      - The phone originates the call successfully over the second line.


       Regards, __Yehavi:
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From emss.mail at gmail.com  Mon Oct  4 04:13:01 2010
From: emss.mail at gmail.com (Eric Masson)
Date: Mon, 04 Oct 2010 13:13:01 +0200
Subject: [Freeswitch-users] sip to spa3102 fallback
In-Reply-To: <i8ajiu$2ce$1@dough.gmane.org>
References: <i8ajiu$2ce$1@dough.gmane.org>
Message-ID: <i8ccs0$9si$1@dough.gmane.org>

Le 03/10/2010 20:55, Eric Masson a ?crit :

Hello,

[Responding to myself]

> I have two sip providers configuration files in
> $(fs_conf}/conf/diaplan/default :
> - 00_keyyo.net.xml
> - 00_spa3102.xml
> I'd like to implement the following scenario for outbound calls :
> - dial using "keyyo" if available
> - fallback to "To PSTN" if keyyo isn't available (not registered,
> already used, for example)

I've found a solution that seems to work but a little problem stays.
I deactivated 00_keyyo.net.xml & 00_spa3102.xml then created a new file 
in the same directory :
00_outbound.xml
<include>
   <extension name="outbound_calls" continue="false">
     <condition field="destination_number" expression = "^0(.*)$">
       <action application="set" data="effective_caller_id_name=**MyID**"/>
       <action application="set" 
data="effective_caller_id_number=**MyPhone**"/>
       <action application="bridge" 
data="sofia/gateway/keyyo.net/0$1|sofia/internal/4$1 at spafenmsf:5061"/>
     </condition>
   </extension>
</include>

If keyyo.net is registered, FS uses keyyo.net gateway, if not FS uses 
the spa.

The problem remaining is that FS takes a long time to detect that 
connection to keyyo.net is lost (I've tested by disconnecting the box 
from the internet).

keyyo.net gateway is defined the following way :
keyyo.net.xml
<include>
   <gateway name="keyyo.net">
     <param name="username" value=**MyUsername**/>
     <param name="password" value=**MyPassword**/>
     <param name="register-proxy" value="keyyo.net"/>
     <param name="register" value="true"/>
     <param name="ping" value="60"/>
   </gateway>
</include>

Any idea, please ?

Kind Regards

Eric Masson



From Russell.Mosemann at cune.org  Mon Oct  4 05:18:35 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Mon, 4 Oct 2010 07:18:35 -0500
Subject: [Freeswitch-users] Accessing wanpipe from openVZ container
Message-ID: <06B07C8A89E8409FBA52108AB33BF5E6@cune.pri>

We are using openVZ with FreeSWITCH running in a container. In another situation, I successfully installed DAHDI in the host node, exported the devices to the container and installed libpri in the container. FS could access the card without any problems.

We are moving to a Sangoma A101D and an A200. I have installed the wanpipe drivers in the host node (make freetdm and make install), exported the devices to the container and installed the signaling in the container (make install_pri). However, that is not quite enough for the container, because mod_freetdm looks for libsangoma, which is in the host node. wancfg_fs does too much, because it attempts to configure wanrouter and FS at the same time.

I've looked through the FS wiki and the FS section of the Sangoma wiki, but all of the directions are for a combined installation. Is there a set of steps to install the wanpipe drivers separately from the signaling and do the individual configuration for each? Thanks.

Russell



From tgraziano at myitdepartment.net  Mon Oct  4 06:41:44 2010
From: tgraziano at myitdepartment.net (Tony Graziano)
Date: Mon, 4 Oct 2010 09:41:44 -0400
Subject: [Freeswitch-users] Some weird problems with Polycom phones.
In-Reply-To: <AANLkTimQy1k+OhAt2QkwwEMATjj4ke+zrzi-teWwYRh9@mail.gmail.com>
References: <AANLkTimQy1k+OhAt2QkwwEMATjj4ke+zrzi-teWwYRh9@mail.gmail.com>
Message-ID: <AANLkTi=2Vh3WZ+Mnr14H-6U8prQTnJMT6Lt3mAf29z0Y@mail.gmail.com>

Was this on firmware 3.3?

On 10/4/10, Yehavi Bourvine <yehavi.bourvine at gmail.com> wrote:
> Hello,
>
>   I have a few oddities with Polycom phones which I just have documented at
> the Wiki. I append them here also for those who might tackle them.
>
>
> The following issues have opened tickets with Polycom's support:
>
>    - *Dial tone on a shared line*: Suppose you have an SLA extension defined
>    on more than one phone. You answer on one phone, put it on hold, go to
> the
>    other phone and press the blinking line in order to have the call there.
> If
>    you press this line and within 1 second pull the handset - you hear a
> dial
>    tone at the background of the session. If you wait more than 1 second or
>    first pull the handset (and then press the line button) - all is ok.
>    - *Dial using the second line*: If you have a phone with more than one
>    extension defined on it, it will will use the first line for outgoing
> calls
>    by default (unless of course you select a specific line). However, there
> is
>    one special case: you dial on-hook and then press the speaker button:
>       - The call is initiated on the first line.
>       - Immediately after getting "Trying" the phone sends CANCEL.
>       - The phone originates the call successfully over the second line.
>
>
>        Regards, __Yehavi:
>

-- 
Sent from my mobile device

======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgraziano at voice.myitdepartment.net
Fax: 434.984.8431

Email: tgraziano at myitdepartment.net

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpdesk at voice.myitdepartment.net
Fax: 434.984.8427

Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/

Why do mathematicians always confuse Halloween and Christmas?
Because 31 Oct = 25 Dec.


From yehavi.bourvine at gmail.com  Mon Oct  4 06:57:58 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 4 Oct 2010 15:57:58 +0200
Subject: [Freeswitch-users] Some weird problems with Polycom phones.
In-Reply-To: <AANLkTi=2Vh3WZ+Mnr14H-6U8prQTnJMT6Lt3mAf29z0Y@mail.gmail.com>
References: <AANLkTimQy1k+OhAt2QkwwEMATjj4ke+zrzi-teWwYRh9@mail.gmail.com>
	<AANLkTi=2Vh3WZ+Mnr14H-6U8prQTnJMT6Lt3mAf29z0Y@mail.gmail.com>
Message-ID: <AANLkTimEybX_Kse+tc7Hd__Q78+py-ei1fJXQfMa439E@mail.gmail.com>

It is on 3.2.3; I cannot download version 3.3.0 (it claims that my account
cannot do it) and I did not get 3.3 from our Polycom distributor  yet.

                                     Regards, __Yehavi:

2010/10/4 Tony Graziano <tgraziano at myitdepartment.net>

> Was this on firmware 3.3?
>
> On 10/4/10, Yehavi Bourvine <yehavi.bourvine at gmail.com> wrote:
> > Hello,
> >
> >   I have a few oddities with Polycom phones which I just have documented
> at
> > the Wiki. I append them here also for those who might tackle them.
> >
> >
> > The following issues have opened tickets with Polycom's support:
> >
> >    - *Dial tone on a shared line*: Suppose you have an SLA extension
> defined
> >    on more than one phone. You answer on one phone, put it on hold, go to
> > the
> >    other phone and press the blinking line in order to have the call
> there.
> > If
> >    you press this line and within 1 second pull the handset - you hear a
> > dial
> >    tone at the background of the session. If you wait more than 1 second
> or
> >    first pull the handset (and then press the line button) - all is ok.
> >    - *Dial using the second line*: If you have a phone with more than one
> >    extension defined on it, it will will use the first line for outgoing
> > calls
> >    by default (unless of course you select a specific line). However,
> there
> > is
> >    one special case: you dial on-hook and then press the speaker button:
> >       - The call is initiated on the first line.
> >       - Immediately after getting "Trying" the phone sends CANCEL.
> >       - The phone originates the call successfully over the second line.
> >
> >
> >        Regards, __Yehavi:
> >
>
> --
> Sent from my mobile device
>
> ======================
> Tony Graziano, Manager
> Telephone: 434.984.8430
> sip: tgraziano at voice.myitdepartment.net
> Fax: 434.984.8431
>
> Email: tgraziano at myitdepartment.net
>
> LAN/Telephony/Security and Control Systems Helpdesk:
> Telephone: 434.984.8426
> sip: helpdesk at voice.myitdepartment.net
> Fax: 434.984.8427
>
> Helpdesk Contract Customers:
> http://www.myitdepartment.net/gethelp/
>
> Why do mathematicians always confuse Halloween and Christmas?
> Because 31 Oct = 25 Dec.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Oct  4 07:08:16 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 4 Oct 2010 09:08:16 -0500
Subject: [Freeswitch-users] Accessing wanpipe from openVZ container
In-Reply-To: <06B07C8A89E8409FBA52108AB33BF5E6@cune.pri>
References: <06B07C8A89E8409FBA52108AB33BF5E6@cune.pri>
Message-ID: <6EA6B28B-A706-47CA-AAEA-A1E7D49B3AB2@freeswitch.org>

Its basically the same thing.. let the VE have access to the hardware.

/b

On Oct 4, 2010, at 7:18 AM, Russell Mosemann wrote:

> We are using openVZ with FreeSWITCH running in a container. In another situation, I successfully installed DAHDI in the host node, exported the devices to the container and installed libpri in the container. FS could access the card without any problems.
> 
> We are moving to a Sangoma A101D and an A200. I have installed the wanpipe drivers in the host node (make freetdm and make install), exported the devices to the container and installed the signaling in the container (make install_pri). However, that is not quite enough for the container, because mod_freetdm looks for libsangoma, which is in the host node. wancfg_fs does too much, because it attempts to configure wanrouter and FS at the same time.
> 
> I've looked through the FS wiki and the FS section of the Sangoma wiki, but all of the directions are for a combined installation. Is there a set of steps to install the wanpipe drivers separately from the signaling and do the individual configuration for each? Thanks.
> 
> Russell



From neillw at aeonvista.com  Mon Oct  4 07:19:30 2010
From: neillw at aeonvista.com (Neill Wilkinson)
Date: Mon, 4 Oct 2010 15:19:30 +0100
Subject: [Freeswitch-users] Setting a domain name in the Contact header
 of an INVITE
Message-ID: <AANLkTinpjjBDGp4MtGs80QDMFyCSRc-3r7b0kAob_XfY@mail.gmail.com>

Gavin,

this kinda thing works for me:

<action application="export" data="sip_contact_user=${sip_from_user}" />

Contact user seems to overwrite the whole field, so something like:

<action application="export"
data="sip_contact_user=gw+test at test.domain.co.uk:5060;transport=udp;gw=text"
/>

Neill.....;o)
Aeonvista Ltd <http://aeonvista.blogspot.com>
*Opening Up New Ideas*

---------- Forwarded message ----------
> From: Gavin Henry <gavin.henry at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Mon, 4 Oct 2010 13:52:08 +0100
> Subject: [Freeswitch-users] Setting a domain name in the Contact header of
> an INVITE
> Hi,
>
> Is this possible?
>
> Contact: <sip:gw+test at test.domain.co.uk:5060;transport=udp;gw=text>
>
> instead of:
>
> Contact: <sip:gw+test at FS_PUBLIC_IP:5060;transport=udp;gw=test>
>
>
>
> I've tried exporting sip_contact_host and domain_name etc.
>
> sip_contact_params only adds string literals to this header.
>
> Cheers.
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.surevoip.co.uk
>
>
>
>
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From anthony.minessale at gmail.com  Mon Oct  4 07:27:47 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 Oct 2010 09:27:47 -0500
Subject: [Freeswitch-users] High Availability and hostname used by Sofia
In-Reply-To: <EA736C6E-2651-43A6-BE6B-D202DC951F38@scarlet-internet.nl>
References: <EA736C6E-2651-43A6-BE6B-D202DC951F38@scarlet-internet.nl>
Message-ID: <AANLkTikJacUOvCCOwc_LNvNRwdVzfoYE4jNchxPBRPS-@mail.gmail.com>

the registrations table intentionally does not consider the hostname
so you can cluster many boxes on the same domain.

On Mon, Oct 4, 2010 at 6:57 AM, Leon de Rooij <leon at scarlet-internet.nl> wrote:
> Hello List,
>
> This is going to be a long mail, please bear with me :-)
>
> I'd like to configure High Availability for a FreeSWITCH installation
> by using a failover server. I already discussed it a bit with Sekil on
> IRC, but I thought it'd be useful to discuss it here as well.
>
>
> The idea is as follows:
>
> - I have 2 servers both with an identical FreeSWITCH installation, FS1
> and FS2.
> - Both servers have different static IP's but also one floating IP by
> means of heartbeat / haresources.
> - Both FS instances have one Sofia SIP profile that use the floating IP.
> - Both FS instances are always started (so the backup server will fail
> to load its SIP profile because the IP is not configured - which is
> expected behaviour)
> - There'll be a /etc/ha.d/resource.d "resource script" that starts the
> SIP profile (through fs_cli -x) the moment it becomes primary
> - The resource script will also send a "sofia recover" after the SIP
> profile has been brought up
>
>
> This way:
>
> - There'll be one logical switch.
> - The moment FS2 becomes primary, users will be reachable through the
> entry in 'sip_registrations' table in the ODBC db, even though they
> may have registered on FS1
> - FS2 will start to send SIP options to NAT'ed useragents (keepalive /
> udp hole punching) the moment the SIP profile is started.
> - No client useragent involvement is necessary when failing over (like
> re-register)
> - Running calls will be continued because of the "sofia recover"
>
>
> Only problem here is that when a useragent registers to us, then FS
> also stores its own hostname in "sip_registrations" table in the
> columns "hostname" and "orig_hostname" (from
> mod_sofia_globals.hostname). This column is also used in the SELECT
> query when bridging to a user - so that means if the user registered
> to FS1, then a failover happens, then if a call is bridged towards
> that user on FS2, it won't find it because it will search with FS2's
> hostname in the SELECT query, right ?
>
> If everything I'm writing so far is correct, then would it be alright
> to override the hostname in sofia.conf ? I think that would solve my
> problem. I already looked it up - it is first set in mod_sofia.c at
> line 4565 through gethostname(). It can be overriden in sofia.c
> (function config_sofia) around line 2845. It would probably also work
> to give both hosts an identical hostname in the OS, but I don't like
> that (because of rsyslog and more).
>
>
> I'm not convinced yet on how to proceed, please let me know if I'm
> going the wrong direction. If I'm successful on getting this running,
> I'll document everything on the wiki - also the haresources script I
> already wrote I'll put in my contrib dir (or elsewhere in main tree if
> preferred).
>
>
> Thanks for your time,
>
> Leon
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From leon at scarlet-internet.nl  Mon Oct  4 07:38:38 2010
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Mon, 4 Oct 2010 16:38:38 +0200
Subject: [Freeswitch-users] High Availability and hostname used by Sofia
In-Reply-To: <AANLkTikJacUOvCCOwc_LNvNRwdVzfoYE4jNchxPBRPS-@mail.gmail.com>
References: <EA736C6E-2651-43A6-BE6B-D202DC951F38@scarlet-internet.nl>
	<AANLkTikJacUOvCCOwc_LNvNRwdVzfoYE4jNchxPBRPS-@mail.gmail.com>
Message-ID: <F3C7286A-4C7F-4406-AE33-8B9207D622D8@scarlet-internet.nl>

Hi Anthony,

Thanks, I will try it out right away.

Kind regards,

Leon



On Oct 4, 2010, at 4:27 PM, Anthony Minessale wrote:

> the registrations table intentionally does not consider the hostname
> so you can cluster many boxes on the same domain.
>
> On Mon, Oct 4, 2010 at 6:57 AM, Leon de Rooij <leon at scarlet-internet.nl 
> > wrote:
>> Hello List,
>>
>> This is going to be a long mail, please bear with me :-)
>>
>> I'd like to configure High Availability for a FreeSWITCH installation
>> by using a failover server. I already discussed it a bit with Sekil  
>> on
>> IRC, but I thought it'd be useful to discuss it here as well.
>>
>>
>> The idea is as follows:
>>
>> - I have 2 servers both with an identical FreeSWITCH installation,  
>> FS1
>> and FS2.
>> - Both servers have different static IP's but also one floating IP by
>> means of heartbeat / haresources.
>> - Both FS instances have one Sofia SIP profile that use the  
>> floating IP.
>> - Both FS instances are always started (so the backup server will  
>> fail
>> to load its SIP profile because the IP is not configured - which is
>> expected behaviour)
>> - There'll be a /etc/ha.d/resource.d "resource script" that starts  
>> the
>> SIP profile (through fs_cli -x) the moment it becomes primary
>> - The resource script will also send a "sofia recover" after the SIP
>> profile has been brought up
>>
>>
>> This way:
>>
>> - There'll be one logical switch.
>> - The moment FS2 becomes primary, users will be reachable through the
>> entry in 'sip_registrations' table in the ODBC db, even though they
>> may have registered on FS1
>> - FS2 will start to send SIP options to NAT'ed useragents  
>> (keepalive /
>> udp hole punching) the moment the SIP profile is started.
>> - No client useragent involvement is necessary when failing over  
>> (like
>> re-register)
>> - Running calls will be continued because of the "sofia recover"
>>
>>
>> Only problem here is that when a useragent registers to us, then FS
>> also stores its own hostname in "sip_registrations" table in the
>> columns "hostname" and "orig_hostname" (from
>> mod_sofia_globals.hostname). This column is also used in the SELECT
>> query when bridging to a user - so that means if the user registered
>> to FS1, then a failover happens, then if a call is bridged towards
>> that user on FS2, it won't find it because it will search with FS2's
>> hostname in the SELECT query, right ?
>>
>> If everything I'm writing so far is correct, then would it be alright
>> to override the hostname in sofia.conf ? I think that would solve my
>> problem. I already looked it up - it is first set in mod_sofia.c at
>> line 4565 through gethostname(). It can be overriden in sofia.c
>> (function config_sofia) around line 2845. It would probably also work
>> to give both hosts an identical hostname in the OS, but I don't like
>> that (because of rsyslog and more).
>>
>>
>> I'm not convinced yet on how to proceed, please let me know if I'm
>> going the wrong direction. If I'm successful on getting this running,
>> I'll document everything on the wiki - also the haresources script I
>> already wrote I'll put in my contrib dir (or elsewhere in main tree  
>> if
>> preferred).
>>
>>
>> Thanks for your time,
>>
>> Leon
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From Russell.Mosemann at cune.org  Mon Oct  4 07:44:44 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Mon, 4 Oct 2010 14:44:44 -0000
Subject: [Freeswitch-users] Accessing wanpipe from openVZ container
In-Reply-To: <6EA6B28B-A706-47CA-AAEA-A1E7D49B3AB2@freeswitch.org>
Message-ID: <20101004144444.C17CA3CDA7A@cuneorg-email.cune.pri>

> Its basically the same thing.. let the VE have access to the hardware.

If you carefully read my previous message, that part is already done. The
issue is installing the required signaling files and libraries in the VE
without also compiling the drivers in the VE.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From moises.silva at gmail.com  Mon Oct  4 07:45:58 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 4 Oct 2010 10:45:58 -0400
Subject: [Freeswitch-users] Accessing wanpipe from openVZ container
In-Reply-To: <06B07C8A89E8409FBA52108AB33BF5E6@cune.pri>
References: <06B07C8A89E8409FBA52108AB33BF5E6@cune.pri>
Message-ID: <AANLkTimHaPoj=7JoAVLGT-uq8sJyD8vAhZNXYAsmU0rj@mail.gmail.com>

Hi Russell,

On Mon, Oct 4, 2010 at 8:18 AM, Russell Mosemann
<Russell.Mosemann at cune.org> wrote:
> We are moving to a Sangoma A101D and an A200. I have installed the wanpipe drivers in the host node (make freetdm and make install), exported the devices to the container and installed the signaling in the container (make install_pri). However, that is not quite enough for the container, because mod_freetdm looks for libsangoma, which is in the host node. wancfg_fs does too much, because it attempts to configure wanrouter and FS at the same time.
>

I have no experience with openVZ, but I know some of our support staff
got it working with xen just fine. I think openVZ should not be a
problem.

You can configure in the host and start devices on the host, then
configure FreeSWITCH/FreeTDM manually on the container (given that
wancfg_fs does too much).

> I've looked through the FS wiki and the FS section of the Sangoma wiki, but all of the directions are for a combined installation. Is there a set of steps to install the wanpipe drivers separately from the signaling and do the individual configuration for each? Thanks.
>

No tools to do it automatically if you want to do it separate.
wancfg_fs is meant for newbie users and as a convenience tool. If you
are playing with openVZ, you have to configure manually, or configure
everything on the host and then just move the signaling configuration
to the container.

Drop by #sangoma at irc.freenode.org if you need a hand from our support staff.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From moises.silva at gmail.com  Mon Oct  4 07:52:51 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 4 Oct 2010 10:52:51 -0400
Subject: [Freeswitch-users] openzap error "RECOVERY_ON_TIMER_EXPIRE"
In-Reply-To: <AANLkTimWj6z1tC1JAFjxhyCXMQhf0i33FRWUcapiGZxy@mail.gmail.com>
References: <AANLkTimWj6z1tC1JAFjxhyCXMQhf0i33FRWUcapiGZxy@mail.gmail.com>
Message-ID: <AANLkTikn3Wr7ei0eM1u0JJs+akT4eU3D=pZTTgNruTPY@mail.gmail.com>

On Mon, Oct 4, 2010 at 8:07 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:
> Hi to all,
>
> I am using Sangoma A101 card and openzap to make outbound call.
> When I am trying to originate call, getting blow error
>
> 2010-10-04 17:34:16.606653 [ERR] switch_ivr_originate.c:2648 Cannot create
> outgoing channel of type [openzap] cause: [RECOVERY_ON_TIMER_EXPIRE]
> 2010-10-04 17:34:16.606653 [DEBUG] switch_ivr_originate.c:3456 Originate
> Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE]

Which signaling? what is your dialing syntax and your openzap
configuration (both openzap.conf and openzap.conf.xml).

Are you aware that openzap is deprecated and replaced by FreeTDM?
http://wiki.freeswitch.org/wiki/FreeTDM

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From fs-list at communicatefreely.net  Mon Oct  4 08:03:23 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Mon, 04 Oct 2010 11:03:23 -0400
Subject: [Freeswitch-users] Call back from voice mail creates loop
Message-ID: <4CA9ECBB.2000609@communicatefreely.net>

Hello,

I'm trying to set up the option in voicemail so that users can return a call directly from VM.

In my voicemail.conf.xml profile I have specified the dialplan as XML and the context as "internal",
which is what I use for calls coming from our endpoints or authenticated users.

In our internal dialplan, we do some pattern matching and call an appropriate lua dialplan script to
do build a set of instructions.

Problem is, the lua scripts are looking at the destination_number variable to decide where to send
the call.  This variable is still set to "voicemail" from when the user called into their voicemail.
 Shouldn't the voicemail app have set that to the number we want to call back?

I used the info app to look at the variables, and I didn't see anything obvious.

Is destination_number the correct variable to use in a lua script that generates a dialplan?

-Tim


From jeff at jefflenk.com  Mon Oct  4 08:12:31 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 4 Oct 2010 08:12:31 -0700 (PDT)
Subject: [Freeswitch-users] mod_managed on linux centos
In-Reply-To: <AANLkTik1CY5-6DUvsLg512bGRr66xLF5H8Gi7WYRenHx@mail.gmail.com>
References: <AANLkTikg4bxav+e04fAmJ_dKTHKERAPk715h9oHeYMRh@mail.gmail.com>
	<1283347083542-5486890.post@n2.nabble.com>
	<AANLkTi=xt9CZBPZLnBX56e2Rjdc_FRf_=er6Yy7w7ZuN@mail.gmail.com>
	<AANLkTi=S=tpzczDM8WSm6N8AeTkN3XNTPda6048sRoJ9@mail.gmail.com>
	<4C801B6E.4050103@puzzled.xs4all.nl>
	<AANLkTimzjbLHbObgsQKqX3t-hfaZWr95UGMi17YxZFHS@mail.gmail.com>
	<4C838C3B.3070202@puzzled.xs4all.nl>
	<AANLkTinP+UfvkM0dmBycy5GqivAXNHGF9RnTBNvV0gV4@mail.gmail.com>
	<AANLkTik1CY5-6DUvsLg512bGRr66xLF5H8Gi7WYRenHx@mail.gmail.com>
Message-ID: <1286205151753-5599563.post@n2.nabble.com>


Have you seen:
http://wiki.freeswitch.org/wiki/Mod_mono

<dllmap dll="mod_managed.dll" target="mod_managed.so" os="!windows"/>

regarding null exceptions?
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-managed-on-linux-centos-tp5485480p5599563.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From riedinger at sns.eu  Mon Oct  4 08:20:38 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Mon, 04 Oct 2010 17:20:38 +0200
Subject: [Freeswitch-users] mod_com_g729 licences used even without
	transcoding
In-Reply-To: <7CAB482F-5FFA-491F-8BF6-EAE4E376C6F1@freeswitch.org>
References: <4CA9B34C.5070400@sns.eu>	<1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>	<4CA9CB28.8010101@sns.eu>	<D76D3E6E-7576-4299-BF16-9A7B29BFC37F@ipeva.fr>
	<7CAB482F-5FFA-491F-8BF6-EAE4E376C6F1@freeswitch.org>
Message-ID: <4CA9F0C6.8060005@sns.eu>

  Using http://files.freeswitch.org/g729/fsg729-158-installer really did 
fix the issue described below.

However, it seems that the licence is used (up), even if there is no transcoding needed. In my CDR I find for all calls, for which G729 is involed at all,

read_codec="G729", read_rate="8000", write_codec="G729", write_rate="8000"

Nevertheless, all of these calls use a G729 licence. Is there any way to 
find out, why mod_com_g729 wants to do any kind of transcoding?

BR
     Jan

now the cod

Am 04.10.2010 15:15, schrieb Brian West:
> http://files.freeswitch.org/g729/fsg729-158-installer
>
> Just the CDN doesn't have the index updated yet.  That will fix your issue.
>
> /b
>
> On Oct 4, 2010, at 8:09 AM, David Ponzone wrote:
>
>> Nope, you have to install them manually.
>> Current is v158.
>> You can download it from the same place than 153, but via direct URL only.
>> It's still Beta I think.
>>
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                         E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973



From riedinger at sns.eu  Mon Oct  4 08:31:30 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Mon, 04 Oct 2010 17:31:30 +0200
Subject: [Freeswitch-users] Substitution of mod_g729 by
 mod_com_g729	causes core dump
In-Reply-To: <7CAB482F-5FFA-491F-8BF6-EAE4E376C6F1@freeswitch.org>
References: <4CA9B34C.5070400@sns.eu>	<1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>	<4CA9CB28.8010101@sns.eu>	<D76D3E6E-7576-4299-BF16-9A7B29BFC37F@ipeva.fr>
	<7CAB482F-5FFA-491F-8BF6-EAE4E376C6F1@freeswitch.org>
Message-ID: <4CA9F352.4070908@sns.eu>

  Hi Brian,

is there way, how I can "verify" that the an installed mod_com_g729 
works with a new Freeswitch version? I frighten to get trouble when I 
will update a system with a lot of live traffic. In  this case it would 
be bad, if you  have to wait some time until the link for the right 
installer gets available.

BR
     Jan

Am 04.10.2010 15:15, schrieb Brian West:
> http://files.freeswitch.org/g729/fsg729-158-installer
>
> Just the CDN doesn't have the index updated yet.  That will fix your issue.
>
> /b
>
> On Oct 4, 2010, at 8:09 AM, David Ponzone wrote:
>
>> Nope, you have to install them manually.
>> Current is v158.
>> You can download it from the same place than 153, but via direct URL only.
>> It's still Beta I think.
>>
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                         E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973



From mkane02 at harris.com  Mon Oct  4 08:34:41 2010
From: mkane02 at harris.com (Kane, Michael (mkane02))
Date: Mon, 4 Oct 2010 11:34:41 -0400
Subject: [Freeswitch-users] FreeSWITCH HA
Message-ID: <AF824FE3F385E543ACA98ADFD670BA6A0184CE72@mlbe2k9.cs.myharris.net>

Will FreeSWITCH HA work if bypass_media = true?

Thanks Mike
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From brian at freeswitch.org  Mon Oct  4 08:37:27 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 4 Oct 2010 10:37:27 -0500
Subject: [Freeswitch-users] Substitution of mod_g729 by mod_com_g729
	causes core dump
In-Reply-To: <4CA9F352.4070908@sns.eu>
References: <4CA9B34C.5070400@sns.eu>	<1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>	<4CA9CB28.8010101@sns.eu>	<D76D3E6E-7576-4299-BF16-9A7B29BFC37F@ipeva.fr>
	<7CAB482F-5FFA-491F-8BF6-EAE4E376C6F1@freeswitch.org>
	<4CA9F352.4070908@sns.eu>
Message-ID: <57C68F46-F99F-4F36-9C3D-ED70F9A79B1B@freeswitch.org>

Then you should always test your system before you blindly update to a new release.

/b

On Oct 4, 2010, at 10:31 AM, Jan Riedinger wrote:

> Hi Brian,
> 
> is there way, how I can "verify" that the an installed mod_com_g729 works with a new Freeswitch version? I frighten to get trouble when I will update a system with a lot of live traffic. In  this case it would be bad, if you  have to wait some time until the link for the right installer gets available.
> 
> BR
>    Jan
> 
> Am 04.10.2010 15:15, schrieb Brian West:
>> http://files.freeswitch.org/g729/fsg729-158-installer
>> 
>> Just the CDN doesn't have the index updated yet.  That will fix your issue.
>> 
>> /b
> 



From riedinger at sns.eu  Mon Oct  4 08:40:26 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Mon, 04 Oct 2010 17:40:26 +0200
Subject: [Freeswitch-users] Substitution of mod_g729 by mod_com_g729
 causes core dump
In-Reply-To: <57C68F46-F99F-4F36-9C3D-ED70F9A79B1B@freeswitch.org>
References: <4CA9B34C.5070400@sns.eu>	<1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>	<4CA9CB28.8010101@sns.eu>	<D76D3E6E-7576-4299-BF16-9A7B29BFC37F@ipeva.fr>
	<7CAB482F-5FFA-491F-8BF6-EAE4E376C6F1@freeswitch.org>
	<4CA9F352.4070908@sns.eu>
	<57C68F46-F99F-4F36-9C3D-ED70F9A79B1B@freeswitch.org>
Message-ID: <4CA9F56A.7020409@sns.eu>

  How can I test a system without a second licence file? Is it needed to 
buy licences even for the test system?

BR
     Jan

Am 04.10.2010 17:37, schrieb Brian West:
> Then you should always test your system before you blindly update to a new release.
>
> /b
>
> On Oct 4, 2010, at 10:31 AM, Jan Riedinger wrote:
>
>> Hi Brian,
>>
>> is there way, how I can "verify" that the an installed mod_com_g729 works with a new Freeswitch version? I frighten to get trouble when I will update a system with a lot of live traffic. In  this case it would be bad, if you  have to wait some time until the link for the right installer gets available.
>>
>> BR
>>     Jan
>>
>> Am 04.10.2010 15:15, schrieb Brian West:
>>> http://files.freeswitch.org/g729/fsg729-158-installer
>>>
>>> Just the CDN doesn't have the index updated yet.  That will fix your issue.
>>>
>>> /b

-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                         E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973



From david.ponzone at ipeva.fr  Mon Oct  4 08:44:03 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 4 Oct 2010 17:44:03 +0200
Subject: [Freeswitch-users] Substitution of mod_g729 by mod_com_g729
	causes core dump
In-Reply-To: <4CA9F352.4070908@sns.eu>
References: <4CA9B34C.5070400@sns.eu>	<1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>	<4CA9CB28.8010101@sns.eu>	<D76D3E6E-7576-4299-BF16-9A7B29BFC37F@ipeva.fr>
	<7CAB482F-5FFA-491F-8BF6-EAE4E376C6F1@freeswitch.org>
	<4CA9F352.4070908@sns.eu>
Message-ID: <64D89B85-41A8-431C-86DD-8239725CB20E@ipeva.fr>

I would recommend you become careful on your FS upgrades when you need/ 
use mod_com_g729.
Hang out on #freeswitch or here, and ask around, if you really need to  
upgrade FS.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 04/10/2010 ? 17:31, Jan Riedinger a ?crit :

>  Hi Brian,
>
> is there way, how I can "verify" that the an installed mod_com_g729
> works with a new Freeswitch version? I frighten to get trouble when I
> will update a system with a lot of live traffic. In  this case it  
> would
> be bad, if you  have to wait some time until the link for the right
> installer gets available.
>
> BR
>     Jan
>
> Am 04.10.2010 15:15, schrieb Brian West:
>> http://files.freeswitch.org/g729/fsg729-158-installer
>>
>> Just the CDN doesn't have the index updated yet.  That will fix  
>> your issue.
>>
>> /b
>>
>> On Oct 4, 2010, at 8:09 AM, David Ponzone wrote:
>>
>>> Nope, you have to install them manually.
>>> Current is v158.
>>> You can download it from the same place than 153, but via direct  
>>> URL only.
>>> It's still Beta I think.
>>>
>>> David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>>> ?tablis ? l'intention exclusive de ses destinataires. Toute  
>>> utilisation ou diffusion non autoris?e est interdite. Tout message  
>>> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
>>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form?  
>>> ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci  
>>> de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> -- 
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                         E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>
> AG Charlottenburg - HRB 71973
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at ipeva.fr  Mon Oct  4 08:46:30 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 4 Oct 2010 17:46:30 +0200
Subject: [Freeswitch-users] mod_com_g729 licences used even without
	transcoding
In-Reply-To: <4CA9F0C6.8060005@sns.eu>
References: <4CA9B34C.5070400@sns.eu>	<1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>	<4CA9CB28.8010101@sns.eu>	<D76D3E6E-7576-4299-BF16-9A7B29BFC37F@ipeva.fr>
	<7CAB482F-5FFA-491F-8BF6-EAE4E376C6F1@freeswitch.org>
	<4CA9F0C6.8060005@sns.eu>
Message-ID: <FFF3584F-4412-4EEE-AA98-0D266DE5DE0C@ipeva.fr>

I just discovered a situation like that.
If the ptime of leg A and leg B does not match, FS will decode/encode  
the stream in order to re-packetize (despite the fact that  
mod_com_g729 is capable to re-packetize even without any license  
installed).

I was thinking that it was not optimal, but Anthony explained some  
minutes ago on #freeswitch why this was the preferred behaviour,  
despite the quality loss it causes.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 04/10/2010 ? 17:20, Jan Riedinger a ?crit :

>  Using http://files.freeswitch.org/g729/fsg729-158-installer really  
> did
> fix the issue described below.
>
> However, it seems that the licence is used (up), even if there is no  
> transcoding needed. In my CDR I find for all calls, for which G729  
> is involed at all,
>
> read_codec="G729", read_rate="8000", write_codec="G729",  
> write_rate="8000"
>
> Nevertheless, all of these calls use a G729 licence. Is there any  
> way to
> find out, why mod_com_g729 wants to do any kind of transcoding?
>
> BR
>     Jan
>
> now the cod
>
> Am 04.10.2010 15:15, schrieb Brian West:
>> http://files.freeswitch.org/g729/fsg729-158-installer
>>
>> Just the CDN doesn't have the index updated yet.  That will fix  
>> your issue.
>>
>> /b
>>
>> On Oct 4, 2010, at 8:09 AM, David Ponzone wrote:
>>
>>> Nope, you have to install them manually.
>>> Current is v158.
>>> You can download it from the same place than 153, but via direct  
>>> URL only.
>>> It's still Beta I think.
>>>
>>> David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>>> ?tablis ? l'intention exclusive de ses destinataires. Toute  
>>> utilisation ou diffusion non autoris?e est interdite. Tout message  
>>> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
>>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form?  
>>> ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci  
>>> de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> -- 
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                         E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>
> AG Charlottenburg - HRB 71973
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at ipeva.fr  Mon Oct  4 08:46:55 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 4 Oct 2010 17:46:55 +0200
Subject: [Freeswitch-users] Substitution of mod_g729 by mod_com_g729
	causes core dump
In-Reply-To: <4CA9F56A.7020409@sns.eu>
References: <4CA9B34C.5070400@sns.eu>	<1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>	<4CA9CB28.8010101@sns.eu>	<D76D3E6E-7576-4299-BF16-9A7B29BFC37F@ipeva.fr>
	<7CAB482F-5FFA-491F-8BF6-EAE4E376C6F1@freeswitch.org>
	<4CA9F352.4070908@sns.eu>
	<57C68F46-F99F-4F36-9C3D-ED70F9A79B1B@freeswitch.org>
	<4CA9F56A.7020409@sns.eu>
Message-ID: <00CE2613-5926-4CE9-BC0B-0DB31E520F7F@ipeva.fr>

Jan,

a on test system, obivously, one license should be enough.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 04/10/2010 ? 17:40, Jan Riedinger a ?crit :

>  How can I test a system without a second licence file? Is it needed  
> to
> buy licences even for the test system?
>
> BR
>     Jan
>
> Am 04.10.2010 17:37, schrieb Brian West:
>> Then you should always test your system before you blindly update  
>> to a new release.
>>
>> /b
>>
>> On Oct 4, 2010, at 10:31 AM, Jan Riedinger wrote:
>>
>>> Hi Brian,
>>>
>>> is there way, how I can "verify" that the an installed  
>>> mod_com_g729 works with a new Freeswitch version? I frighten to  
>>> get trouble when I will update a system with a lot of live  
>>> traffic. In  this case it would be bad, if you  have to wait some  
>>> time until the link for the right installer gets available.
>>>
>>> BR
>>>    Jan
>>>
>>> Am 04.10.2010 15:15, schrieb Brian West:
>>>> http://files.freeswitch.org/g729/fsg729-158-installer
>>>>
>>>> Just the CDN doesn't have the index updated yet.  That will fix  
>>>> your issue.
>>>>
>>>> /b
>
> -- 
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                         E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>
> AG Charlottenburg - HRB 71973
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From b_ball_henry at hotmail.com  Mon Oct  4 08:48:33 2010
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Mon, 4 Oct 2010 23:48:33 +0800
Subject: [Freeswitch-users] High Availability and hostname used by Sofia
In-Reply-To: <AANLkTikJacUOvCCOwc_LNvNRwdVzfoYE4jNchxPBRPS-@mail.gmail.com>
References: <EA736C6E-2651-43A6-BE6B-D202DC951F38@scarlet-internet.nl>
	<AANLkTikJacUOvCCOwc_LNvNRwdVzfoYE4jNchxPBRPS-@mail.gmail.com>
Message-ID: <AANLkTimF36eFM1Cjvni7t3LqFGEUUapzhM8sm6y8=ys-@mail.gmail.com>

Anthony:

But if I the FS servers do not share a floating IP but does share the
registration DB via Core ODBC. Even though the same domain is forced in the
sofia profile, the registrations will not be treated as they are on every FS
box I have correct?

The idea is to have OpenSIPS in front of FS servers to distribute
registrations. But would like to have any of the FS servers be able to
contact the registered user on the shared DB. Is it possible somehow?


Henry Huang Unified Communication System R&D, Founder of UniC Solution
US: +1 (626) 606-3306 | ??(Taiwan): +886 933847619
Contact Me [image:
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IM [image: Google Talk/] red_rain_seven at gmail.com [image:
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MSN/] b_ball_henry at hotmail.com
?"You are the embodiment of the information you choose to accept and act
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<http://www.wisestamp.com/apps/quotes?utm_source=extension&utm_medium=email&utm_term=quotes&utm_campaign=apps>



On Mon, Oct 4, 2010 at 10:27 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> the registrations table intentionally does not consider the hostname
> so you can cluster many boxes on the same domain.
>
> On Mon, Oct 4, 2010 at 6:57 AM, Leon de Rooij <leon at scarlet-internet.nl>
> wrote:
> > Hello List,
> >
> > This is going to be a long mail, please bear with me :-)
> >
> > I'd like to configure High Availability for a FreeSWITCH installation
> > by using a failover server. I already discussed it a bit with Sekil on
> > IRC, but I thought it'd be useful to discuss it here as well.
> >
> >
> > The idea is as follows:
> >
> > - I have 2 servers both with an identical FreeSWITCH installation, FS1
> > and FS2.
> > - Both servers have different static IP's but also one floating IP by
> > means of heartbeat / haresources.
> > - Both FS instances have one Sofia SIP profile that use the floating IP.
> > - Both FS instances are always started (so the backup server will fail
> > to load its SIP profile because the IP is not configured - which is
> > expected behaviour)
> > - There'll be a /etc/ha.d/resource.d "resource script" that starts the
> > SIP profile (through fs_cli -x) the moment it becomes primary
> > - The resource script will also send a "sofia recover" after the SIP
> > profile has been brought up
> >
> >
> > This way:
> >
> > - There'll be one logical switch.
> > - The moment FS2 becomes primary, users will be reachable through the
> > entry in 'sip_registrations' table in the ODBC db, even though they
> > may have registered on FS1
> > - FS2 will start to send SIP options to NAT'ed useragents (keepalive /
> > udp hole punching) the moment the SIP profile is started.
> > - No client useragent involvement is necessary when failing over (like
> > re-register)
> > - Running calls will be continued because of the "sofia recover"
> >
> >
> > Only problem here is that when a useragent registers to us, then FS
> > also stores its own hostname in "sip_registrations" table in the
> > columns "hostname" and "orig_hostname" (from
> > mod_sofia_globals.hostname). This column is also used in the SELECT
> > query when bridging to a user - so that means if the user registered
> > to FS1, then a failover happens, then if a call is bridged towards
> > that user on FS2, it won't find it because it will search with FS2's
> > hostname in the SELECT query, right ?
> >
> > If everything I'm writing so far is correct, then would it be alright
> > to override the hostname in sofia.conf ? I think that would solve my
> > problem. I already looked it up - it is first set in mod_sofia.c at
> > line 4565 through gethostname(). It can be overriden in sofia.c
> > (function config_sofia) around line 2845. It would probably also work
> > to give both hosts an identical hostname in the OS, but I don't like
> > that (because of rsyslog and more).
> >
> >
> > I'm not convinced yet on how to proceed, please let me know if I'm
> > going the wrong direction. If I'm successful on getting this running,
> > I'll document everything on the wiki - also the haresources script I
> > already wrote I'll put in my contrib dir (or elsewhere in main tree if
> > preferred).
> >
> >
> > Thanks for your time,
> >
> > Leon
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mnhassan at usa.net  Mon Oct  4 08:55:03 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Mon, 4 Oct 2010 21:55:03 +0600
Subject: [Freeswitch-users] mod_com_g729 licences used even without
	transcoding
In-Reply-To: <4CA9F0C6.8060005@sns.eu>
References: <4CA9B34C.5070400@sns.eu>
	<1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>
	<4CA9CB28.8010101@sns.eu>
	<D76D3E6E-7576-4299-BF16-9A7B29BFC37F@ipeva.fr>
	<7CAB482F-5FFA-491F-8BF6-EAE4E376C6F1@freeswitch.org>
	<4CA9F0C6.8060005@sns.eu>
Message-ID: <AANLkTinj5R9UTFcGVoFWxP4yRncedntraG-gzsNAmTcE@mail.gmail.com>

Is there any IVR or music on hold involved?

Regards
HASSAN


On 2010-10-04, Jan Riedinger <riedinger at sns.eu> wrote:
>   Using http://files.freeswitch.org/g729/fsg729-158-installer really did
> fix the issue described below.
>
> However, it seems that the licence is used (up), even if there is no
> transcoding needed. In my CDR I find for all calls, for which G729 is
> involed at all,
>
> read_codec="G729", read_rate="8000", write_codec="G729", write_rate="8000"
>
> Nevertheless, all of these calls use a G729 licence. Is there any way to
> find out, why mod_com_g729 wants to do any kind of transcoding?
>
> BR
>      Jan
>
> now the cod
>
> Am 04.10.2010 15:15, schrieb Brian West:
>> http://files.freeswitch.org/g729/fsg729-158-installer
>>
>> Just the CDN doesn't have the index updated yet.  That will fix your
>> issue.
>>
>> /b
>>
>> On Oct 4, 2010, at 8:09 AM, David Ponzone wrote:
>>
>>> Nope, you have to install them manually.
>>> Current is v158.
>>> You can download it from the same place than 153, but via direct URL
>>> only.
>>> It's still Beta I think.
>>>
>>> David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou
>>> diffusion non autoris?e est interdite. Tout message ?lectronique est
>>> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de
>>> ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et
>>> d'avertir l'exp?diteur.
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> --
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                          E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>
> AG Charlottenburg - HRB 71973
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device


From mnhassan at usa.net  Mon Oct  4 08:57:21 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Mon, 4 Oct 2010 21:57:21 +0600
Subject: [Freeswitch-users] mod_com_g729 licences used even without
	transcoding
In-Reply-To: <FFF3584F-4412-4EEE-AA98-0D266DE5DE0C@ipeva.fr>
References: <4CA9B34C.5070400@sns.eu>
	<1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>
	<4CA9CB28.8010101@sns.eu>
	<D76D3E6E-7576-4299-BF16-9A7B29BFC37F@ipeva.fr>
	<7CAB482F-5FFA-491F-8BF6-EAE4E376C6F1@freeswitch.org>
	<4CA9F0C6.8060005@sns.eu>
	<FFF3584F-4412-4EEE-AA98-0D266DE5DE0C@ipeva.fr>
Message-ID: <AANLkTik9z2CqUeEeXwPXx8kNBa9=fZ17Vb63EyYKjEZe@mail.gmail.com>

Could you please send a copy of the explanation on mirror?

Regards
HASSAN


On 2010-10-04, David Ponzone <david.ponzone at ipeva.fr> wrote:
> I just discovered a situation like that.
> If the ptime of leg A and leg B does not match, FS will decode/encode
> the stream in order to re-packetize (despite the fact that
> mod_com_g729 is capable to re-packetize even without any license
> installed).
>
> I was thinking that it was not optimal, but Anthony explained some
> minutes ago on #freeswitch why this was the preferred behaviour,
> despite the quality loss it causes.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
> ? l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion non autoris?e est interdite. Tout message ?lectronique est
> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre
> de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes
> pas destinataire de ce message, merci de le d?truire imm?diatement et
> d'avertir l'exp?diteur.
>
>
>
>
> Le 04/10/2010 ? 17:20, Jan Riedinger a ?crit :
>
>>  Using http://files.freeswitch.org/g729/fsg729-158-installer really
>> did
>> fix the issue described below.
>>
>> However, it seems that the licence is used (up), even if there is no
>> transcoding needed. In my CDR I find for all calls, for which G729
>> is involed at all,
>>
>> read_codec="G729", read_rate="8000", write_codec="G729",
>> write_rate="8000"
>>
>> Nevertheless, all of these calls use a G729 licence. Is there any
>> way to
>> find out, why mod_com_g729 wants to do any kind of transcoding?
>>
>> BR
>>     Jan
>>
>> now the cod
>>
>> Am 04.10.2010 15:15, schrieb Brian West:
>>> http://files.freeswitch.org/g729/fsg729-158-installer
>>>
>>> Just the CDN doesn't have the index updated yet.  That will fix
>>> your issue.
>>>
>>> /b
>>>
>>> On Oct 4, 2010, at 8:09 AM, David Ponzone wrote:
>>>
>>>> Nope, you have to install them manually.
>>>> Current is v158.
>>>> You can download it from the same place than 153, but via direct
>>>> URL only.
>>>> It's still Beta I think.
>>>>
>>>> David Ponzone  Direction Technique
>>>> email: david.ponzone at ipeva.fr
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>>
>>>> Service Client IPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>
>>>> Ce message et toutes les pi?ces jointes sont confidentiels et
>>>> ?tablis ? l'intention exclusive de ses destinataires. Toute
>>>> utilisation ou diffusion non autoris?e est interdite. Tout message
>>>> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute
>>>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form?
>>>> ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci
>>>> de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> --
>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>                                         E-Mail:  riedinger at sns.eu
>> SNS Consult GmbH                        ICQ   :  163-237-041
>> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>
>> AG Charlottenburg - HRB 71973
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>

-- 
Sent from my mobile device


From adminjew at gmail.com  Mon Oct  4 09:04:47 2010
From: adminjew at gmail.com (Yitzchok)
Date: Mon, 4 Oct 2010 12:04:47 -0400
Subject: [Freeswitch-users] mod_managed on linux centos
In-Reply-To: <1286205151753-5599563.post@n2.nabble.com>
References: <AANLkTikg4bxav+e04fAmJ_dKTHKERAPk715h9oHeYMRh@mail.gmail.com>
	<1283347083542-5486890.post@n2.nabble.com>
	<AANLkTi=xt9CZBPZLnBX56e2Rjdc_FRf_=er6Yy7w7ZuN@mail.gmail.com>
	<AANLkTi=S=tpzczDM8WSm6N8AeTkN3XNTPda6048sRoJ9@mail.gmail.com>
	<4C801B6E.4050103@puzzled.xs4all.nl>
	<AANLkTimzjbLHbObgsQKqX3t-hfaZWr95UGMi17YxZFHS@mail.gmail.com>
	<4C838C3B.3070202@puzzled.xs4all.nl>
	<AANLkTinP+UfvkM0dmBycy5GqivAXNHGF9RnTBNvV0gV4@mail.gmail.com>
	<AANLkTik1CY5-6DUvsLg512bGRr66xLF5H8Gi7WYRenHx@mail.gmail.com>
	<1286205151753-5599563.post@n2.nabble.com>
Message-ID: <AANLkTinKemVBa5HNSrJk6EOfWYM5wd7CyL4L-JX-BawQ@mail.gmail.com>

I have already done that but it doesn't help also note that on the first
call it works fine.


Yitzchok


On Mon, Oct 4, 2010 at 11:12 AM, Jeff Lenk <jeff at jefflenk.com> wrote:

>
> Have you seen:
> http://wiki.freeswitch.org/wiki/Mod_mono
>
> <dllmap dll="mod_managed.dll" target="mod_managed.so" os="!windows"/>
>
> regarding null exceptions?
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/mod-managed-on-linux-centos-tp5485480p5599563.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From riedinger at sns.eu  Mon Oct  4 09:11:05 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Mon, 04 Oct 2010 18:11:05 +0200
Subject: [Freeswitch-users] mod_com_g729 licences used even
	without	transcoding
In-Reply-To: <FFF3584F-4412-4EEE-AA98-0D266DE5DE0C@ipeva.fr>
References: <4CA9B34C.5070400@sns.eu>	<1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>	<4CA9CB28.8010101@sns.eu>	<D76D3E6E-7576-4299-BF16-9A7B29BFC37F@ipeva.fr>	<7CAB482F-5FFA-491F-8BF6-EAE4E376C6F1@freeswitch.org>	<4CA9F0C6.8060005@sns.eu>
	<FFF3584F-4412-4EEE-AA98-0D266DE5DE0C@ipeva.fr>
Message-ID: <4CA9FC99.5060202@sns.eu>

  I think, it would be a could idea to make it configurable, if 
mod_com_g729 shall decode/encode the stream to re-packetize. With the 
current behaviour its usage is  too expensive for me in respect of money 
needed for licences and CPU power. Most of my calls are calls are g729 
to g729, transcodeding is needed only very rare.



Am 04.10.2010 17:46, schrieb David Ponzone:
> I just discovered a situation like that.
> If the ptime of leg A and leg B does not match, FS will decode/encode 
> the stream in order to re-packetize (despite the fact that 
> mod_com_g729 is capable to re-packetize even without any license 
> installed).
>
> I was thinking that it was not optimal, but Anthony explained some 
> minutes ago on #freeswitch why this was the preferred behaviour, 
> despite the quality loss it causes.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IP eva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 04/10/2010 ? 17:20, Jan Riedinger a ?crit :
>
>>  Using http://files.freeswitch.org/g729/fsg729-158-installer really did
>> fix the issue described below.
>>
>> However, it seems that the licence is used (up), even if there is no 
>> transcoding needed. In my CDR I find for all calls, for which G729 is 
>> involed at all,
>>
>> read_codec="G729", read_rate="8000", write_codec="G729", 
>> write_rate="8000"
>>
>> Nevertheless, all of these calls use a G729 licence. Is there any way to
>> find out, why mod_com_g729 wants to do any kind of transcoding?
>>
>> BR
>>     Jan
>>
>> now the cod
>>
>> Am 04.10.2010 15:15, schrieb Brian West:
>>> http://files.freeswitch.org/g729/fsg729-158-installer
>>>
>>> Just the CDN doesn't have the index updated yet.  That will fix your 
>>> issue.
>>>
>>> /b
>>>
>>> On Oct 4, 2010, at 8:09 AM, David Ponzone wrote:
>>>
>>>> Nope, you have to install them manually.
>>>> Current is v158.
>>>> You can download it from the same place than 153, but via direct 
>>>> URL only.
>>>> It's still Beta I think.
>>>>
>>>> David Ponzone  Direction Technique
>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>>
>>>> Service Client IPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr  - www.ipeva-studio.com <http://www.ipeva-studio.com>
>>>>
>>>> Ce message et toutes les pi?ces jointes sont confidentiels et 
>>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>>> utilisation ou diffusion non autoris?e est interdite. Tout message 
>>>> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute 
>>>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou 
>>>> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de 
>>>> le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org 
>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> -- 
>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>                                         E-Mail: riedinger at sns.eu 
>> <mailto:riedinger at sns.eu>
>> SNS Consult GmbH                        ICQ   :  163-237-041
>> S?dwestkorso 49a                        MSN   : jan at sns-consult.de 
>> <mailto:jan at sns-consult.de>
>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>
>> AG Charlottenburg - HRB 71973
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                         E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973

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From Russell.Mosemann at cune.org  Mon Oct  4 09:02:29 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Mon, 4 Oct 2010 16:02:29 -0000
Subject: [Freeswitch-users] Accessing wanpipe from openVZ container
In-Reply-To: <AANLkTimHaPoj=7JoAVLGT-uq8sJyD8vAhZNXYAsmU0rj@mail.gmail.com>
Message-ID: <20101004160229.EED21337647@cuneorg-email.cune.pri>

Moises Silva <moises.silva at gmail.com> said:

> I have no experience with openVZ, but I know some of our support staff
> got it working with xen just fine. I think openVZ should not be a
> problem.

I don't think there will be any problems. I have gotten this arrangement
to work nicely with a different manufacturer's card. The devices are
exported to the container so that they show up in /dev, as one would
normally expect. The difference is that the drivers are in the host node,
and the signaling is in the container.

> No tools to do it automatically if you want to do it separate.

Darn, I was hoping it would be as simple a "make install_driver". I will
dig into the installation files and figure out how to do it manually. Thanks.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From david.ponzone at ipeva.fr  Mon Oct  4 09:56:20 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 4 Oct 2010 18:56:20 +0200
Subject: [Freeswitch-users] mod_com_g729 licences used even without
	transcoding
In-Reply-To: <4CA9FC99.5060202@sns.eu>
References: <4CA9B34C.5070400@sns.eu>	<1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>	<4CA9CB28.8010101@sns.eu>	<D76D3E6E-7576-4299-BF16-9A7B29BFC37F@ipeva.fr>	<7CAB482F-5FFA-491F-8BF6-EAE4E376C6F1@freeswitch.org>	<4CA9F0C6.8060005@sns.eu>
	<FFF3584F-4412-4EEE-AA98-0D266DE5DE0C@ipeva.fr>
	<4CA9FC99.5060202@sns.eu>
Message-ID: <D3248AA8-EB9E-40A2-B49E-4E8391B4F09C@ipeva.fr>

You could also check with your carrier why they dont accept the custom  
ptime you need, or normalize your customers' config.

I have a nice extension I made that denies calls if ptime is not 20ms,  
with early-neg or late-neg:

   <extension name="check-ptime" continue="true">
     <condition field="${sip_use_codec_ptime}" expression="20"  
break="on-true"/>
     <condition field="${ep_codec_string}" expression="@20i">
       <anti-action application="respond" data="488"/>
       <anti-action application="hangup" data="88"/>
     </condition>
   </extension>

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 04/10/2010 ? 18:11, Jan Riedinger a ?crit :

> I think, it would be a could idea to make it configurable, if  
> mod_com_g729 shall decode/encode the stream to re-packetize. With  
> the current behaviour its usage is  too expensive for me in respect  
> of money needed for licences and CPU power. Most of my calls are  
> calls are g729 to g729, transcodeding is needed only very rare.
>
>
>
> Am 04.10.2010 17:46, schrieb David Ponzone:
>>
>> I just discovered a situation like that.
>> If the ptime of leg A and leg B does not match, FS will decode/ 
>> encode the stream in order to re-packetize (despite the fact that  
>> mod_com_g729 is capable to re-packetize even without any license  
>> installed).
>>
>> I was thinking that it was not optimal, but Anthony explained some  
>> minutes ago on #freeswitch why this was the preferred behaviour,  
>> despite the quality loss it causes.
>>
>> David Ponzone   Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client  IP eva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ? l'intention exclusive de ses destinataires. Toute  
>> utilisation ou diffusion non autoris?e est interdite. Tout message  
>> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
>> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de  
>> le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>>
>> Le 04/10/2010 ? 17:20, Jan Riedinger a ?crit :
>>
>>>  Using http://files.freeswitch.org/g729/fsg729-158-installer  
>>> really did
>>> fix the issue described below.
>>>
>>> However, it seems that the licence is used (up), even if there is  
>>> no transcoding needed. In my CDR I find for all calls, for which  
>>> G729 is involed at all,
>>>
>>> read_codec="G729", read_rate="8000", write_codec="G729",  
>>> write_rate="8000"
>>>
>>> Nevertheless, all of these calls use a G729 licence. Is there any  
>>> way to
>>> find out, why mod_com_g729 wants to do any kind of transcoding?
>>>
>>> BR
>>>     Jan
>>>
>>> now the cod
>>>
>>> Am 04.10.2010 15:15, schrieb Brian West:
>>>> http://files.freeswitch.org/g729/fsg729-158-installer
>>>>
>>>> Just the CDN doesn't have the index updated yet.  That will fix  
>>>> your issue.
>>>>
>>>> /b
>>>>
>>>> On Oct 4, 2010, at 8:09 AM, David Ponzone wrote:
>>>>
>>>>> Nope, you have to install them manually.
>>>>> Current is v158.
>>>>> You can download it from the same place than 153, but via direct  
>>>>> URL only.
>>>>> It's still Beta I think.
>>>>>
>>>>> David Ponzone  Direction Technique
>>>>> email: david.ponzone at ipeva.fr
>>>>> tel:      01 74 03 18 97
>>>>> gsm:   06 66 98 76 34
>>>>>
>>>>> Service Client IPeva
>>>>> tel:      0811 46 26 26
>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>
>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>>>>> ?tablis ? l'intention exclusive de ses destinataires. Toute  
>>>>> utilisation ou diffusion non autoris?e est interdite. Tout  
>>>>> message ?lectronique est susceptible d'alt?ration. IPeva d?cline  
>>>>> toute responsabilit? au titre de ce message s'il a ?t? alt?r?,  
>>>>> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
>>>>> message, merci de le d?truire imm?diatement et d'avertir  
>>>>> l'exp?diteur.
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> -- 
>>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>>                                         E-Mail:  riedinger at sns.eu
>>> SNS Consult GmbH                        ICQ   :  163-237-041
>>> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
>>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>>
>>> AG Charlottenburg - HRB 71973
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> -- 
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                         E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>
> AG Charlottenburg - HRB 71973

-------------- next part --------------
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From david.ponzone at ipeva.fr  Mon Oct  4 09:56:20 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 4 Oct 2010 18:56:20 +0200
Subject: [Freeswitch-users] mod_com_g729 licences used even without
	transcoding
In-Reply-To: <4CA9FC99.5060202@sns.eu>
References: <4CA9B34C.5070400@sns.eu>	<1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>	<4CA9CB28.8010101@sns.eu>	<D76D3E6E-7576-4299-BF16-9A7B29BFC37F@ipeva.fr>	<7CAB482F-5FFA-491F-8BF6-EAE4E376C6F1@freeswitch.org>	<4CA9F0C6.8060005@sns.eu>
	<FFF3584F-4412-4EEE-AA98-0D266DE5DE0C@ipeva.fr>
	<4CA9FC99.5060202@sns.eu>
Message-ID: <D3248AA8-EB9E-40A2-B49E-4E8391B4F09C@ipeva.fr>

You could also check with your carrier why they dont accept the custom  
ptime you need, or normalize your customers' config.

I have a nice extension I made that denies calls if ptime is not 20ms,  
with early-neg or late-neg:

   <extension name="check-ptime" continue="true">
     <condition field="${sip_use_codec_ptime}" expression="20"  
break="on-true"/>
     <condition field="${ep_codec_string}" expression="@20i">
       <anti-action application="respond" data="488"/>
       <anti-action application="hangup" data="88"/>
     </condition>
   </extension>

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 04/10/2010 ? 18:11, Jan Riedinger a ?crit :

> I think, it would be a could idea to make it configurable, if  
> mod_com_g729 shall decode/encode the stream to re-packetize. With  
> the current behaviour its usage is  too expensive for me in respect  
> of money needed for licences and CPU power. Most of my calls are  
> calls are g729 to g729, transcodeding is needed only very rare.
>
>
>
> Am 04.10.2010 17:46, schrieb David Ponzone:
>>
>> I just discovered a situation like that.
>> If the ptime of leg A and leg B does not match, FS will decode/ 
>> encode the stream in order to re-packetize (despite the fact that  
>> mod_com_g729 is capable to re-packetize even without any license  
>> installed).
>>
>> I was thinking that it was not optimal, but Anthony explained some  
>> minutes ago on #freeswitch why this was the preferred behaviour,  
>> despite the quality loss it causes.
>>
>> David Ponzone   Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client  IP eva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ? l'intention exclusive de ses destinataires. Toute  
>> utilisation ou diffusion non autoris?e est interdite. Tout message  
>> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
>> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de  
>> le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>>
>> Le 04/10/2010 ? 17:20, Jan Riedinger a ?crit :
>>
>>>  Using http://files.freeswitch.org/g729/fsg729-158-installer  
>>> really did
>>> fix the issue described below.
>>>
>>> However, it seems that the licence is used (up), even if there is  
>>> no transcoding needed. In my CDR I find for all calls, for which  
>>> G729 is involed at all,
>>>
>>> read_codec="G729", read_rate="8000", write_codec="G729",  
>>> write_rate="8000"
>>>
>>> Nevertheless, all of these calls use a G729 licence. Is there any  
>>> way to
>>> find out, why mod_com_g729 wants to do any kind of transcoding?
>>>
>>> BR
>>>     Jan
>>>
>>> now the cod
>>>
>>> Am 04.10.2010 15:15, schrieb Brian West:
>>>> http://files.freeswitch.org/g729/fsg729-158-installer
>>>>
>>>> Just the CDN doesn't have the index updated yet.  That will fix  
>>>> your issue.
>>>>
>>>> /b
>>>>
>>>> On Oct 4, 2010, at 8:09 AM, David Ponzone wrote:
>>>>
>>>>> Nope, you have to install them manually.
>>>>> Current is v158.
>>>>> You can download it from the same place than 153, but via direct  
>>>>> URL only.
>>>>> It's still Beta I think.
>>>>>
>>>>> David Ponzone  Direction Technique
>>>>> email: david.ponzone at ipeva.fr
>>>>> tel:      01 74 03 18 97
>>>>> gsm:   06 66 98 76 34
>>>>>
>>>>> Service Client IPeva
>>>>> tel:      0811 46 26 26
>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>
>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>>>>> ?tablis ? l'intention exclusive de ses destinataires. Toute  
>>>>> utilisation ou diffusion non autoris?e est interdite. Tout  
>>>>> message ?lectronique est susceptible d'alt?ration. IPeva d?cline  
>>>>> toute responsabilit? au titre de ce message s'il a ?t? alt?r?,  
>>>>> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
>>>>> message, merci de le d?truire imm?diatement et d'avertir  
>>>>> l'exp?diteur.
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> -- 
>>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>>                                         E-Mail:  riedinger at sns.eu
>>> SNS Consult GmbH                        ICQ   :  163-237-041
>>> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
>>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>>
>>> AG Charlottenburg - HRB 71973
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> -- 
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                         E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>
> AG Charlottenburg - HRB 71973

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From riedinger at sns.eu  Mon Oct  4 10:18:13 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Mon, 04 Oct 2010 19:18:13 +0200
Subject: [Freeswitch-users] mod_com_g729 licences used even
	without	transcoding
In-Reply-To: <D3248AA8-EB9E-40A2-B49E-4E8391B4F09C@ipeva.fr>
References: <4CA9B34C.5070400@sns.eu>	<1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>	<4CA9CB28.8010101@sns.eu>	<D76D3E6E-7576-4299-BF16-9A7B29BFC37F@ipeva.fr>	<7CAB482F-5FFA-491F-8BF6-EAE4E376C6F1@freeswitch.org>	<4CA9F0C6.8060005@sns.eu>	<FFF3584F-4412-4EEE-AA98-0D266DE5DE0C@ipeva.fr>	<4CA9FC99.5060202@sns.eu>
	<D3248AA8-EB9E-40A2-B49E-4E8391B4F09C@ipeva.fr>
Message-ID: <4CAA0C55.2040109@sns.eu>

  I'm trying to do wholesale business with some dozen of interconnects. 
It's hopeless to force all customers/clients to use the same ptime - and 
I think there is no real reason to force them. I'm already happy, if 
they all offer the usage of g729 ;-). Often they pass through the calls 
themselves.

BTW - I never understood, why 20 ms is the default frame size. In this 
way you have about 20 bytes payload and 40 bytes overhead for the TCP/IP 
header, which is a pretty bad ratio.

BR
     Jan

Am 04.10.2010 18:56, schrieb David Ponzone:
> You could also check with your carrier why they dont accept the custom 
> ptime you need, or normalize your customers' config.
>
> I have a nice extension I made that denies calls if ptime is not 20ms, 
> with early-neg or late-neg:
>
> <extension name="check-ptime" continue="true">
> <condition field="${sip_use_codec_ptime}" expression="20" 
> break="on-true"/>
> <condition field="${ep_codec_string}" expression="@20i">
> <anti-action application="respond" data="488"/>
> <anti-action application="hangup" data="88"/>
> </condition>
> </extension>
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IP eva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 04/10/2010 ? 18:11, Jan Riedinger a ?crit :
>
>> I think, it would be a could idea to make it configurable, if 
>> mod_com_g729 shall decode/encode the stream to re-packetize. With the 
>> current behaviour its usage is  too expensive for me in respect of 
>> money needed for licences and CPU power. Most of my calls are calls 
>> are g729 to g729, transcodeding is needed only very rare.
>>
>>
>>
>> Am 04.10.2010 17:46, schrieb David Ponzone:
>>> I just discovered a situation like that.
>>> If the ptime of leg A and leg B does not match, FS will 
>>> decode/encode the stream in order to re-packetize (despite the fact 
>>> that mod_com_g729 is capable to re-packetize even without any 
>>> license installed).
>>>
>>> I was thinking that it was not optimal, but Anthony explained some 
>>> minutes ago on #freeswitch why this was the preferred behaviour, 
>>> despite the quality loss it causes.
>>>
>>> David Ponzone Direction Technique
>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IP eva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
>>> <BLOCKED::http://www.ipeva-studio.com/>
>>>
>>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>> utilisation ou diffusion non autoris?e est interdite. Tout message 
>>> ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline toute 
>>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou 
>>> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le 
>>> d?truire imm?diatement et d'avertir l'exp?diteur./
>>> /
>>> /
>>>
>>>
>>>
>>> Le 04/10/2010 ? 17:20, Jan Riedinger a ?crit :
>>>
>>>>  Using http://files.freeswitch.org/g729/fsg729-158-installer really 
>>>> did
>>>> fix the issue described below.
>>>>
>>>> However, it seems that the licence is used (up), even if there is 
>>>> no transcoding needed. In my CDR I find for all calls, for which 
>>>> G729 is involed at all,
>>>>
>>>> read_codec="G729", read_rate="8000", write_codec="G729", 
>>>> write_rate="8000"
>>>>
>>>> Nevertheless, all of these calls use a G729 licence. Is there any 
>>>> way to
>>>> find out, why mod_com_g729 wants to do any kind of transcoding?
>>>>
>>>> BR
>>>>     Jan
>>>>
>>>> now the cod
>>>>
>>>> Am 04.10.2010 15:15, schrieb Brian West:
>>>>> http://files.freeswitch.org/g729/fsg729-158-installer
>>>>>
>>>>> Just the CDN doesn't have the index updated yet.  That will fix 
>>>>> your issue.
>>>>>
>>>>> /b
>>>>>
>>>>> On Oct 4, 2010, at 8:09 AM, David Ponzone wrote:
>>>>>
>>>>>> Nope, you have to install them manually.
>>>>>> Current is v158.
>>>>>> You can download it from the same place than 153, but via direct 
>>>>>> URL only.
>>>>>> It's still Beta I think.
>>>>>>
>>>>>> David Ponzone  Direction Technique
>>>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>>>> tel:      01 74 03 18 97
>>>>>> gsm:   06 66 98 76 34
>>>>>>
>>>>>> Service Client IPeva
>>>>>> tel:      0811 46 26 26
>>>>>> www.ipeva.fr <http://www.ipeva.fr>  - www.ipeva-studio.com 
>>>>>> <http://www.ipeva-studio.com>
>>>>>>
>>>>>> Ce message et toutes les pi?ces jointes sont confidentiels et 
>>>>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>>>>> utilisation ou diffusion non autoris?e est interdite. Tout 
>>>>>> message ?lectronique est susceptible d'alt?ration. IPeva d?cline 
>>>>>> toute responsabilit? au titre de ce message s'il a ?t? alt?r?, 
>>>>>> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce 
>>>>>> message, merci de le d?truire imm?diatement et d'avertir 
>>>>>> l'exp?diteur.
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org 
>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>> -- 
>>>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>>>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>>>                                         E-Mail: riedinger at sns.eu 
>>>> <mailto:riedinger at sns.eu>
>>>> SNS Consult GmbH                        ICQ   :  163-237-041
>>>> S?dwestkorso 49a                        MSN   : jan at sns-consult.de 
>>>> <mailto:jan at sns-consult.de>
>>>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>>>
>>>> AG Charlottenburg - HRB 71973
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org 
>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> -- 
>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>                                          E-Mail:riedinger at sns.eu  <mailto:riedinger at sns.eu>
>> SNS Consult GmbH                        ICQ   :  163-237-041
>> S?dwestkorso 49a                        MSN   :jan at sns-consult.de  <mailto:jan at sns-consult.de>
>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>
>> AG Charlottenburg - HRB 71973
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                         E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973

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From msc at freeswitch.org  Mon Oct  4 10:29:30 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Oct 2010 10:29:30 -0700
Subject: [Freeswitch-users] No autoanswer with Originate call
In-Reply-To: <1286181373.20572.5.camel@localhost>
References: <1286181373.20572.5.camel@localhost>
Message-ID: <AANLkTimmweNy+xdwQDuuuhTt0OEMfccpqfwUwYz50OHy@mail.gmail.com>

Can you pastebin a debug log of working vs. non-working? It would probably
be helpful to see the entire extension as well as a complete Lua script.
(You can write a minimal Lua script that demonstrates the behavior and then
others can test.) Of course, the most important question: are you running
latest HEAD? :)

-MC

On Mon, Oct 4, 2010 at 1:36 AM, Eli Hayun <elihayunfs at gmail.com> wrote:

>  Hi
> When I am dialing from A phone to B phone and I do :
>
>       <action application="set" data="sip_auto_answer=true" />
>        <action application="set"
> data="sip_h_Call-Info=<sip:${domain_name}>;answer-after=2" />
>
> everything is OK
>
> When I try from lua script
> api:executeString("originate [originate string] AUTO_ANS xml default")
>
> and in the dialplan I put the set autoanswer=true it is not working.
>
> any help?
>
> Eli
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Oct  4 10:33:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Oct 2010 10:33:26 -0700
Subject: [Freeswitch-users] sip to spa3102 fallback
In-Reply-To: <i8ajiu$2ce$1@dough.gmane.org>
References: <i8ajiu$2ce$1@dough.gmane.org>
Message-ID: <AANLkTimh9-4BMCz5_9b_NNcwrScAv8mcSwOA5wbpGv5C@mail.gmail.com>

How about using pipe-separated dial strings in your bridge app?
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Failover

-MC

On Sun, Oct 3, 2010 at 11:55 AM, Eric Masson <emss.mail at gmail.com> wrote:

> Hello,
>
> I have two sip providers configuration files in
> $(fs_conf}/conf/diaplan/default :
> - 00_keyyo.net.xml
> <include>
>   <extension name="keyyo" continue="false">
>     <condition field="destination_number" expression = "^(.*)$">
>       <action application="set" data="effective_caller_id_name=**MyID**"/>
>       <action application="set"
> data="effective_caller_id_number=**Myphonenumber"/>
>       <action application="bridge" data="sofia/gateway/keyyo.net/$1"/>
>     </condition>
>   </extension>
> </include>
>
> - 00_spa3102.xml
> <include>
>   <extension name="To PSTN">
>     <condition field="destination_number" expression="^(.*)$">
>       <action application="bridge"
> data="sofia/internal/${destination_number}@spafenmsf:5061" />
>     </condition>
>   </extension>
> </include>
>
> I'd like to implement the following scenario for outbound calls :
> - dial using "keyyo" if available
> - fallback to "To PSTN" if keyyo isn't available (not registered,
> already used, for example)
>
> I've googled for freeswitch failover setups but didn't find anything
> that could have enlightened me (I'm kind of new in fs).
>
> Where could I find docs to achieve the exposed goal please ?
>
> Kind Regards
>
> Eric Masson
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Mon Oct  4 10:34:26 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 Oct 2010 12:34:26 -0500
Subject: [Freeswitch-users] mod_com_g729 licences used even without
	transcoding
In-Reply-To: <4CAA0C55.2040109@sns.eu>
References: <4CA9B34C.5070400@sns.eu>
	<1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>
	<4CA9CB28.8010101@sns.eu>
	<D76D3E6E-7576-4299-BF16-9A7B29BFC37F@ipeva.fr>
	<7CAB482F-5FFA-491F-8BF6-EAE4E376C6F1@freeswitch.org>
	<4CA9F0C6.8060005@sns.eu>
	<FFF3584F-4412-4EEE-AA98-0D266DE5DE0C@ipeva.fr>
	<4CA9FC99.5060202@sns.eu>
	<D3248AA8-EB9E-40A2-B49E-4E8391B4F09C@ipeva.fr>
	<4CAA0C55.2040109@sns.eu>
Message-ID: <AANLkTim531sqEheNx-Mq4q93rwtqtVjfAK4qgk83uao3@mail.gmail.com>

because it's based on G.711 the original codec used.

It's a balance of latency to packet len.

G.729 is poorly packetized at 20ms with only 10 bytes per payload.

We can do something to make it work the way you want, but then I
refuse to care if you get jittery calls because you are sending the
wrong ptime in both directions.



On Mon, Oct 4, 2010 at 12:18 PM, Jan Riedinger <riedinger at sns.eu> wrote:
> I'm trying to do wholesale business with some dozen of interconnects. It's
> hopeless to force all customers/clients to use the same ptime - and I think
> there is no real reason to force them. I'm already happy, if they all offer
> the usage of g729 ;-). Often they pass through the calls themselves.
>
> BTW - I never understood, why 20 ms is the default frame size. In this way
> you have about 20 bytes payload and 40 bytes overhead for the TCP/IP header,
> which is a pretty bad ratio.
>
> BR
> ??? Jan
>
> Am 04.10.2010 18:56, schrieb David Ponzone:
>
> You could also check with your carrier why they dont accept the custom ptime
> you need, or normalize your customers' config.
> I have a nice extension I made that denies calls if ptime is not 20ms, with
> early-neg or late-neg:
> ??<extension name="check-ptime" continue="true">
> ?? ?<condition field="${sip_use_codec_ptime}" expression="20"
> break="on-true"/>
> ?? ?<condition field="${ep_codec_string}" expression="@20i">
> ?? ? ?<anti-action application="respond" data="488"/>
> ?? ? ?<anti-action application="hangup" data="88"/>
> ?? ?</condition>
> ??</extension>
> David Ponzone ? Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client? IP eva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 04/10/2010 ? 18:11, Jan Riedinger a ?crit :
>
> I think, it would be a could idea to make it configurable, if mod_com_g729
> shall decode/encode the stream to re-packetize. With the current behaviour
> its usage is? too expensive for me in respect of money needed for licences
> and CPU power. Most of my calls are calls are g729 to g729, transcodeding is
> needed only very rare.
>
>
>
> Am 04.10.2010 17:46, schrieb David Ponzone:
>
> I just discovered a situation like that.
> If the ptime of leg A and leg B does not match, FS will decode/encode the
> stream in order to re-packetize (despite the fact that mod_com_g729 is
> capable to re-packetize even without any license installed).
> I was thinking that it was not optimal, but Anthony explained some minutes
> ago on #freeswitch why this was the preferred behaviour, despite the quality
> loss it causes.
> David Ponzone ? Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client? IP eva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 04/10/2010 ? 17:20, Jan Riedinger a ?crit :
>
> ?Using http://files.freeswitch.org/g729/fsg729-158-installer really did
> fix the issue described below.
>
> However, it seems that the licence is used (up), even if there is no
> transcoding needed. In my CDR I find for all calls, for which G729 is
> involed at all,
>
> read_codec="G729", read_rate="8000", write_codec="G729", write_rate="8000"
>
> Nevertheless, all of these calls use a G729 licence. Is there any way to
> find out, why mod_com_g729 wants to do any kind of transcoding?
>
> BR
> ????Jan
>
> now the cod
>
> Am 04.10.2010 15:15, schrieb Brian West:
>
> http://files.freeswitch.org/g729/fsg729-158-installer
>
> Just the CDN doesn't have the index updated yet. ?That will fix your issue.
>
> /b
>
> On Oct 4, 2010, at 8:09 AM, David Ponzone wrote:
>
> Nope, you have to install them manually.
>
> Current is v158.
>
> You can download it from the same place than 153, but via direct URL only.
>
> It's still Beta I think.
>
> David Ponzone ?Direction Technique
>
> email: david.ponzone at ipeva.fr
>
> tel: ?????01 74 03 18 97
>
> gsm: ??06 66 98 76 34
>
> Service Client IPeva
>
> tel: ?????0811 46 26 26
>
> www.ipeva.fr ?- ??www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
> --
> Jan Riedinger ??????????????????????????Phone : ?+49-30-39 73 19 66
> Dipl.-Inf. | Managing Director ?????????Fax ??: ?+49-30-39 73 19 64
> ????????????????????????????????????????E-Mail: ?riedinger at sns.eu
> SNS Consult GmbH ???????????????????????ICQ ??: ?163-237-041
> S?dwestkorso 49a ???????????????????????MSN ??: ?jan at sns-consult.de
> 14197 Berlin GERMANY ???????????????????Skype : ?Jan Riedinger
>
> AG Charlottenburg - HRB 71973
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> --
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                         E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>
> AG Charlottenburg - HRB 71973
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> --
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                         E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>
> AG Charlottenburg - HRB 71973
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
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FreeSWITCH Developer Conference
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pstn:+19193869900


From msc at freeswitch.org  Mon Oct  4 10:40:18 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Oct 2010 10:40:18 -0700
Subject: [Freeswitch-users] Call back from voice mail creates loop
In-Reply-To: <4CA9ECBB.2000609@communicatefreely.net>
References: <4CA9ECBB.2000609@communicatefreely.net>
Message-ID: <AANLkTik24EaqTsfazwBPBZ4UG8eFPTZF-8XFMQ6xNFkb@mail.gmail.com>

Are you trying to bridge the current leg (user <--> voicemail) to another
endpoint? If so, how are you doing that? Are you transferring the leg back
into the dialplan for processing?

Pastebin all the relevant info, including a debug log of the call looping.

-MC

On Mon, Oct 4, 2010 at 8:03 AM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

> Hello,
>
> I'm trying to set up the option in voicemail so that users can return a
> call directly from VM.
>
> In my voicemail.conf.xml profile I have specified the dialplan as XML and
> the context as "internal",
> which is what I use for calls coming from our endpoints or authenticated
> users.
>
> In our internal dialplan, we do some pattern matching and call an
> appropriate lua dialplan script to
> do build a set of instructions.
>
> Problem is, the lua scripts are looking at the destination_number variable
> to decide where to send
> the call.  This variable is still set to "voicemail" from when the user
> called into their voicemail.
>  Shouldn't the voicemail app have set that to the number we want to call
> back?
>
> I used the info app to look at the variables, and I didn't see anything
> obvious.
>
> Is destination_number the correct variable to use in a lua script that
> generates a dialplan?
>
> -Tim
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Oct  4 10:42:04 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Oct 2010 10:42:04 -0700
Subject: [Freeswitch-users] FreeSWITCH HA
In-Reply-To: <AF824FE3F385E543ACA98ADFD670BA6A0184CE72@mlbe2k9.cs.myharris.net>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CE72@mlbe2k9.cs.myharris.net>
Message-ID: <AANLkTim3xT2V7BJUsWxiEGPVZxp_BNQgmaB1AUG=HRb1@mail.gmail.com>

I would think so. In fact, the endpoints probably wouldn't notice any blips
at all since media is flowing around FS.
-MC

On Mon, Oct 4, 2010 at 8:34 AM, Kane, Michael (mkane02)
<mkane02 at harris.com>wrote:

>  Will FreeSWITCH HA work if bypass_media = true?
>
> Thanks Mike
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mkane02 at harris.com  Mon Oct  4 11:00:21 2010
From: mkane02 at harris.com (Kane, Michael (mkane02))
Date: Mon, 4 Oct 2010 14:00:21 -0400
Subject: [Freeswitch-users] FreeSWITCH HA
In-Reply-To: <AANLkTim3xT2V7BJUsWxiEGPVZxp_BNQgmaB1AUG=HRb1@mail.gmail.com>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CE72@mlbe2k9.cs.myharris.net>
	<AANLkTim3xT2V7BJUsWxiEGPVZxp_BNQgmaB1AUG=HRb1@mail.gmail.com>
Message-ID: <AF824FE3F385E543ACA98ADFD670BA6A0184CE73@mlbe2k9.cs.myharris.net>

Thanks.  Is there alternative documentation aside from
http://wiki.freeswitch.org/wiki/Freeswitch_HA?  

 

Thanks Mike

 

 

 

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Monday, October 04, 2010 1:42 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FreeSWITCH HA

 

I would think so. In fact, the endpoints probably wouldn't notice any
blips at all since media is flowing around FS.
-MC

On Mon, Oct 4, 2010 at 8:34 AM, Kane, Michael (mkane02)
<mkane02 at harris.com> wrote:

Will FreeSWITCH HA work if bypass_media = true?

Thanks Mike


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

 

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From mario_fs at mgtech.com  Mon Oct  4 11:07:48 2010
From: mario_fs at mgtech.com (Mario)
Date: Mon, 04 Oct 2010 11:07:48 -0700
Subject: [Freeswitch-users] voicemail.tpl vs notify-voicemail.tpl
Message-ID: <4CAA17F4.5060402@mgtech.com>

I now have email working but could not find info about the differences
between voicemail.tpl and notify-voicemail.tpl. They look the same and I
wonder when each are used. I see they are referenced in voicemail.xml.
Thanks.
Mario


From bottleman at icf.org.ru  Mon Oct  4 11:11:56 2010
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Mon, 4 Oct 2010 22:11:56 +0400 (MSD)
Subject: [Freeswitch-users] Codec question: about the G729 codec
In-Reply-To: <1285771120946-5584195.post@n2.nabble.com>
References: <AANLkTimujf_CFT3HXTrmwzeGNYRoWF4aAGrzF+5bSap5@mail.gmail.com>
	<B9E6617B-9A5B-4B4E-8F4B-084D1F134CAA@freeswitch.org>
	<1285337934295-5567000.post@n2.nabble.com>
	<8F2CD116-937E-4D06-AEA4-869130559D9F@freeswitch.org>
	<alpine.DEB.2.00.1009271230500.22492@icf.org.ru>
	<1285771120946-5584195.post@n2.nabble.com>
Message-ID: <alpine.DEB.2.00.1010042210200.11501@icf.org.ru>

On 2010-09-29 07:38 -0700, mazilo wrote freeswitch-users at lists.freeswitch.org:

m>
m>
m>Georgiewskiy Yuriy wrote:
m>> 
m>> On 2010-09-24 09:22 -0500, Brian West wrote FreeSWITCH Users Help:
m>> 
m>> BW>It could be but it won't include the patent indemnity, isn't optimized
m>> and doesn't support the project.  ;)
m>> BW>
m>> BW>/b
m>> BW>
m>> BW>On Sep 24, 2010, at 9:18 AM, mazilo wrote:
m>> BW>
m>> BW>> How about the  http://www.itu.int/rec/T-REC-G.729-200701-I/en ITU
m>> G729 CoDec
m>> BW>> source ? Can this be ported to FS?
m>> 
m>> It's already ported, see http://github.com/Deepwalker/fs_itu_g729
m>Cool and thanks for the link. I believe this will be available for the next
m>releases of FS.

I think no, it violates patents in some countryes ad cannot be included in fs trie.

m>-----
m>don't and stop are the ONLY two 4-letter words considered offensive to men,
m>but not when used together.

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From msc at freeswitch.org  Mon Oct  4 11:32:41 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Oct 2010 11:32:41 -0700
Subject: [Freeswitch-users] FreeSWITCH HA
In-Reply-To: <AF824FE3F385E543ACA98ADFD670BA6A0184CE73@mlbe2k9.cs.myharris.net>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CE72@mlbe2k9.cs.myharris.net>
	<AANLkTim3xT2V7BJUsWxiEGPVZxp_BNQgmaB1AUG=HRb1@mail.gmail.com>
	<AF824FE3F385E543ACA98ADFD670BA6A0184CE73@mlbe2k9.cs.myharris.net>
Message-ID: <AANLkTi=qeO4ydgrLTnBmQpuvZ0quTKVWtf70tLwMaJ3K@mail.gmail.com>

That's pretty much it. If anyone out there has used HA and can add to the
docs it would be most appreciated.
-MC

On Mon, Oct 4, 2010 at 11:00 AM, Kane, Michael (mkane02) <mkane02 at harris.com
> wrote:

>   Thanks.  Is there alternative documentation aside from
> http://wiki.freeswitch.org/wiki/Freeswitch_HA?
>
>
>
> Thanks Mike
>
>
>
>
>
>
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, October 04, 2010 1:42 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] FreeSWITCH HA
>
>
>
> I would think so. In fact, the endpoints probably wouldn't notice any blips
> at all since media is flowing around FS.
> -MC
>
> On Mon, Oct 4, 2010 at 8:34 AM, Kane, Michael (mkane02) <
> mkane02 at harris.com> wrote:
>
> Will FreeSWITCH HA work if bypass_media = true?
>
> Thanks Mike
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Oct  4 11:39:48 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Oct 2010 11:39:48 -0700
Subject: [Freeswitch-users] Is anyone familiar with Telemaque?
Message-ID: <AANLkTimib+3T7L9sP28HN6BREJF5ca2j2nou-84YU=Ln@mail.gmail.com>

Dear FreeSWITCH Community,

Some of you may have stumbled upon this news item:

http://blogs.oracle.com/mysql/2010/10/innovating_with_open_source_call_center_services.html

Evidently there is a company in France (Telemaque) who is using MySQL +
Kamailio + FreeSWITCH for some heavy duty call processing. If you are at all
familiar with this company please let me know. I'd like to learn more about
what they are doing. (Je ne parle pas boucoup de Francais :( )

Thanks for your help,
MC
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From mkane02 at harris.com  Mon Oct  4 11:45:06 2010
From: mkane02 at harris.com (Kane, Michael (mkane02))
Date: Mon, 4 Oct 2010 14:45:06 -0400
Subject: [Freeswitch-users] FreeSWITCH HA
In-Reply-To: <AANLkTi=qeO4ydgrLTnBmQpuvZ0quTKVWtf70tLwMaJ3K@mail.gmail.com>
References: <AF824FE3F385E543ACA98ADFD670BA6A0184CE72@mlbe2k9.cs.myharris.net><AANLkTim3xT2V7BJUsWxiEGPVZxp_BNQgmaB1AUG=HRb1@mail.gmail.com><AF824FE3F385E543ACA98ADFD670BA6A0184CE73@mlbe2k9.cs.myharris.net>
	<AANLkTi=qeO4ydgrLTnBmQpuvZ0quTKVWtf70tLwMaJ3K@mail.gmail.com>
Message-ID: <AF824FE3F385E543ACA98ADFD670BA6A0184CE74@mlbe2k9.cs.myharris.net>

Michael, much appreciated.  I read earlier Henry H. has a pretty good
understanding of HA and is willing to document.  I'll do the same as I
maneuver through the implementation.  Again, thanks.

 

Mike

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Monday, October 04, 2010 2:33 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FreeSWITCH HA

 

That's pretty much it. If anyone out there has used HA and can add to
the docs it would be most appreciated.
-MC

On Mon, Oct 4, 2010 at 11:00 AM, Kane, Michael (mkane02)
<mkane02 at harris.com> wrote:

Thanks.  Is there alternative documentation aside from
http://wiki.freeswitch.org/wiki/Freeswitch_HA?  

 

Thanks Mike

 

 

 

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Monday, October 04, 2010 1:42 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FreeSWITCH HA

 

I would think so. In fact, the endpoints probably wouldn't notice any
blips at all since media is flowing around FS.
-MC

On Mon, Oct 4, 2010 at 8:34 AM, Kane, Michael (mkane02)
<mkane02 at harris.com> wrote:

Will FreeSWITCH HA work if bypass_media = true?

Thanks Mike


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

 


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

 

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From peder at networkoblivion.com  Mon Oct  4 11:58:31 2010
From: peder at networkoblivion.com (Peder)
Date: Mon, 4 Oct 2010 13:58:31 -0500
Subject: [Freeswitch-users] voicemail.tpl vs notify-voicemail.tpl
In-Reply-To: <4CAA17F4.5060402@mgtech.com>
References: <4CAA17F4.5060402@mgtech.com>
Message-ID: <080201cb63f6$236928b0$6a3b7a10$@com>

There are two types of notifications.  Voicemail.tpl is intended for you to
receive an email with the message as an attachment.  Notify is typically
used to send a message to a pager (if anybody still has one) or maybe as a
text message that just says you have an email, without an actual attachment
of the message.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mario
Sent: Monday, October 04, 2010 1:08 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] voicemail.tpl vs notify-voicemail.tpl

I now have email working but could not find info about the differences
between voicemail.tpl and notify-voicemail.tpl. They look the same and I
wonder when each are used. I see they are referenced in voicemail.xml.
Thanks.
Mario

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



From david.ponzone at ipeva.fr  Mon Oct  4 12:07:57 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 4 Oct 2010 21:07:57 +0200
Subject: [Freeswitch-users] mod_com_g729 licences used even without
	transcoding
In-Reply-To: <AANLkTim531sqEheNx-Mq4q93rwtqtVjfAK4qgk83uao3@mail.gmail.com>
References: <4CA9B34C.5070400@sns.eu>
	<1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>
	<4CA9CB28.8010101@sns.eu>
	<D76D3E6E-7576-4299-BF16-9A7B29BFC37F@ipeva.fr>
	<7CAB482F-5FFA-491F-8BF6-EAE4E376C6F1@freeswitch.org>
	<4CA9F0C6.8060005@sns.eu>
	<FFF3584F-4412-4EEE-AA98-0D266DE5DE0C@ipeva.fr>
	<4CA9FC99.5060202@sns.eu>
	<D3248AA8-EB9E-40A2-B49E-4E8391B4F09C@ipeva.fr>
	<4CAA0C55.2040109@sns.eu>
	<AANLkTim531sqEheNx-Mq4q93rwtqtVjfAK4qgk83uao3@mail.gmail.com>
Message-ID: <D9CE33A7-4F2A-464B-BEC3-7863192E0F9B@ipeva.fr>

Tony,

I'd rather have a consistent ptime between both legs, but I just  
noticed that the new carrier I use does not honor my request to use a  
ptime else than 20ms.
They just send back 20ms.
I will solve that with them sooner or later, but I suppose I am not  
the only one in this situation.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 04/10/2010 ? 19:34, Anthony Minessale a ?crit :

> because it's based on G.711 the original codec used.
>
> It's a balance of latency to packet len.
>
> G.729 is poorly packetized at 20ms with only 10 bytes per payload.
>
> We can do something to make it work the way you want, but then I
> refuse to care if you get jittery calls because you are sending the
> wrong ptime in both directions.
>
>
>
> On Mon, Oct 4, 2010 at 12:18 PM, Jan Riedinger <riedinger at sns.eu>  
> wrote:
>> I'm trying to do wholesale business with some dozen of  
>> interconnects. It's
>> hopeless to force all customers/clients to use the same ptime - and  
>> I think
>> there is no real reason to force them. I'm already happy, if they  
>> all offer
>> the usage of g729 ;-). Often they pass through the calls themselves.
>>
>> BTW - I never understood, why 20 ms is the default frame size. In  
>> this way
>> you have about 20 bytes payload and 40 bytes overhead for the TCP/ 
>> IP header,
>> which is a pretty bad ratio.
>>
>> BR
>>     Jan
>>
>> Am 04.10.2010 18:56, schrieb David Ponzone:
>>
>> You could also check with your carrier why they dont accept the  
>> custom ptime
>> you need, or normalize your customers' config.
>> I have a nice extension I made that denies calls if ptime is not  
>> 20ms, with
>> early-neg or late-neg:
>>   <extension name="check-ptime" continue="true">
>>     <condition field="${sip_use_codec_ptime}" expression="20"
>> break="on-true"/>
>>     <condition field="${ep_codec_string}" expression="@20i">
>>       <anti-action application="respond" data="488"/>
>>       <anti-action application="hangup" data="88"/>
>>     </condition>
>>   </extension>
>> David Ponzone   Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client  IP eva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> Le 04/10/2010 ? 18:11, Jan Riedinger a ?crit :
>>
>> I think, it would be a could idea to make it configurable, if  
>> mod_com_g729
>> shall decode/encode the stream to re-packetize. With the current  
>> behaviour
>> its usage is  too expensive for me in respect of money needed for  
>> licences
>> and CPU power. Most of my calls are calls are g729 to g729,  
>> transcodeding is
>> needed only very rare.
>>
>>
>>
>> Am 04.10.2010 17:46, schrieb David Ponzone:
>>
>> I just discovered a situation like that.
>> If the ptime of leg A and leg B does not match, FS will decode/ 
>> encode the
>> stream in order to re-packetize (despite the fact that mod_com_g729  
>> is
>> capable to re-packetize even without any license installed).
>> I was thinking that it was not optimal, but Anthony explained some  
>> minutes
>> ago on #freeswitch why this was the preferred behaviour, despite  
>> the quality
>> loss it causes.
>> David Ponzone   Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client  IP eva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> Le 04/10/2010 ? 17:20, Jan Riedinger a ?crit :
>>
>>  Using http://files.freeswitch.org/g729/fsg729-158-installer really  
>> did
>> fix the issue described below.
>>
>> However, it seems that the licence is used (up), even if there is no
>> transcoding needed. In my CDR I find for all calls, for which G729 is
>> involed at all,
>>
>> read_codec="G729", read_rate="8000", write_codec="G729",  
>> write_rate="8000"
>>
>> Nevertheless, all of these calls use a G729 licence. Is there any  
>> way to
>> find out, why mod_com_g729 wants to do any kind of transcoding?
>>
>> BR
>>     Jan
>>
>> now the cod
>>
>> Am 04.10.2010 15:15, schrieb Brian West:
>>
>> http://files.freeswitch.org/g729/fsg729-158-installer
>>
>> Just the CDN doesn't have the index updated yet.  That will fix  
>> your issue.
>>
>> /b
>>
>> On Oct 4, 2010, at 8:09 AM, David Ponzone wrote:
>>
>> Nope, you have to install them manually.
>>
>> Current is v158.
>>
>> You can download it from the same place than 153, but via direct  
>> URL only.
>>
>> It's still Beta I think.
>>
>> David Ponzone  Direction Technique
>>
>> email: david.ponzone at ipeva.fr
>>
>> tel:      01 74 03 18 97
>>
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>>
>> tel:      0811 46 26 26
>>
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> _______________________________________________
>>
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> http://www.freeswitch.org
>>
>> --
>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>                                         E-Mail:  riedinger at sns.eu
>> SNS Consult GmbH                        ICQ   :  163-237-041
>> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>
>> AG Charlottenburg - HRB 71973
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> --
>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>                                        E-Mail:  riedinger at sns.eu
>> SNS Consult GmbH                        ICQ   :  163-237-041
>> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>
>> AG Charlottenburg - HRB 71973
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> --
>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>                                        E-Mail:  riedinger at sns.eu
>> SNS Consult GmbH                        ICQ   :  163-237-041
>> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>
>> AG Charlottenburg - HRB 71973
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From fs-list at communicatefreely.net  Mon Oct  4 12:55:42 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Mon, 04 Oct 2010 15:55:42 -0400
Subject: [Freeswitch-users] Call back from voice mail creates loop
In-Reply-To: <AANLkTik24EaqTsfazwBPBZ4UG8eFPTZF-8XFMQ6xNFkb@mail.gmail.com>
References: <4CA9ECBB.2000609@communicatefreely.net>
	<AANLkTik24EaqTsfazwBPBZ4UG8eFPTZF-8XFMQ6xNFkb@mail.gmail.com>
Message-ID: <4CAA313E.3070008@communicatefreely.net>

Michael Collins wrote:
> Are you trying to bridge the current leg (user <--> voicemail) to
> another endpoint? If so, how are you doing that? Are you transferring
> the leg back into the dialplan for processing?
> 
I may have just answered my own question - quite by accident while working on another problem.

I was sending the call back to the dial plan for processing, but I was using the execute_extension
application instead of transfer

It looks like execute_extension doesn't affect the destination_number variable, whereas transfer
changes the destination number, and sets the previously dialed number as RDNIS.

Changing my logic to use transfer instead of execute_extension seems to have solved things.

-Tim


From msc at freeswitch.org  Mon Oct  4 13:55:25 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Oct 2010 13:55:25 -0700
Subject: [Freeswitch-users] Call back from voice mail creates loop
In-Reply-To: <4CAA313E.3070008@communicatefreely.net>
References: <4CA9ECBB.2000609@communicatefreely.net>
	<AANLkTik24EaqTsfazwBPBZ4UG8eFPTZF-8XFMQ6xNFkb@mail.gmail.com>
	<4CAA313E.3070008@communicatefreely.net>
Message-ID: <AANLkTimg5iRzRSYKpY4uT3O9R+-ifYU09jCCyHDYpmq-@mail.gmail.com>

Yes, transfer is your friend in this scenario. :)
-MC

On Mon, Oct 4, 2010 at 12:55 PM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

> Michael Collins wrote:
> > Are you trying to bridge the current leg (user <--> voicemail) to
> > another endpoint? If so, how are you doing that? Are you transferring
> > the leg back into the dialplan for processing?
> >
> I may have just answered my own question - quite by accident while working
> on another problem.
>
> I was sending the call back to the dial plan for processing, but I was
> using the execute_extension
> application instead of transfer
>
> It looks like execute_extension doesn't affect the destination_number
> variable, whereas transfer
> changes the destination number, and sets the previously dialed number as
> RDNIS.
>
> Changing my logic to use transfer instead of execute_extension seems to
> have solved things.
>
> -Tim
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mario_fs at mgtech.com  Mon Oct  4 14:06:11 2010
From: mario_fs at mgtech.com (Mario)
Date: Mon, 04 Oct 2010 14:06:11 -0700
Subject: [Freeswitch-users] voicemail.tpl vs notify-voicemail.tpl
In-Reply-To: <080201cb63f6$236928b0$6a3b7a10$@com>
References: <4CAA17F4.5060402@mgtech.com> <080201cb63f6$236928b0$6a3b7a10$@com>
Message-ID: <4CAA41C3.6070504@mgtech.com>

Thank you! Email works great!

On 10/04/2010 11:58 AM, Peder wrote:
> There are two types of notifications.  Voicemail.tpl is intended for you to
> receive an email with the message as an attachment.  Notify is typically
> used to send a message to a pager (if anybody still has one) or maybe as a
> text message that just says you have an email, without an actual attachment
> of the message.
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mario
> Sent: Monday, October 04, 2010 1:08 PM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] voicemail.tpl vs notify-voicemail.tpl
> 
> I now have email working but could not find info about the differences
> between voicemail.tpl and notify-voicemail.tpl. They look the same and I
> wonder when each are used. I see they are referenced in voicemail.xml.
> Thanks.
> Mario
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
*Mario*


From riedinger at sns.eu  Mon Oct  4 14:08:02 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Mon, 04 Oct 2010 23:08:02 +0200
Subject: [Freeswitch-users] mod_com_g729 licences used even
	without	transcoding
In-Reply-To: <D9CE33A7-4F2A-464B-BEC3-7863192E0F9B@ipeva.fr>
References: <4CA9B34C.5070400@sns.eu>	<1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>	<4CA9CB28.8010101@sns.eu>	<D76D3E6E-7576-4299-BF16-9A7B29BFC37F@ipeva.fr>	<7CAB482F-5FFA-491F-8BF6-EAE4E376C6F1@freeswitch.org>	<4CA9F0C6.8060005@sns.eu>	<FFF3584F-4412-4EEE-AA98-0D266DE5DE0C@ipeva.fr>	<4CA9FC99.5060202@sns.eu>	<D3248AA8-EB9E-40A2-B49E-4E8391B4F09C@ipeva.fr>	<4CAA0C55.2040109@sns.eu>	<AANLkTim531sqEheNx-Mq4q93rwtqtVjfAK4qgk83uao3@mail.gmail.com>
	<D9CE33A7-4F2A-464B-BEC3-7863192E0F9B@ipeva.fr>
Message-ID: <4CAA4232.4000300@sns.eu>

  I'm doing voip configuration for about 8 years. Because of the 
overhead of the tcp/ip header I usually configured my Ciscos to use g729 
with a frame size of 40 or 60 bytes. This setting is pretty often 
ignored by the other side. Thus it happens often that both sides are 
sending with different frame sizes. However, I remember only one case, 
where this caused any problems.

Thus my preferred behaviour for mod_com_g729 would be, that it doesn't 
re-packetize and pass the packet in pass through mode for g729 to g729 
calls, until it isn't configured explicitly in another manner. A licence 
should be only used and transcoding should be only done, if caller and 
callee are using different codecs.

It seems that there are  three variants to handle g729 to g729 calls 
with different ptime on the side of caller and callee:
1.
Encode and decode packets with the correct ptime.

Advantages:
Exclusion of jitter problems
Assurance that the correct ptime is used on the side of caller and callee.

Disadvantages:
High expenses for licences and CPU load

2.
Re-packetizing without de- and encoding

Advantages:
Low  CPU load, no licences are needed

DisadvantagesI
According Anthony there could be a jitter problem, because a wrong ptime 
is used to both sides.

3.
Pass-through the packages untouched

Advantages:
Lowest CPU usage, no licences are needed

Disadvantages:
In some - according my experience very very rare cases -  there could be 
an incompatibility between caller and callee because of the usage of a 
different ptime.

I assume that mode_g729 is working with variant 3, thus it can't be such 
a bad option. As I explained above, I would like to go with this variant.

Of course, the very best solution would be, to make the behaviour 
configurable per profile or even better per call. But I don't know if 
this can be done with a reasonable effort.

BR
    Jan




Am 04.10.2010 21:07, schrieb David Ponzone:
> Tony,
>
> I'd rather have a consistent ptime between both legs, but I just 
> noticed that the new carrier I use does not honor my request to use a 
> ptime else than 20ms.
> They just send back 20ms.
> I will solve that with them sooner or later, but I suppose I am not 
> the only one in this situation.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IP eva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 04/10/2010 ? 19:34, Anthony Minessale a ?crit :
>
>> because it's based on G.711 the original codec used.
>>
>> It's a balance of latency to packet len.
>>
>> G.729 is poorly packetized at 20ms with only 10 bytes per payload.
>>
>> We can do something to make it work the way you want, but then I
>> refuse to care if you get jittery calls because you are sending the
>> wrong ptime in both directions.
>>
>>
>>
>> On Mon, Oct 4, 2010 at 12:18 PM, Jan Riedinger <riedinger at sns.eu 
>> <mailto:riedinger at sns.eu>> wrote:
>>> I'm trying to do wholesale business with some dozen of 
>>> interconnects. It's
>>> hopeless to force all customers/clients to use the same ptime - and 
>>> I think
>>> there is no real reason to force them. I'm already happy, if they 
>>> all offer
>>> the usage of g729 ;-). Often they pass through the calls themselves.
>>>
>>> BTW - I never understood, why 20 ms is the default frame size. In 
>>> this way
>>> you have about 20 bytes payload and 40 bytes overhead for the TCP/IP 
>>> header,
>>> which is a pretty bad ratio.
>>>
>>> BR
>>>     Jan
>>>
>>> Am 04.10.2010 18:56, schrieb David Ponzone:
>>>
>>> You could also check with your carrier why they dont accept the 
>>> custom ptime
>>> you need, or normalize your customers' config.
>>> I have a nice extension I made that denies calls if ptime is not 
>>> 20ms, with
>>> early-neg or late-neg:
>>> <extension name="check-ptime" continue="true">
>>> <condition field="${sip_use_codec_ptime}" expression="20"
>>> break="on-true"/>
>>> <condition field="${ep_codec_string}" expression="@20i">
>>> <anti-action application="respond" data="488"/>
>>> <anti-action application="hangup" data="88"/>
>>> </condition>
>>> </extension>
>>> David Ponzone   Direction Technique
>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>> Service Client  IP eva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  - www.ipeva-studio.com <http://www.ipeva-studio.com>
>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou 
>>> diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce 
>>> message s'il
>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>
>>>
>>>
>>> Le 04/10/2010 ? 18:11, Jan Riedinger a ?crit :
>>>
>>> I think, it would be a could idea to make it configurable, if 
>>> mod_com_g729
>>> shall decode/encode the stream to re-packetize. With the current 
>>> behaviour
>>> its usage is  too expensive for me in respect of money needed for 
>>> licences
>>> and CPU power. Most of my calls are calls are g729 to g729, 
>>> transcodeding is
>>> needed only very rare.
>>>
>>>
>>>
>>> Am 04.10.2010 17:46, schrieb David Ponzone:
>>>
>>> I just discovered a situation like that.
>>> If the ptime of leg A and leg B does not match, FS will 
>>> decode/encode the
>>> stream in order to re-packetize (despite the fact that mod_com_g729 is
>>> capable to re-packetize even without any license installed).
>>> I was thinking that it was not optimal, but Anthony explained some 
>>> minutes
>>> ago on #freeswitch why this was the preferred behaviour, despite the 
>>> quality
>>> loss it causes.
>>> David Ponzone   Direction Technique
>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>> Service Client  IP eva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  - www.ipeva-studio.com <http://www.ipeva-studio.com>
>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou 
>>> diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce 
>>> message s'il
>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>
>>>
>>>
>>> Le 04/10/2010 ? 17:20, Jan Riedinger a ?crit :
>>>
>>>  Using http://files.freeswitch.org/g729/fsg729-158-installer really did
>>> fix the issue described below.
>>>
>>> However, it seems that the licence is used (up), even if there is no
>>> transcoding needed. In my CDR I find for all calls, for which G729 is
>>> involed at all,
>>>
>>> read_codec="G729", read_rate="8000", write_codec="G729", 
>>> write_rate="8000"
>>>
>>> Nevertheless, all of these calls use a G729 licence. Is there any way to
>>> find out, why mod_com_g729 wants to do any kind of transcoding?
>>>
>>> BR
>>>     Jan
>>>
>>> now the cod
>>>
>>> Am 04.10.2010 15:15, schrieb Brian West:
>>>
>>> http://files.freeswitch.org/g729/fsg729-158-installer
>>>
>>> Just the CDN doesn't have the index updated yet.  That will fix your 
>>> issue.
>>>
>>> /b
>>>
>>> On Oct 4, 2010, at 8:09 AM, David Ponzone wrote:
>>>
>>> Nope, you have to install them manually.
>>>
>>> Current is v158.
>>>
>>> You can download it from the same place than 153, but via direct URL 
>>> only.
>>>
>>> It's still Beta I think.
>>>
>>> David Ponzone  Direction Technique
>>>
>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>
>>> tel:      01 74 03 18 97
>>>
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>>
>>> tel:      0811 46 26 26
>>>
>>> www.ipeva.fr  - www.ipeva-studio.com <http://www.ipeva-studio.com>
>>>
>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou 
>>> diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce 
>>> message s'il
>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>
>>>
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org 
>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>> --
>>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>>                                         E-Mail: riedinger at sns.eu 
>>> <mailto:riedinger at sns.eu>
>>> SNS Consult GmbH                        ICQ   :  163-237-041
>>> S?dwestkorso 49a                        MSN   : jan at sns-consult.de 
>>> <mailto:jan at sns-consult.de>
>>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>>
>>> AG Charlottenburg - HRB 71973
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org 
>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org 
>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> --
>>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>>                                        E-Mail: riedinger at sns.eu 
>>> <mailto:riedinger at sns.eu>
>>> SNS Consult GmbH                        ICQ   :  163-237-041
>>> S?dwestkorso 49a                        MSN   : jan at sns-consult.de 
>>> <mailto:jan at sns-consult.de>
>>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>>
>>> AG Charlottenburg - HRB 71973
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org 
>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> --
>>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>>                                        E-Mail: riedinger at sns.eu 
>>> <mailto:riedinger at sns.eu>
>>> SNS Consult GmbH                        ICQ   :  163-237-041
>>> S?dwestkorso 49a                        MSN   : jan at sns-consult.de 
>>> <mailto:jan at sns-consult.de>
>>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>>
>>> AG Charlottenburg - HRB 71973
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org 
>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                         E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973

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From david.ponzone at ipeva.fr  Mon Oct  4 14:29:55 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 4 Oct 2010 23:29:55 +0200
Subject: [Freeswitch-users] mod_com_g729 licences used even without
	transcoding
In-Reply-To: <AANLkTim531sqEheNx-Mq4q93rwtqtVjfAK4qgk83uao3@mail.gmail.com>
References: <4CA9B34C.5070400@sns.eu>
	<1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>
	<4CA9CB28.8010101@sns.eu>
	<D76D3E6E-7576-4299-BF16-9A7B29BFC37F@ipeva.fr>
	<7CAB482F-5FFA-491F-8BF6-EAE4E376C6F1@freeswitch.org>
	<4CA9F0C6.8060005@sns.eu>
	<FFF3584F-4412-4EEE-AA98-0D266DE5DE0C@ipeva.fr>
	<4CA9FC99.5060202@sns.eu>
	<D3248AA8-EB9E-40A2-B49E-4E8391B4F09C@ipeva.fr>
	<4CAA0C55.2040109@sns.eu>
	<AANLkTim531sqEheNx-Mq4q93rwtqtVjfAK4qgk83uao3@mail.gmail.com>
Message-ID: <C00AD253-44AB-45E3-B3A0-51D1B5EE409D@ipeva.fr>

Tony,

Shouldn't that 10 bytes be a 20 ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 04/10/2010 ? 19:34, Anthony Minessale a ?crit :

> because it's based on G.711 the original codec used.
>
> It's a balance of latency to packet len.
>
> G.729 is poorly packetized at 20ms with only 10 bytes per payload.
>
> We can do something to make it work the way you want, but then I
> refuse to care if you get jittery calls because you are sending the
> wrong ptime in both directions.
>
>
>
> On Mon, Oct 4, 2010 at 12:18 PM, Jan Riedinger <riedinger at sns.eu>  
> wrote:
>> I'm trying to do wholesale business with some dozen of  
>> interconnects. It's
>> hopeless to force all customers/clients to use the same ptime - and  
>> I think
>> there is no real reason to force them. I'm already happy, if they  
>> all offer
>> the usage of g729 ;-). Often they pass through the calls themselves.
>>
>> BTW - I never understood, why 20 ms is the default frame size. In  
>> this way
>> you have about 20 bytes payload and 40 bytes overhead for the TCP/ 
>> IP header,
>> which is a pretty bad ratio.
>>
>> BR
>>     Jan
>>
>> Am 04.10.2010 18:56, schrieb David Ponzone:
>>
>> You could also check with your carrier why they dont accept the  
>> custom ptime
>> you need, or normalize your customers' config.
>> I have a nice extension I made that denies calls if ptime is not  
>> 20ms, with
>> early-neg or late-neg:
>>   <extension name="check-ptime" continue="true">
>>     <condition field="${sip_use_codec_ptime}" expression="20"
>> break="on-true"/>
>>     <condition field="${ep_codec_string}" expression="@20i">
>>       <anti-action application="respond" data="488"/>
>>       <anti-action application="hangup" data="88"/>
>>     </condition>
>>   </extension>
>> David Ponzone   Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client  IP eva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> Le 04/10/2010 ? 18:11, Jan Riedinger a ?crit :
>>
>> I think, it would be a could idea to make it configurable, if  
>> mod_com_g729
>> shall decode/encode the stream to re-packetize. With the current  
>> behaviour
>> its usage is  too expensive for me in respect of money needed for  
>> licences
>> and CPU power. Most of my calls are calls are g729 to g729,  
>> transcodeding is
>> needed only very rare.
>>
>>
>>
>> Am 04.10.2010 17:46, schrieb David Ponzone:
>>
>> I just discovered a situation like that.
>> If the ptime of leg A and leg B does not match, FS will decode/ 
>> encode the
>> stream in order to re-packetize (despite the fact that mod_com_g729  
>> is
>> capable to re-packetize even without any license installed).
>> I was thinking that it was not optimal, but Anthony explained some  
>> minutes
>> ago on #freeswitch why this was the preferred behaviour, despite  
>> the quality
>> loss it causes.
>> David Ponzone   Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client  IP eva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> Le 04/10/2010 ? 17:20, Jan Riedinger a ?crit :
>>
>>  Using http://files.freeswitch.org/g729/fsg729-158-installer really  
>> did
>> fix the issue described below.
>>
>> However, it seems that the licence is used (up), even if there is no
>> transcoding needed. In my CDR I find for all calls, for which G729 is
>> involed at all,
>>
>> read_codec="G729", read_rate="8000", write_codec="G729",  
>> write_rate="8000"
>>
>> Nevertheless, all of these calls use a G729 licence. Is there any  
>> way to
>> find out, why mod_com_g729 wants to do any kind of transcoding?
>>
>> BR
>>     Jan
>>
>> now the cod
>>
>> Am 04.10.2010 15:15, schrieb Brian West:
>>
>> http://files.freeswitch.org/g729/fsg729-158-installer
>>
>> Just the CDN doesn't have the index updated yet.  That will fix  
>> your issue.
>>
>> /b
>>
>> On Oct 4, 2010, at 8:09 AM, David Ponzone wrote:
>>
>> Nope, you have to install them manually.
>>
>> Current is v158.
>>
>> You can download it from the same place than 153, but via direct  
>> URL only.
>>
>> It's still Beta I think.
>>
>> David Ponzone  Direction Technique
>>
>> email: david.ponzone at ipeva.fr
>>
>> tel:      01 74 03 18 97
>>
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>>
>> tel:      0811 46 26 26
>>
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> _______________________________________________
>>
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> http://www.freeswitch.org
>>
>> --
>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>                                         E-Mail:  riedinger at sns.eu
>> SNS Consult GmbH                        ICQ   :  163-237-041
>> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>
>> AG Charlottenburg - HRB 71973
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> --
>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>                                        E-Mail:  riedinger at sns.eu
>> SNS Consult GmbH                        ICQ   :  163-237-041
>> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>
>> AG Charlottenburg - HRB 71973
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> --
>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>                                        E-Mail:  riedinger at sns.eu
>> SNS Consult GmbH                        ICQ   :  163-237-041
>> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>
>> AG Charlottenburg - HRB 71973
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Mon Oct  4 14:43:48 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 Oct 2010 16:43:48 -0500
Subject: [Freeswitch-users] mod_com_g729 licences used even without
	transcoding
In-Reply-To: <C00AD253-44AB-45E3-B3A0-51D1B5EE409D@ipeva.fr>
References: <4CA9B34C.5070400@sns.eu>
	<1C707F9F-261F-4CF8-B291-66CC3C614BF8@ipeva.fr>
	<4CA9CB28.8010101@sns.eu>
	<D76D3E6E-7576-4299-BF16-9A7B29BFC37F@ipeva.fr>
	<7CAB482F-5FFA-491F-8BF6-EAE4E376C6F1@freeswitch.org>
	<4CA9F0C6.8060005@sns.eu>
	<FFF3584F-4412-4EEE-AA98-0D266DE5DE0C@ipeva.fr>
	<4CA9FC99.5060202@sns.eu>
	<D3248AA8-EB9E-40A2-B49E-4E8391B4F09C@ipeva.fr>
	<4CAA0C55.2040109@sns.eu>
	<AANLkTim531sqEheNx-Mq4q93rwtqtVjfAK4qgk83uao3@mail.gmail.com>
	<C00AD253-44AB-45E3-B3A0-51D1B5EE409D@ipeva.fr>
Message-ID: <AANLkTimZA=YugOYY7HyWKDJ+spfGObdxtPayEQsfknk_@mail.gmail.com>

yes

On Mon, Oct 4, 2010 at 4:29 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
> Tony,
> Shouldn't that 10 bytes be a 20 ?
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 04/10/2010 ? 19:34, Anthony Minessale a ?crit :
>
> because it's based on G.711 the original codec used.
>
> It's a balance of latency to packet len.
>
> G.729 is poorly packetized at 20ms with only 10 bytes per payload.
>
> We can do something to make it work the way you want, but then I
> refuse to care if you get jittery calls because you are sending the
> wrong ptime in both directions.
>
>
>
> On Mon, Oct 4, 2010 at 12:18 PM, Jan Riedinger <riedinger at sns.eu> wrote:
>
> I'm trying to do wholesale business with some dozen of interconnects. It's
>
> hopeless to force all customers/clients to use the same ptime - and I think
>
> there is no real reason to force them. I'm already happy, if they all offer
>
> the usage of g729 ;-). Often they pass through the calls themselves.
>
> BTW - I never understood, why 20 ms is the default frame size. In this way
>
> you have about 20 bytes payload and 40 bytes overhead for the TCP/IP header,
>
> which is a pretty bad ratio.
>
> BR
>
> ??? Jan
>
> Am 04.10.2010 18:56, schrieb David Ponzone:
>
> You could also check with your carrier why they dont accept the custom ptime
>
> you need, or normalize your customers' config.
>
> I have a nice extension I made that denies calls if ptime is not 20ms, with
>
> early-neg or late-neg:
>
> ??<extension name="check-ptime" continue="true">
>
> ?? ?<condition field="${sip_use_codec_ptime}" expression="20"
>
> break="on-true"/>
>
> ?? ?<condition field="${ep_codec_string}" expression="@20i">
>
> ?? ? ?<anti-action application="respond" data="488"/>
>
> ?? ? ?<anti-action application="hangup" data="88"/>
>
> ?? ?</condition>
>
> ??</extension>
>
> David Ponzone ? Direction Technique
>
> email: david.ponzone at ipeva.fr
>
> tel: ? ? ?01 74 03 18 97
>
> gsm: ? 06 66 98 76 34
>
> Service Client? IP eva
>
> tel: ? ? ?0811 46 26 26
>
> www.ipeva.fr? -? ?www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>
> non autoris?e est interdite. Tout message ?lectronique est susceptible
>
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 04/10/2010 ? 18:11, Jan Riedinger a ?crit :
>
> I think, it would be a could idea to make it configurable, if mod_com_g729
>
> shall decode/encode the stream to re-packetize. With the current behaviour
>
> its usage is? too expensive for me in respect of money needed for licences
>
> and CPU power. Most of my calls are calls are g729 to g729, transcodeding is
>
> needed only very rare.
>
>
>
> Am 04.10.2010 17:46, schrieb David Ponzone:
>
> I just discovered a situation like that.
>
> If the ptime of leg A and leg B does not match, FS will decode/encode the
>
> stream in order to re-packetize (despite the fact that mod_com_g729 is
>
> capable to re-packetize even without any license installed).
>
> I was thinking that it was not optimal, but Anthony explained some minutes
>
> ago on #freeswitch why this was the preferred behaviour, despite the quality
>
> loss it causes.
>
> David Ponzone ? Direction Technique
>
> email: david.ponzone at ipeva.fr
>
> tel: ? ? ?01 74 03 18 97
>
> gsm: ? 06 66 98 76 34
>
> Service Client? IP eva
>
> tel: ? ? ?0811 46 26 26
>
> www.ipeva.fr? -? ?www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>
> non autoris?e est interdite. Tout message ?lectronique est susceptible
>
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 04/10/2010 ? 17:20, Jan Riedinger a ?crit :
>
> ?Using http://files.freeswitch.org/g729/fsg729-158-installer really did
>
> fix the issue described below.
>
> However, it seems that the licence is used (up), even if there is no
>
> transcoding needed. In my CDR I find for all calls, for which G729 is
>
> involed at all,
>
> read_codec="G729", read_rate="8000", write_codec="G729", write_rate="8000"
>
> Nevertheless, all of these calls use a G729 licence. Is there any way to
>
> find out, why mod_com_g729 wants to do any kind of transcoding?
>
> BR
>
> ????Jan
>
> now the cod
>
> Am 04.10.2010 15:15, schrieb Brian West:
>
> http://files.freeswitch.org/g729/fsg729-158-installer
>
> Just the CDN doesn't have the index updated yet. ?That will fix your issue.
>
> /b
>
> On Oct 4, 2010, at 8:09 AM, David Ponzone wrote:
>
> Nope, you have to install them manually.
>
> Current is v158.
>
> You can download it from the same place than 153, but via direct URL only.
>
> It's still Beta I think.
>
> David Ponzone ?Direction Technique
>
> email: david.ponzone at ipeva.fr
>
> tel: ?????01 74 03 18 97
>
> gsm: ??06 66 98 76 34
>
> Service Client IPeva
>
> tel: ?????0811 46 26 26
>
> www.ipeva.fr ?- ??www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>
> non autoris?e est interdite. Tout message ?lectronique est susceptible
>
> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
> --
>
> Jan Riedinger ??????????????????????????Phone : ?+49-30-39 73 19 66
>
> Dipl.-Inf. | Managing Director ?????????Fax ??: ?+49-30-39 73 19 64
>
> ????????????????????????????????????????E-Mail: ?riedinger at sns.eu
>
> SNS Consult GmbH ???????????????????????ICQ ??: ?163-237-041
>
> S?dwestkorso 49a ???????????????????????MSN ??: ?jan at sns-consult.de
>
> 14197 Berlin GERMANY ???????????????????Skype : ?Jan Riedinger
>
> AG Charlottenburg - HRB 71973
>
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
> --
>
> Jan Riedinger ??????????????????????????Phone : ?+49-30-39 73 19 66
>
> Dipl.-Inf. | Managing Director ?????????Fax ??: ?+49-30-39 73 19 64
>
> ???????????????????????????????????????E-Mail: ?riedinger at sns.eu
>
> SNS Consult GmbH ???????????????????????ICQ ??: ?163-237-041
>
> S?dwestkorso 49a ???????????????????????MSN ??: ?jan at sns-consult.de
>
> 14197 Berlin GERMANY ???????????????????Skype : ?Jan Riedinger
>
> AG Charlottenburg - HRB 71973
>
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
> --
>
> Jan Riedinger ??????????????????????????Phone : ?+49-30-39 73 19 66
>
> Dipl.-Inf. | Managing Director ?????????Fax ??: ?+49-30-39 73 19 64
>
> ???????????????????????????????????????E-Mail: ?riedinger at sns.eu
>
> SNS Consult GmbH ???????????????????????ICQ ??: ?163-237-041
>
> S?dwestkorso 49a ???????????????????????MSN ??: ?jan at sns-consult.de
>
> 14197 Berlin GERMANY ???????????????????Skype : ?Jan Riedinger
>
> AG Charlottenburg - HRB 71973
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> 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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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
>
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> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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-- 
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


From shamun.toha at gmail.com  Mon Oct  4 15:24:42 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Tue, 5 Oct 2010 00:24:42 +0200
Subject: [Freeswitch-users] FreeSwitch - mod_opal,
	mod_h323 never get installed
In-Reply-To: <AANLkTi=gF1N13NMyGhmgmNw+mJTyhjO06g0o_Pr2Hg9_@mail.gmail.com>
References: <AANLkTi=9Hw6dGtzdaKwYR+oA2R1y5P0vfuHJqJW8Fncd@mail.gmail.com>
	<AANLkTi=wvXA+mYPdCjSL07Wk=psBSXrYh58b7gkP0LTO@mail.gmail.com>
	<AANLkTi=gF1N13NMyGhmgmNw+mJTyhjO06g0o_Pr2Hg9_@mail.gmail.com>
Message-ID: <AANLkTinLi8OL8+fOBOZE8FJ_wYSeHNCk3Su02-Q88umW@mail.gmail.com>

Hello,  Follow updated wiki but mod_opal is installed, mod_h323 didnt get
installed. Please guide for following failure. Thanks & Regards


[root at example freeswitch]# make mod_h323-install
 /bin/sh /usr/local/src/freeswitch/quiet_libtool   --mode=install
/usr/bin/install -c   libfreeswitch.la '/usr/local/freeswitch/lib'
quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.so.1.0.0
/usr/local/freeswitch/lib/libfreeswitch.so.1.0.0
quiet_libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f
libfreeswitch.so.1.0.0 libfreeswitch.so.1 || { rm -f libfreeswitch.so.1 &&
ln -s libfreeswitch.so.1.0.0 libfreeswitch.so.1; }; })
quiet_libtool: install: (cd /usr/local/freeswitch/lib && { ln -s -f
libfreeswitch.so.1.0.0 libfreeswitch.so || { rm -f libfreeswitch.so && ln -s
libfreeswitch.so.1.0.0 libfreeswitch.so; }; })
quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.lai
/usr/local/freeswitch/lib/libfreeswitch.la
quiet_libtool: install: /usr/bin/install -c .libs/libfreeswitch.a
/usr/local/freeswitch/lib/libfreeswitch.a
quiet_libtool: install: chmod 644 /usr/local/freeswitch/lib/libfreeswitch.a
quiet_libtool: install: ranlib /usr/local/freeswitch/lib/libfreeswitch.a
quiet_libtool: finish:
PATH="/usr/lib/qt-3.3/bin:/usr/kerberos/sbin:/usr/kerberos/bin:/usr/lib/ccache:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/sbin"
ldconfig -n /usr/local/freeswitch/lib
----------------------------------------------------------------------
Libraries have been installed in:
   /usr/local/freeswitch/lib

If you ever happen to want to link against installed libraries
in a given directory, LIBDIR, you must either use libtool, and
specify the full pathname of the library, or use the `-LLIBDIR'
flag during linking and do at least one of the following:
   - add LIBDIR to the `LD_LIBRARY_PATH' environment variable
     during execution
   - add LIBDIR to the `LD_RUN_PATH' environment variable
     during linking
   - use the `-Wl,-rpath -Wl,LIBDIR' linker flag
   - have your system administrator add LIBDIR to `/etc/ld.so.conf'

See any operating system documentation about shared libraries for
more information, such as the ld(1) and ld.so(8) manual pages.
----------------------------------------------------------------------

making install mod_h323
Compiling
/usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp...
quiet_libtool: compile:  g++ -g -ggdb -I/usr/local/include/ptlib
-I/usr/local/include/openh323 -I. -DPTRACING=1 -D_REENTRANT -fno-exceptions
-I/usr/local/src/freeswitch/src/include
-I/usr/local/src/freeswitch/src/include
-I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
-DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
-DHAVE_CONFIG_H -c
/usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp  -fPIC
-DPIC -o .libs/mod_h323.o
quiet_libtool: compile:  g++ -g -ggdb -I/usr/local/include/ptlib
-I/usr/local/include/openh323 -I. -DPTRACING=1 -D_REENTRANT -fno-exceptions
-I/usr/local/src/freeswitch/src/include
-I/usr/local/src/freeswitch/src/include
-I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
-DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
-DHAVE_CONFIG_H -c
/usr/local/src/freeswitch/src/mod/endpoints/mod_h323/mod_h323.cpp -o
mod_h323.o >/dev/null 2>&1
Creating mod_h323.la...
/usr/bin/ld: cannot find -lopenh323
collect2: ld returned 1 exit status
cat: .libs/mod_h323.log: No such file or directory
make[3]: *** [mod_h323.la] Error 1
make[2]: *** [install] Error 1
make[1]: *** [mod_h323-install] Error 1
make: *** [mod_h323-install] Error 2
[root at example freeswitch]#



On Sat, Oct 2, 2010 at 9:18 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

>
>
> On Fri, Oct 1, 2010 at 3:15 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> What about mod_h323?
>>
>>
>>
> Updated a bit: http://wiki.freeswitch.org/wiki/Mod_h323, you can't do
> anything wrong... it simply works.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From shamun.toha at gmail.com  Mon Oct  4 16:05:00 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Tue, 5 Oct 2010 01:05:00 +0200
Subject: [Freeswitch-users] FreeSwitch - mod_opal,
	mod_h323 never get installed
In-Reply-To: <AANLkTi=gF1N13NMyGhmgmNw+mJTyhjO06g0o_Pr2Hg9_@mail.gmail.com>
References: <AANLkTi=9Hw6dGtzdaKwYR+oA2R1y5P0vfuHJqJW8Fncd@mail.gmail.com>
	<AANLkTi=wvXA+mYPdCjSL07Wk=psBSXrYh58b7gkP0LTO@mail.gmail.com>
	<AANLkTi=gF1N13NMyGhmgmNw+mJTyhjO06g0o_Pr2Hg9_@mail.gmail.com>
Message-ID: <AANLkTinj79-7xmwuoTwr72FLTNTf0_LHVEZvzSV2oGGM@mail.gmail.com>

Hello, After following wiki mod_h323 it crash, when load mod_h323 is
applied. Please look as following:



################################################################################
#
#    Let this work !!     mod_h323
#
################################################################################


Step 1:
========
mkdir -p ~/h323
cd ~/h323

svn co
http://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/tags/v2_8_2/ptlib

export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig
export LD_LIBRARY_PATH=/usr/local/lib
cd ptlib
./configure
make
make install

Step 2:
========
cd ~/h323
wget
http://waix.dl.sourceforge.net/project/openh323gk/Sources/2.3.2/h323plus-20100525.tar.gz

tar xzvf h323plus-20100525.tar.gz
cd h323plus-20100525
export PTLIBDIR=~/h323/ptlib
./configure
make
make install



Step 3:
========

cd /usr/local/src/freeswitch

make mod_h323-clean
make mod_h323
make mod_h323-install

-- Successfully installed


Step 4:
=======
vi /usr/local/freeswitch/conf/autoload_configs/h323.conf.xml: put the
following

<configuration name="h323.conf" description="H323 Endpoints">
 <settings>
   <param name="trace-level" value="10"/>
   <param name="context" value="default"/>
   <param name="dialplan" value="XML"/>
   <param name="codec-prefs" value="PCMA,PCMU,GSM,G729"/>
   <param name="use-rtp-timer" value="true"/> <!-- enable RTP timer - should
always be enabled -->
   <param name="rtp-timer-name" value="soft"/> <!-- Timer name, soft is
default -->
   <!-- <param name="ptime-override-value" value="20"/> --> <!-- Override
negotiated ptime value with this value -->
   <param name="gk-address" value=""/>    <!-- empty to disable, "*" to
search LAN -->
   <param name="gk-identifer" value=""/>  <!-- optional name of gk -->
   <param name="gk-interface" value=""/>  <!-- mandatory listener interface
name -->
   <param name="gk-retry" value="30"/>  <!-- optional GK register retry
timer -->
   <param name="faststart" value="true"/>  <!-- optional -->
   <param name="h245tunneling" value="true"/>  <!-- optional -->
   <param name="h245insetup" value="true"/>  <!-- optional -->
   <param name="jitter-size" value="60"/>  <!-- optional -->
   <param name="progress-indication" value="8"/> <!-- optional - PI value in
progress message-->
   <param name="alerting-indication" value="8"/> <!-- optional - PI value in
alerting message-->
   <param name="endpoint-name" value="fs"/>
   <param name="fax-old-asn" value="true"/>
 </settings>
 <listeners>
   <listener name="default">
     <param name="h323-ip" value="$${local_ip_v4}"/>
     <param name="h323-port" value="1720"/>
   </listener>
 </listeners>
</configuration>


Step 5:
======


  _____              ______        _____ _____ ____ _   _
  |  ___| __ ___  ___/ ___\ \      / /_ _|_   _/ ___| | | |
  | |_ | '__/ _ \/ _ \___ \\ \ /\ / / | |  | || |   | |_| |
  |  _|| | |  __/  __/___) |\ V  V /  | |  | || |___|  _  |
  |_|  |_|  \___|\___|____/  \_/\_/  |___| |_| \____|_| |_|

************************************************************
* Anthony Minessale II, Michael Jerris, Brian West, Others *
* FreeSWITCH (http://www.freeswitch.org)                   *
* Paypal Donations Appreciated: paypal at freeswitch.org      *
* Brought to you by ClueCon http://www.cluecon.com/        *
************************************************************

2010-10-05 01:00:32.133303 [CONSOLE] switch_core.c:1649
FreeSWITCH Version 1.0.head (git-828960a 2010-09-25 12-51-42 -0500) Started.
Max Sessions[1000]
Session Rate[30]
SQL [Enabled]
freeswitch at example> load mod_h323
2010-10-05 01:00:58.396229 [INFO] mod_enum.c:808 ENUM Reloaded
2010-10-05 01:00:58.397198 [INFO] switch_time.c:950 Timezone reloaded 530
definitions
2010-10-05 01:00:58.505516 [CONSOLE] mod_h323.cpp:147 Starting loading
mod_h323
2010-10-05 01:00:58.518634 [CONSOLE] mod_h323.cpp:164 H323 mod initialized
and running
2010-10-05 01:00:58.518634 [CONSOLE] switch_loadable_module.c:944
Successfully Loaded [mod_h323]
2010-10-05 01:00:58.518634 [NOTICE] switch_loadable_module.c:145 Adding
Endpoint 'h323'

+OK Reloading XML
+OK

Segmentation fault (core dumped)
[root at example bin]#






On Sat, Oct 2, 2010 at 9:18 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

>
>
> On Fri, Oct 1, 2010 at 3:15 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> What about mod_h323?
>>
>>
>>
> Updated a bit: http://wiki.freeswitch.org/wiki/Mod_h323, you can't do
> anything wrong... it simply works.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
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From brian at freeswitch.org  Mon Oct  4 16:07:24 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 4 Oct 2010 18:07:24 -0500
Subject: [Freeswitch-users] FreeSwitch - mod_opal,
	mod_h323 never get installed
In-Reply-To: <AANLkTinLi8OL8+fOBOZE8FJ_wYSeHNCk3Su02-Q88umW@mail.gmail.com>
References: <AANLkTi=9Hw6dGtzdaKwYR+oA2R1y5P0vfuHJqJW8Fncd@mail.gmail.com>
	<AANLkTi=wvXA+mYPdCjSL07Wk=psBSXrYh58b7gkP0LTO@mail.gmail.com>
	<AANLkTi=gF1N13NMyGhmgmNw+mJTyhjO06g0o_Pr2Hg9_@mail.gmail.com>
	<AANLkTinLi8OL8+fOBOZE8FJ_wYSeHNCk3Su02-Q88umW@mail.gmail.com>
Message-ID: <0A8FD745-200D-404B-A264-AA54BEAD271B@freeswitch.org>


On Oct 4, 2010, at 5:24 PM, Shamun toha md wrote:

> /usr/bin/ld: cannot find -lopenh323
> collect2: ld returned 1 exit status



From mario_fs at mgtech.com  Mon Oct  4 16:18:18 2010
From: mario_fs at mgtech.com (Mario)
Date: Mon, 04 Oct 2010 16:18:18 -0700
Subject: [Freeswitch-users] Caller ID lost on second bridge in extension
Message-ID: <4CAA60BA.9040601@mgtech.com>

I have a 15 second ring to local extensions, then 15 seconds to local +
2 cell phones. However, when the second bridge is executed the caller ID
is replaced by 0000000000. I tried removing pieces below like the
execute app but it made no difference. Not found on the wiki or web
either. Anyone know where/why it's lost and how to keep it? Thanks. Mario


    <extension name="u1phone">
      <condition field="destination_number" expression="^2920$">
        <!--  Make sure they hear rings after playing hi  -->
	<action application="set" data="ringback=${us-ring}"/>
         <!--action application="playback" data="phrase:mvpeople:100"/-->
        <action application="execute_extension" data="2921 XML default"/>
         <!-- Next lines not needed here but seems a good idea if
special fesatures are used -->
        <action application="bind_meta_app" data="1 b s
execute_extension::dx XML features"/>
        <action application="bind_meta_app" data="2 b s
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
        <action application="bind_meta_app" data="3 b s
execute_extension::cf XML features"/>
        <!-- Next lines allow control after bridge timeout and hangup
after bridge -->
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
        <!-- ring home extensions only for ## seconds -->
        <action application="set" data="call_timeout=15"/>
        <action application="bridge" data="group/home@${domain_name}"/>
        <!-- Say we are keep trying.... -->
        <action application="execute_extension" data="2922 XML default"/>
        <action application="playback"
data="/usr/local/freeswitch/conf/mvsounds/we_trying_please_hold.wav"/>
        <!-- Setup so they hear music while ringing a second time -->
	<action application="set" data="ringback=$${hold_music}"/>
        <!-- this is needed to allow call_timeout to work after bridging
to a gateway -->
        <action application="set" data="ignore_early_media=true"/>
        <!-- ring home extensions +cell phone for ## seconds  NOTE THE
NEXT 2 LINES DID NOT WORK TO SET TIMEOUT -->
        <!--action application="set" data="bridge_answer_timeout=10"/-->
        <!--action application="set" data="call_timeout=15"/-->
        <!-- Any of the next 3 bridge lines work as long as :_: is used
to ring simoultaneously -->
        <action application="bridge"
data="<originate_timeout=15>group/home@${domain_name}:_:sofia/gateway/uid1/19161234567:_:sofia/gateway/uid1/19162223333"/>
        <!--action application="bridge"
data="group/home@${domain_name}:_:sofia/gateway/uid1/19161234567"/-->
        <!--action application="bridge"
data="<originate_timeout=15>sofia/gateway/uid1/19161234567,user/100@${domain_name}"/-->
        <!-- No answer go to voicemail -->
        <action application="answer"/>
        <action application="voicemail" data="default ${domain_name} 100"/>
      </condition>
   </extension>


From mario_fs at mgtech.com  Mon Oct  4 16:53:05 2010
From: mario_fs at mgtech.com (Mario)
Date: Mon, 04 Oct 2010 16:53:05 -0700
Subject: [Freeswitch-users] After several hours FS registers ITSP every
	minute, requires FS restart
Message-ID: <4CAA68E1.40508@mgtech.com>

I have 2 accounts with an ITSP that with the SIP profile below. After
Freeswtich has been up for several hours, especially during my overnight
testing, the accounts start to register every minute. This would not be
really bad except that incoming calls don't work and I have to restart
FS. I have the sofia trace below that also includes 407 Proxy
Authentication Required messages but that does not seem to affect
anything, the account works. Using the git version from this past
Saturday: FreeSWITCH Version 1.0.head (git-f288e3c 2010-10-01 23-35-51
-0400).
Thanks, Mario

<include>
      <gateway name="uuid1">
      <param name="proxy" value="sip.itsp.com"/>
      <param name="username" value="uuid1"/>
      <param name="from-user" value="uuid1"/>
      <param name="password" value="password"/>
      <param name="expire-seconds" value="600"/>
      <param name="register" value="true"/>
      <param name="retry-seconds" value="30"/>
    </gateway>
</include>


2010-10-04 16:36:09.758277 [NOTICE] sofia_reg.c:342 Registering uuid1
send 865 bytes to udp/[216.143.130.36]:5060 at 23:36:09.759509:


   ------------------------------------------------------------------------
   REGISTER sip:sip.itsp.com SIP/2.0
   Via: SIP/2.0/UDP 106.88.82.100:5080;rport;branch=z9hG4bKamKeDBQrZ9j3m
   Max-Forwards: 70
   From: <sip:uuid1 at sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc
   To: <sip:uuid1 at sip.itsp.com;transport=udp>
   Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7
   CSeq: 2768939 REGISTER
   Contact: <sip:gw+uuid1 at 106.88.82.100:5080;transport=udp;gw=uuid1>
   Expires: 60
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f288e3c 2010-10-01
23-35-51 -0400
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Proxy-Authorization: Digest username="uuid1", realm="rnktel.com",
nonce="4caa64a90001484bedc7fa432b7fba4e53832e2474cb5555", algorithm=MD5,
uri="sip:sip.itsp.com", response="7290bc4ec3cd32798c74d9ebd62c247e"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 567 bytes from udp/[216.143.130.36]:5060 at 23:36:09.866684:
   ------------------------------------------------------------------------
   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/UDP
106.88.82.100:5080;rport=5080;branch=z9hG4bKamKeDBQrZ9j3m;received=209.143.244.127
   From: <sip:uuid1 at sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc
   To:
<sip:uuid1 at sip.itsp.com;transport=udp>;tag=9214955d68c027e4c99a14a397962871.075c
   Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7
   CSeq: 2768939 REGISTER
   Proxy-Authenticate: Digest realm="rnktel.com",
nonce="4caa64e500015288f99aa6d49e6686bbe3e8d9d066275ce8", stale=true
   Server: OpenSIPS (1.6.3-notls (i386/freebsd))
   Content-Length: 0

   ------------------------------------------------------------------------
send 865 bytes to udp/[216.143.130.36]:5060 at 23:36:09.867061:
   ------------------------------------------------------------------------
   REGISTER sip:sip.itsp.com SIP/2.0
   Via: SIP/2.0/UDP 106.88.82.100:5080;rport;branch=z9hG4bKBXc7e67Uvj9Ng
   Max-Forwards: 70
   From: <sip:uuid1 at sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc
   To: <sip:uuid1 at sip.itsp.com;transport=udp>
   Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7
   CSeq: 2768940 REGISTER
   Contact: <sip:gw+uuid1 at 106.88.82.100:5080;transport=udp;gw=uuid1>
   Expires: 60
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f288e3c 2010-10-01
23-35-51 -0400
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Proxy-Authorization: Digest username="uuid1", realm="rnktel.com",
nonce="4caa64e500015288f99aa6d49e6686bbe3e8d9d066275ce8", algorithm=MD5,
uri="sip:sip.itsp.com", response="ea063d8a376643e491e1c41ce09dcf6e"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 549 bytes from udp/[216.143.130.36]:5060 at 23:36:09.975863:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
106.88.82.100:5080;rport=5080;branch=z9hG4bKBXc7e67Uvj9Ng;received=209.143.244.127
   From: <sip:uuid1 at sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc
   To:
<sip:uuid1 at sip.itsp.com;transport=udp>;tag=9214955d68c027e4c99a14a397962871.aa06
   Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7
   CSeq: 2768940 REGISTER
   Contact:
<sip:gw+uuid1 at 209.143.244.127:5080;transport=udp;gw=uuid1>;expires=60,
<sip:uuid1 at 209.143.244.127:5061>;expires=279
   Server: OpenSIPS (1.6.3-notls (i386/freebsd))
   Content-Length: 0

   ------------------------------------------------------------------------
2010-10-04 16:37:09.575144 [NOTICE] sofia_reg.c:342 Registering uuid1
send 865 bytes to udp/[216.143.130.36]:5060 at 23:37:09.623804:


   ------------------------------------------------------------------------
   REGISTER sip:sip.itsp.com SIP/2.0
   Via: SIP/2.0/UDP 106.88.82.100:5080;rport;branch=z9hG4bKc65Zg1rZSUZ8B
   Max-Forwards: 70
   From: <sip:uuid1 at sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc
   To: <sip:uuid1 at sip.itsp.com;transport=udp>
   Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7
   CSeq: 2768941 REGISTER
   Contact: <sip:gw+uuid1 at 106.88.82.100:5080;transport=udp;gw=uuid1>
   Expires: 60
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f288e3c 2010-10-01
23-35-51 -0400
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Proxy-Authorization: Digest username="uuid1", realm="rnktel.com",
nonce="4caa64e500015288f99aa6d49e6686bbe3e8d9d066275ce8", algorithm=MD5,
uri="sip:sip.itsp.com", response="ea063d8a376643e491e1c41ce09dcf6e"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 567 bytes from udp/[216.143.130.36]:5060 at 23:37:09.723821:
   ------------------------------------------------------------------------
   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/UDP
106.88.82.100:5080;rport=5080;branch=z9hG4bKc65Zg1rZSUZ8B;received=209.143.244.127
   From: <sip:uuid1 at sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc
   To:
<sip:uuid1 at sip.itsp.com;transport=udp>;tag=9214955d68c027e4c99a14a397962871.9bd6
   Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7
   CSeq: 2768941 REGISTER
   Proxy-Authenticate: Digest realm="rnktel.com",
nonce="4caa652100015c59a0f47e6209ee434ff2e43a548d62af15", stale=true
   Server: OpenSIPS (1.6.3-notls (i386/freebsd))
   Content-Length: 0

   ------------------------------------------------------------------------
send 865 bytes to udp/[216.143.130.36]:5060 at 23:37:09.724214:
   ------------------------------------------------------------------------
   REGISTER sip:sip.itsp.com SIP/2.0
   Via: SIP/2.0/UDP 106.88.82.100:5080;rport;branch=z9hG4bKDFZrjv92p4NUQ
   Max-Forwards: 70
   From: <sip:uuid1 at sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc
   To: <sip:uuid1 at sip.itsp.com;transport=udp>
   Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7
   CSeq: 2768942 REGISTER
   Contact: <sip:gw+uuid1 at 106.88.82.100:5080;transport=udp;gw=uuid1>
   Expires: 60
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f288e3c 2010-10-01
23-35-51 -0400
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Proxy-Authorization: Digest username="uuid1", realm="rnktel.com",
nonce="4caa652100015c59a0f47e6209ee434ff2e43a548d62af15", algorithm=MD5,
uri="sip:sip.itsp.com", response="48be3b8135d84350c145baf028460cfb"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 549 bytes from udp/[216.143.130.36]:5060 at 23:37:09.824242:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
106.88.82.100:5080;rport=5080;branch=z9hG4bKDFZrjv92p4NUQ;received=209.143.244.127
   From: <sip:uuid1 at sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc
   To:
<sip:uuid1 at sip.itsp.com;transport=udp>;tag=9214955d68c027e4c99a14a397962871.c855
   Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7
   CSeq: 2768942 REGISTER
   Contact:
<sip:gw+uuid1 at 209.143.244.127:5080;transport=udp;gw=uuid1>;expires=60,
<sip:uuid1 at 209.143.244.127:5061>;expires=219
   Server: OpenSIPS (1.6.3-notls (i386/freebsd))
   Content-Length: 0




From brian at freeswitch.org  Mon Oct  4 18:18:21 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 4 Oct 2010 20:18:21 -0500
Subject: [Freeswitch-users] After several hours FS registers ITSP every
	minute, requires FS restart
In-Reply-To: <4CAA68E1.40508@mgtech.com>
References: <4CAA68E1.40508@mgtech.com>
Message-ID: <CE016105-5A4F-4F16-9DB5-50DEB9179AB4@freeswitch.org>

Looks like your ITSP keeps pushing your expire time down lower and lower each time.  You have to honor what they send you... I suspect they keep pushing it down till its in a tight loop.  What does the spec say about this?

/b

On Oct 4, 2010, at 6:53 PM, Mario wrote:

> I have 2 accounts with an ITSP that with the SIP profile below. After
> Freeswtich has been up for several hours, especially during my overnight
> testing, the accounts start to register every minute. This would not be
> really bad except that incoming calls don't work and I have to restart
> FS. I have the sofia trace below that also includes 407 Proxy
> Authentication Required messages but that does not seem to affect
> anything, the account works. Using the git version from this past
> Saturday: FreeSWITCH Version 1.0.head (git-f288e3c 2010-10-01 23-35-51
> -0400).
> Thanks, Mario
> 
> <include>
>      <gateway name="uuid1">
>      <param name="proxy" value="sip.itsp.com"/>
>      <param name="username" value="uuid1"/>
>      <param name="from-user" value="uuid1"/>
>      <param name="password" value="password"/>
>      <param name="expire-seconds" value="600"/>
>      <param name="register" value="true"/>
>      <param name="retry-seconds" value="30"/>
>    </gateway>
> </include>



From mario_fs at mgtech.com  Mon Oct  4 18:31:34 2010
From: mario_fs at mgtech.com (Mario)
Date: Mon, 04 Oct 2010 18:31:34 -0700
Subject: [Freeswitch-users] After several hours FS registers ITSP every
 minute, requires FS restart
In-Reply-To: <CE016105-5A4F-4F16-9DB5-50DEB9179AB4@freeswitch.org>
References: <4CAA68E1.40508@mgtech.com>
	<CE016105-5A4F-4F16-9DB5-50DEB9179AB4@freeswitch.org>
Message-ID: <4CAA7FF6.3040307@mgtech.com>

That's odd because it does not happen on the existing SPA9000 PBX which
I am trying to replace. Could FS be telling them to do this? The log
shows a re-register every 5 minutes forever. Wonder why it's different
for FS....

On 10/04/10 18:18, Brian West wrote:
> Looks like your ITSP keeps pushing your expire time down lower and lower each time.  You have to honor what they send you... I suspect they keep pushing it down till its in a tight loop.  What does the spec say about this?
> 
> /b
> 
> On Oct 4, 2010, at 6:53 PM, Mario wrote:
> 
>> I have 2 accounts with an ITSP that with the SIP profile below. After
>> Freeswtich has been up for several hours, especially during my overnight
>> testing, the accounts start to register every minute. This would not be
>> really bad except that incoming calls don't work and I have to restart
>> FS. I have the sofia trace below that also includes 407 Proxy
>> Authentication Required messages but that does not seem to affect
>> anything, the account works. Using the git version from this past
>> Saturday: FreeSWITCH Version 1.0.head (git-f288e3c 2010-10-01 23-35-51
>> -0400).
>> Thanks, Mario
>>
>> <include>
>>      <gateway name="uuid1">
>>      <param name="proxy" value="sip.itsp.com"/>
>>      <param name="username" value="uuid1"/>
>>      <param name="from-user" value="uuid1"/>
>>      <param name="password" value="password"/>
>>      <param name="expire-seconds" value="600"/>
>>      <param name="register" value="true"/>
>>      <param name="retry-seconds" value="30"/>
>>    </gateway>
>> </include>
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From Joshua.Foshee at LogixCom.com  Mon Oct  4 14:09:57 2010
From: Joshua.Foshee at LogixCom.com (Joshua Foshee)
Date: Mon, 4 Oct 2010 16:09:57 -0500
Subject: [Freeswitch-users] Play_fsv application
Message-ID: <06502C073AD9394AADB3CA7FD94931BC0519DF25@okc1x1.Logixcom.com>

I have something strange happen when I play back a fsv file. It will
play super-fast. You can hear the audio go fast too then you get a long
busy. Any ideas what would cause the speed to increase on playback?

 

Thanks,

Josh

 

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From mike at jerris.com  Mon Oct  4 19:25:00 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 Oct 2010 22:25:00 -0400
Subject: [Freeswitch-users] High Availability and hostname used by Sofia
In-Reply-To: <AANLkTimF36eFM1Cjvni7t3LqFGEUUapzhM8sm6y8=ys-@mail.gmail.com>
References: <EA736C6E-2651-43A6-BE6B-D202DC951F38@scarlet-internet.nl>
	<AANLkTikJacUOvCCOwc_LNvNRwdVzfoYE4jNchxPBRPS-@mail.gmail.com>
	<AANLkTimF36eFM1Cjvni7t3LqFGEUUapzhM8sm6y8=ys-@mail.gmail.com>
Message-ID: <EED809D9-CE14-49A1-B5DE-BE81182F2DE4@jerris.com>

Its probably not possible to do so due to that nat mappings on the clients router.  if you are having the proxy juggle the distribution of the registration anyways, you might want to just have your freeswitch always send the calls to the proxy and have it handle the registration and all communication with the client

Mike

On Oct 4, 2010, at 11:48 AM, Henry Huang wrote:

> Anthony:
> 
> But if I the FS servers do not share a floating IP but does share the registration DB via Core ODBC. Even though the same domain is forced in the sofia profile, the registrations will not be treated as they are on every FS box I have correct?
> 
> The idea is to have OpenSIPS in front of FS servers to distribute registrations. But would like to have any of the FS servers be able to contact the registered user on the shared DB. Is it possible somehow?
> 
> 
> Henry Huang Unified Communication System R&D, Founder of UniC Solution
> US: +1 (626) 606-3306 | ??(Taiwan): +886 933847619
> Contact Me   
> IM  red_rain_seven at gmail.com  unicsolution  b_ball_henry at hotmail.com
> ?	"You are the embodiment of the information you choose to accept and act upon. To change your circumstances you need to change your thinking and subsequent actions." - Adlin Sinclair	?
>   Get this email app!  
> 
> 
> 
> On Mon, Oct 4, 2010 at 10:27 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> the registrations table intentionally does not consider the hostname
> so you can cluster many boxes on the same domain.
> 
> On Mon, Oct 4, 2010 at 6:57 AM, Leon de Rooij <leon at scarlet-internet.nl> wrote:
> > Hello List,
> >
> > This is going to be a long mail, please bear with me :-)
> >
> > I'd like to configure High Availability for a FreeSWITCH installation
> > by using a failover server. I already discussed it a bit with Sekil on
> > IRC, but I thought it'd be useful to discuss it here as well.
> >
> >
> > The idea is as follows:
> >
> > - I have 2 servers both with an identical FreeSWITCH installation, FS1
> > and FS2.
> > - Both servers have different static IP's but also one floating IP by
> > means of heartbeat / haresources.
> > - Both FS instances have one Sofia SIP profile that use the floating IP.
> > - Both FS instances are always started (so the backup server will fail
> > to load its SIP profile because the IP is not configured - which is
> > expected behaviour)
> > - There'll be a /etc/ha.d/resource.d "resource script" that starts the
> > SIP profile (through fs_cli -x) the moment it becomes primary
> > - The resource script will also send a "sofia recover" after the SIP
> > profile has been brought up
> >
> >
> > This way:
> >
> > - There'll be one logical switch.
> > - The moment FS2 becomes primary, users will be reachable through the
> > entry in 'sip_registrations' table in the ODBC db, even though they
> > may have registered on FS1
> > - FS2 will start to send SIP options to NAT'ed useragents (keepalive /
> > udp hole punching) the moment the SIP profile is started.
> > - No client useragent involvement is necessary when failing over (like
> > re-register)
> > - Running calls will be continued because of the "sofia recover"
> >
> >
> > Only problem here is that when a useragent registers to us, then FS
> > also stores its own hostname in "sip_registrations" table in the
> > columns "hostname" and "orig_hostname" (from
> > mod_sofia_globals.hostname). This column is also used in the SELECT
> > query when bridging to a user - so that means if the user registered
> > to FS1, then a failover happens, then if a call is bridged towards
> > that user on FS2, it won't find it because it will search with FS2's
> > hostname in the SELECT query, right ?
> >
> > If everything I'm writing so far is correct, then would it be alright
> > to override the hostname in sofia.conf ? I think that would solve my
> > problem. I already looked it up - it is first set in mod_sofia.c at
> > line 4565 through gethostname(). It can be overriden in sofia.c
> > (function config_sofia) around line 2845. It would probably also work
> > to give both hosts an identical hostname in the OS, but I don't like
> > that (because of rsyslog and more).
> >
> >
> > I'm not convinced yet on how to proceed, please let me know if I'm
> > going the wrong direction. If I'm successful on getting this running,
> > I'll document everything on the wiki - also the haresources script I
> > already wrote I'll put in my contrib dir (or elsewhere in main tree if
> > preferred).
> >
> >
> > Thanks for your time,
> >
> > Leon
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From b_ball_henry at hotmail.com  Mon Oct  4 20:55:21 2010
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Tue, 5 Oct 2010 11:55:21 +0800
Subject: [Freeswitch-users] High Availability and hostname used by Sofia
In-Reply-To: <EED809D9-CE14-49A1-B5DE-BE81182F2DE4@jerris.com>
References: <EA736C6E-2651-43A6-BE6B-D202DC951F38@scarlet-internet.nl>
	<AANLkTikJacUOvCCOwc_LNvNRwdVzfoYE4jNchxPBRPS-@mail.gmail.com>
	<AANLkTimF36eFM1Cjvni7t3LqFGEUUapzhM8sm6y8=ys-@mail.gmail.com>
	<EED809D9-CE14-49A1-B5DE-BE81182F2DE4@jerris.com>
Message-ID: <AANLkTinkUBRrvwNd0fjix+K_Us4LwLyyaxrhjYtwvtY8@mail.gmail.com>

Mike:

Despite the NAT Mapping issue on the client side. Is it possible on the FS
server side to recognize the same client even though it was registered to
another FS server in the first place ? (Keep in mind that the registration
DB is shared between all FS servers).


Henry

On Tue, Oct 5, 2010 at 10:25 AM, Michael Jerris <mike at jerris.com> wrote:

> Its probably not possible to do so due to that nat mappings on the clients
> router.  if you are having the proxy juggle the distribution of the
> registration anyways, you might want to just have your freeswitch always
> send the calls to the proxy and have it handle the registration and all
> communication with the client
>
> Mike
>
> On Oct 4, 2010, at 11:48 AM, Henry Huang wrote:
>
> Anthony:
>
> But if I the FS servers do not share a floating IP but does share the
> registration DB via Core ODBC. Even though the same domain is forced in the
> sofia profile, the registrations will not be treated as they are on every FS
> box I have correct?
>
> The idea is to have OpenSIPS in front of FS servers to distribute
> registrations. But would like to have any of the FS servers be able to
> contact the registered user on the shared DB. Is it possible somehow?
>
>
> Henry Huang Unified Communication System R&D, Founder of UniC Solution
> US: +1 (626) 606-3306 | ??(Taiwan): +886 933847619
> Contact Me [image: LinkedIn]<http://www.linkedin.com/pub/henry-huang/1/654/578> [image:
> Facebook] <http://www.facebook.com/profile.php?id=595148342> [image:
> Twitter] <http://twitter.com/unicsolution>
> IM [image: Google Talk/] red_rain_seven at gmail.com [image: Skype/]unicsolution [image:
> MSN/] b_ball_henry at hotmail.com
>  ?"You are the embodiment of the information you choose to accept and act
> upon. To change your circumstances you need to change your thinking and
> subsequent actions." - Adlin Sinclair<http://www.quotesdaddy.com/quote/6942/adlin-sinclair/you-are-the-embodiment-of-the-information-you-choose>
> ?  Get this email app! <http://www.wisestamp.com/apps/quotes?utm_source=extension&utm_medium=email&utm_term=quotes&utm_campaign=apps>
>
>
>
> On Mon, Oct 4, 2010 at 10:27 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> the registrations table intentionally does not consider the hostname
>> so you can cluster many boxes on the same domain.
>>
>> On Mon, Oct 4, 2010 at 6:57 AM, Leon de Rooij <leon at scarlet-internet.nl>
>> wrote:
>> > Hello List,
>> >
>> > This is going to be a long mail, please bear with me :-)
>> >
>> > I'd like to configure High Availability for a FreeSWITCH installation
>> > by using a failover server. I already discussed it a bit with Sekil on
>> > IRC, but I thought it'd be useful to discuss it here as well.
>> >
>> >
>> > The idea is as follows:
>> >
>> > - I have 2 servers both with an identical FreeSWITCH installation, FS1
>> > and FS2.
>> > - Both servers have different static IP's but also one floating IP by
>> > means of heartbeat / haresources.
>> > - Both FS instances have one Sofia SIP profile that use the floating IP.
>> > - Both FS instances are always started (so the backup server will fail
>> > to load its SIP profile because the IP is not configured - which is
>> > expected behaviour)
>> > - There'll be a /etc/ha.d/resource.d "resource script" that starts the
>> > SIP profile (through fs_cli -x) the moment it becomes primary
>> > - The resource script will also send a "sofia recover" after the SIP
>> > profile has been brought up
>> >
>> >
>> > This way:
>> >
>> > - There'll be one logical switch.
>> > - The moment FS2 becomes primary, users will be reachable through the
>> > entry in 'sip_registrations' table in the ODBC db, even though they
>> > may have registered on FS1
>> > - FS2 will start to send SIP options to NAT'ed useragents (keepalive /
>> > udp hole punching) the moment the SIP profile is started.
>> > - No client useragent involvement is necessary when failing over (like
>> > re-register)
>> > - Running calls will be continued because of the "sofia recover"
>> >
>> >
>> > Only problem here is that when a useragent registers to us, then FS
>> > also stores its own hostname in "sip_registrations" table in the
>> > columns "hostname" and "orig_hostname" (from
>> > mod_sofia_globals.hostname). This column is also used in the SELECT
>> > query when bridging to a user - so that means if the user registered
>> > to FS1, then a failover happens, then if a call is bridged towards
>> > that user on FS2, it won't find it because it will search with FS2's
>> > hostname in the SELECT query, right ?
>> >
>> > If everything I'm writing so far is correct, then would it be alright
>> > to override the hostname in sofia.conf ? I think that would solve my
>> > problem. I already looked it up - it is first set in mod_sofia.c at
>> > line 4565 through gethostname(). It can be overriden in sofia.c
>> > (function config_sofia) around line 2845. It would probably also work
>> > to give both hosts an identical hostname in the OS, but I don't like
>> > that (because of rsyslog and more).
>> >
>> >
>> > I'm not convinced yet on how to proceed, please let me know if I'm
>> > going the wrong direction. If I'm successful on getting this running,
>> > I'll document everything on the wiki - also the haresources script I
>> > already wrote I'll put in my contrib dir (or elsewhere in main tree if
>> > preferred).
>> >
>> >
>> > Thanks for your time,
>> >
>> > Leon
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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 <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From william.suffill at gmail.com  Mon Oct  4 21:51:37 2010
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 5 Oct 2010 00:51:37 -0400
Subject: [Freeswitch-users] Caller ID lost on second bridge in extension
In-Reply-To: <4CAA60BA.9040601@mgtech.com>
References: <4CAA60BA.9040601@mgtech.com>
Message-ID: <AANLkTiksUFQuOQFSjFPaSDbOp28wUrb962p-+8PDd5sp@mail.gmail.com>

All 0's is the default callerid from vars.xml from the stock configs. You
can change it there or set it in the dialplan prior or pass it as part of
the bridge command.

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Setting_Outgoing_CallerID

should help.


-- W

PS: Feel free to ask if any other questions just getting a lil late to give
it too much thought atm.
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From covici at ccs.covici.com  Tue Oct  5 00:55:51 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 05 Oct 2010 03:55:51 -0400
Subject: [Freeswitch-users] Are there mandatory extensions in any context?
Message-ID: <11917.1286265351@ccs.covici.com>

Hi.  I notice that there are a few extensions at the beginning of the
default context in the dialplan such as unroll-loops and a few more.  I
wonder if I make a new context for some specific purpose, to hold
conferences, for instance -- do I need such extensions in that context
as well?

thanks in advance for any suggestions.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From fdelawarde at wirelessmundi.com  Tue Oct  5 01:19:38 2010
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Tue, 05 Oct 2010 10:19:38 +0200
Subject: [Freeswitch-users] FreeSwitch Out of threads!
Message-ID: <1286266778.5790.22.camel@luna.tc.commsmundi.com>

Hello,

I decided to do a bit of "stability" testing leaving 100 sessions
calling a playback "tone_stream://$${tetris}" on my dual-core CPU with
4G of RAM during a few hours.


After 5 hours working flawelessly, I started getting lots of:

2010-10-04 23:53:14.188596 [CRIT] switch_event.c:342 Out of threads!


Later combined with:

2010-10-04 23:59:02.692961 [CRIT] switch_event.c:325 Event system overloading


Both messages (the second one at an increased rate) followed by a few:

2010-10-05 00:25:43.711929 [CRIT] switch_time.c:755 Virtual Migration Detected! Syncing Clock


Just before a crash where this time I did not get a core (??)...


Any idea what I could have done wrong? 


Fran?ois.
PS: This was NOT intended to be a stress test, I just wanted to check
how FS would handle at low-medium load during a few hours.



From david.ponzone at ipeva.fr  Tue Oct  5 01:35:20 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 5 Oct 2010 10:35:20 +0200
Subject: [Freeswitch-users] FreeSwitch Out of threads!
In-Reply-To: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
References: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
Message-ID: <0B146F6F-D6A7-4AAB-962B-93FE6DEB52E7@ipeva.fr>

Fran?ois,

are the 100 sessions the same from the beginning to the end ?
If yes, I am afraid this is going to be considered as a stress test,  
as this would not happen with live traffic.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 05/10/2010 ? 10:19, Fran?ois Delawarde a ?crit :

> Hello,
>
> I decided to do a bit of "stability" testing leaving 100 sessions
> calling a playback "tone_stream://$${tetris}" on my dual-core CPU with
> 4G of RAM during a few hours.
>
>
> After 5 hours working flawelessly, I started getting lots of:
>
> 2010-10-04 23:53:14.188596 [CRIT] switch_event.c:342 Out of threads!
>
>
> Later combined with:
>
> 2010-10-04 23:59:02.692961 [CRIT] switch_event.c:325 Event system  
> overloading
>
>
> Both messages (the second one at an increased rate) followed by a few:
>
> 2010-10-05 00:25:43.711929 [CRIT] switch_time.c:755 Virtual  
> Migration Detected! Syncing Clock
>
>
> Just before a crash where this time I did not get a core (??)...
>
>
> Any idea what I could have done wrong?
>
>
> Fran?ois.
> PS: This was NOT intended to be a stress test, I just wanted to check
> how FS would handle at low-medium load during a few hours.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mnhassan at usa.net  Tue Oct  5 01:38:36 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Tue, 5 Oct 2010 14:38:36 +0600
Subject: [Freeswitch-users] FreeSwitch Out of threads!
In-Reply-To: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
References: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
Message-ID: <AANLkTing67qXyj3msDQaxun2XOgocRDVYXvPh9hsZWFW@mail.gmail.com>

Did you increase the number of file descriptors?

Regards
HASSAN


On 2010-10-05, Fran?ois Delawarde <fdelawarde at wirelessmundi.com> wrote:
> Hello,
>
> I decided to do a bit of "stability" testing leaving 100 sessions
> calling a playback "tone_stream://$${tetris}" on my dual-core CPU with
> 4G of RAM during a few hours.
>
>
> After 5 hours working flawelessly, I started getting lots of:
>
> 2010-10-04 23:53:14.188596 [CRIT] switch_event.c:342 Out of threads!
>
>
> Later combined with:
>
> 2010-10-04 23:59:02.692961 [CRIT] switch_event.c:325 Event system
> overloading
>
>
> Both messages (the second one at an increased rate) followed by a few:
>
> 2010-10-05 00:25:43.711929 [CRIT] switch_time.c:755 Virtual Migration
> Detected! Syncing Clock
>
>
> Just before a crash where this time I did not get a core (??)...
>
>
> Any idea what I could have done wrong?
>
>
> Fran?ois.
> PS: This was NOT intended to be a stress test, I just wanted to check
> how FS would handle at low-medium load during a few hours.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device


From mnhassan at usa.net  Tue Oct  5 01:41:33 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Tue, 5 Oct 2010 14:41:33 +0600
Subject: [Freeswitch-users] Are there mandatory extensions in any
	context?
In-Reply-To: <11917.1286265351@ccs.covici.com>
References: <11917.1286265351@ccs.covici.com>
Message-ID: <AANLkTikdatPF9ek66zio7afsppsKZ2M38SxdLynHHEQb@mail.gmail.com>

Nope. The default extension is there to provide some working examples
of things that you can do with FreeSWITCH. For our operation we
created a completely new profile with separate IP and dialplan that
has only what we want.

Regards
HASSAN


On 2010-10-05, covici at ccs.covici.com <covici at ccs.covici.com> wrote:
> Hi.  I notice that there are a few extensions at the beginning of the
> default context in the dialplan such as unroll-loops and a few more.  I
> wonder if I make a new context for some specific purpose, to hold
> conferences, for instance -- do I need such extensions in that context
> as well?
>
> thanks in advance for any suggestions.
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>          John Covici
>          covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device


From t.mahe at telemaque.fr  Tue Oct  5 01:46:22 2010
From: t.mahe at telemaque.fr (=?ISO-8859-1?Q?Tristan_Mah=E9?=)
Date: Tue, 05 Oct 2010 10:46:22 +0200
Subject: [Freeswitch-users] Is anyone familiar with Telemaque?
In-Reply-To: <AANLkTimib+3T7L9sP28HN6BREJF5ca2j2nou-84YU=Ln@mail.gmail.com>
References: <AANLkTimib+3T7L9sP28HN6BREJF5ca2j2nou-84YU=Ln@mail.gmail.com>
Message-ID: <4CAAE5DE.6080407@telemaque.fr>

  There's someone working for them on list :)

Feel free to ask me questions Michael.

Regards,

Gled.

Le 10/04/2010 08:39 PM, Michael Collins a ?crit :
> Dear FreeSWITCH Community,
>
> Some of you may have stumbled upon this news item:
>
> http://blogs.oracle.com/mysql/2010/10/innovating_with_open_source_call_center_services.html
>
> Evidently there is a company in France (Telemaque) who is using MySQL 
> + Kamailio + FreeSWITCH for some heavy duty call processing. If you 
> are at all familiar with this company please let me know. I'd like to 
> learn more about what they are doing. (Je ne parle pas boucoup de 
> Francais :( )
>
> Thanks for your help,
> MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From dome at tel.co.th  Tue Oct  5 02:10:20 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 5 Oct 2010 16:10:20 +0700
Subject: [Freeswitch-users] FreeSwitch Out of threads!
In-Reply-To: <AANLkTing67qXyj3msDQaxun2XOgocRDVYXvPh9hsZWFW@mail.gmail.com>
References: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
	<AANLkTing67qXyj3msDQaxun2XOgocRDVYXvPh9hsZWFW@mail.gmail.com>
Message-ID: <AANLkTimYA=XM4pPJRrUOA_er5WkhZfqhJQO0E4E6LKgj@mail.gmail.com>

Please use 64bit and
ulimit -c unlimited
ulimit -d unlimited
ulimit -f unlimited
ulimit -i unlimited
ulimit -n 999999
ulimit -q unlimited
ulimit -u unlimited
ulimit -v unlimited
ulimit -x unlimited
ulimit -s 240
ulimit -l unlimited

before start freeswitch


2010/10/5 Nyamul Hassan <mnhassan at usa.net>:
> Did you increase the number of file descriptors?
>
> Regards
> HASSAN
>
>
> On 2010-10-05, Fran?ois Delawarde <fdelawarde at wirelessmundi.com> wrote:
>> Hello,
>>
>> I decided to do a bit of "stability" testing leaving 100 sessions
>> calling a playback "tone_stream://$${tetris}" on my dual-core CPU with
>> 4G of RAM during a few hours.
>>
>>
>> After 5 hours working flawelessly, I started getting lots of:
>>
>> 2010-10-04 23:53:14.188596 [CRIT] switch_event.c:342 Out of threads!
>>
>>
>> Later combined with:
>>
>> 2010-10-04 23:59:02.692961 [CRIT] switch_event.c:325 Event system
>> overloading
>>
>>
>> Both messages (the second one at an increased rate) followed by a few:
>>
>> 2010-10-05 00:25:43.711929 [CRIT] switch_time.c:755 Virtual Migration
>> Detected! Syncing Clock
>>
>>
>> Just before a crash where this time I did not get a core (??)...
>>
>>
>> Any idea what I could have done wrong?
>>
>>
>> Fran?ois.
>> PS: This was NOT intended to be a stress test, I just wanted to check
>> how FS would handle at low-medium load during a few hours.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> --
> Sent from my mobile device
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From fdelawarde at wirelessmundi.com  Tue Oct  5 02:45:27 2010
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Tue, 05 Oct 2010 11:45:27 +0200
Subject: [Freeswitch-users] FreeSwitch Out of threads!
In-Reply-To: <AANLkTimYA=XM4pPJRrUOA_er5WkhZfqhJQO0E4E6LKgj@mail.gmail.com>
References: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
	<AANLkTing67qXyj3msDQaxun2XOgocRDVYXvPh9hsZWFW@mail.gmail.com>
	<AANLkTimYA=XM4pPJRrUOA_er5WkhZfqhJQO0E4E6LKgj@mail.gmail.com>
Message-ID: <1286271927.5790.33.camel@luna.tc.commsmundi.com>

I maintain "only" 100 sessions at all times, with a duration of 20-30s,
and no more than 10cps, so it is NOT a stress test, come on!

I mean this would not even load an Asterisk server, and loading the
server was not an objective, just to maintain low-medium load over a
period of time. It worked very well during 5h.

Why would I need to ulimit the crap out of my server then? :-)

Fran?ois.

On Tue, 2010-10-05 at 16:10 +0700, Dome Charoenyost wrote:
> Please use 64bit and
> ulimit -c unlimited
> ulimit -d unlimited
> ulimit -f unlimited
> ulimit -i unlimited
> ulimit -n 999999
> ulimit -q unlimited
> ulimit -u unlimited
> ulimit -v unlimited
> ulimit -x unlimited
> ulimit -s 240
> ulimit -l unlimited
> 
> before start freeswitch
> 
> 
> 2010/10/5 Nyamul Hassan <mnhassan at usa.net>:
> > Did you increase the number of file descriptors?
> >
> > Regards
> > HASSAN
> >
> >
> > On 2010-10-05, Fran?ois Delawarde <fdelawarde at wirelessmundi.com> wrote:
> >> Hello,
> >>
> >> I decided to do a bit of "stability" testing leaving 100 sessions
> >> calling a playback "tone_stream://$${tetris}" on my dual-core CPU with
> >> 4G of RAM during a few hours.
> >>
> >>
> >> After 5 hours working flawelessly, I started getting lots of:
> >>
> >> 2010-10-04 23:53:14.188596 [CRIT] switch_event.c:342 Out of threads!
> >>
> >>
> >> Later combined with:
> >>
> >> 2010-10-04 23:59:02.692961 [CRIT] switch_event.c:325 Event system
> >> overloading
> >>
> >>
> >> Both messages (the second one at an increased rate) followed by a few:
> >>
> >> 2010-10-05 00:25:43.711929 [CRIT] switch_time.c:755 Virtual Migration
> >> Detected! Syncing Clock
> >>
> >>
> >> Just before a crash where this time I did not get a core (??)...
> >>
> >>
> >> Any idea what I could have done wrong?
> >>
> >>
> >> Fran?ois.
> >> PS: This was NOT intended to be a stress test, I just wanted to check
> >> how FS would handle at low-medium load during a few hours.
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> > --
> > Sent from my mobile device
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From elihayunfs at gmail.com  Tue Oct  5 03:45:50 2010
From: elihayunfs at gmail.com (Eli Hayun)
Date: Tue, 05 Oct 2010 12:45:50 +0200
Subject: [Freeswitch-users] No autoanswer with Originate call
In-Reply-To: <AANLkTimmweNy+xdwQDuuuhTt0OEMfccpqfwUwYz50OHy@mail.gmail.com>
References: <1286181373.20572.5.camel@localhost>
	<AANLkTimmweNy+xdwQDuuuhTt0OEMfccpqfwUwYz50OHy@mail.gmail.com>
Message-ID: <1286275550.20572.6.camel@localhost>

Thanks for the reply.

I did a test on Polycom phone and it is working fine.
The same script on Snome didn't work

Eli

On Mon, 2010-10-04 at 10:29 -0700, Michael Collins wrote:
> Can you pastebin a debug log of working vs. non-working? It would
> probably be helpful to see the entire extension as well as a complete
> Lua script. (You can write a minimal Lua script that demonstrates the
> behavior and then others can test.) Of course, the most important
> question: are you running latest HEAD? :)
> 
> -MC
> 
> On Mon, Oct 4, 2010 at 1:36 AM, Eli Hayun <elihayunfs at gmail.com>
> wrote:
>         Hi
>         When I am dialing from A phone to B phone and I do :
>         
>               <action application="set" data="sip_auto_answer=true" />
>                <action application="set" data="sip_h_Call-Info=<sip:
>         ${domain_name}>;answer-after=2" />
>         
>         everything is OK
>         
>         When I try from lua script 
>         api:executeString("originate [originate string] AUTO_ANS xml
>         default")
>         
>         and in the dialplan I put the set autoanswer=true it is not
>         working.
>         
>         any help?
>         
>         Eli
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From bernhard.suttner at winet.ch  Tue Oct  5 04:00:53 2010
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Tue, 5 Oct 2010 13:00:53 +0200
Subject: [Freeswitch-users] api_on_answer
Message-ID: <f3635107-6a96-412b-892d-0d4a5423ed25@winet.ch>

Hi,

I use api_on_answer to call a lua script. Within the lua script I want access to session variables but the lua variable "session" does not exists. How could I get access to them?

Thanks in advance.

Best regards,
Bernhard




From fdelawarde at wirelessmundi.com  Tue Oct  5 04:18:35 2010
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Tue, 05 Oct 2010 13:18:35 +0200
Subject: [Freeswitch-users] api_on_answer
In-Reply-To: <f3635107-6a96-412b-892d-0d4a5423ed25@winet.ch>
References: <f3635107-6a96-412b-892d-0d4a5423ed25@winet.ch>
Message-ID: <1286277515.5790.41.camel@luna.tc.commsmundi.com>

Hi,

Try to run your lua with execute_on_answer instead of api_on_answer.

Fran?ois.


On Tue, 2010-10-05 at 13:00 +0200, Bernhard Suttner wrote:
> Hi,
> 
> I use api_on_answer to call a lua script. Within the lua script I want access to session variables but the lua variable "session" does not exists. How could I get access to them?
> 
> Thanks in advance.
> 
> Best regards,
> Bernhard




From ivdreg at gmail.com  Tue Oct  5 05:14:35 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Tue, 5 Oct 2010 15:14:35 +0300
Subject: [Freeswitch-users] FreeSWITCH overrides/dose no accept hangup cause
Message-ID: <AANLkTin6o0nsBM=PeMQKi+Cdv1iut_H-t0v6r1nNDkwR@mail.gmail.com>

Hi All,

I have 2 FS boxes that communicates each other. First originates call to
second and on second FS perl scrips process a call. On some errors during
processing I hangup call with specific hangup cause and everything is OK.

For example:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.10.1.101;rport=5060;branch=z9hG4bK77698B1e1BZ3j
From: "11111111111" <sip:11111111111 at 10.10.1.101<sip%3A11111111111 at 10.10.1.101>
>;tag=X628H5HUDy86S
To: <sip:999999999999 at 10.10.1.110:7060>;tag=Ztg1r5S60rQFF
Call-ID: cca7076d-4b12-122e-44b6-7fd0651f7bfe
CSeq: 2802191 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, refer
*Reason: Q.850;cause=38;text="NETWORK_OUT_OF_ORDER"*
Content-Length: 0
Remote-Party-ID: "999999999999"
<sip:999999999999 at 10.10.1.110<sip%3A999999999999 at 10.10.1.110>
>;party=calling;privacy=off;screen=no

As you can see Q.850 cause is 38 - NETWORK OUT OF ORDER but on first FS
application info says:

variable_proto_specific_hangup_cause: [sip:503]
variable_sip_hangup_phrase: [Service Unavailable]
variable_originate_disposition: [NORMAL_TEMPORARY_FAILURE]

SIP Cause of both NORMAL_TEMPORARY_FAILURE and NETWORK_OUT_OF_ORDER is 503
and is the same but Q.850 cause is different.
Is it normal behavior for 1.0.6 ? Is it also possible first FS to respect
Q.850 cause ?

Thanks
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From bernhard.suttner at winet.ch  Tue Oct  5 05:29:21 2010
From: bernhard.suttner at winet.ch (Bernhard Suttner)
Date: Tue, 5 Oct 2010 14:29:21 +0200
Subject: [Freeswitch-users] api_on_answer
In-Reply-To: <1286277515.5790.41.camel@luna.tc.commsmundi.com>
References: <f3635107-6a96-412b-892d-0d4a5423ed25@winet.ch>
	<1286277515.5790.41.camel@luna.tc.commsmundi.com>
Message-ID: <6fe552f8-5356-466d-96eb-1010f5a7604a@winet.ch>

Hi,

thanks. That would be a solution, but I would prefer to have the api and not a application.

Best regards,
Bernhard

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Fran?ois Delawarde
Gesendet: Dienstag, 5. Oktober 2010 13:19
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] api_on_answer

Hi,

Try to run your lua with execute_on_answer instead of api_on_answer.

Fran?ois.


On Tue, 2010-10-05 at 13:00 +0200, Bernhard Suttner wrote:
> Hi,
> 
> I use api_on_answer to call a lua script. Within the lua script I want access to session variables but the lua variable "session" does not exists. How could I get access to them?
> 
> Thanks in advance.
> 
> Best regards,
> Bernhard



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



From tculjaga at gmail.com  Tue Oct  5 06:45:08 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 5 Oct 2010 15:45:08 +0200
Subject: [Freeswitch-users] FreeSwitch - mod_opal,
	mod_h323 never get installed
In-Reply-To: <AANLkTinj79-7xmwuoTwr72FLTNTf0_LHVEZvzSV2oGGM@mail.gmail.com>
References: <AANLkTi=9Hw6dGtzdaKwYR+oA2R1y5P0vfuHJqJW8Fncd@mail.gmail.com>
	<AANLkTi=wvXA+mYPdCjSL07Wk=psBSXrYh58b7gkP0LTO@mail.gmail.com>
	<AANLkTi=gF1N13NMyGhmgmNw+mJTyhjO06g0o_Pr2Hg9_@mail.gmail.com>
	<AANLkTinj79-7xmwuoTwr72FLTNTf0_LHVEZvzSV2oGGM@mail.gmail.com>
Message-ID: <AANLkTinOPDhyLcCcLh=W1y4Zr8zxws_YLPTHKtNiko9m@mail.gmail.com>

On Tue, Oct 5, 2010 at 1:05 AM, Shamun toha md <shamun.toha at gmail.com>wrote:

> Hello, After following wiki mod_h323 it crash, when load mod_h323 is
> applied. Please look as following:
>
>
>
>
> ################################################################################
> #
> #    Let this work !!     mod_h323
> #
>
> ################################################################################
>
>
> Step 1:
> ========
> mkdir -p ~/h323
> cd ~/h323
>
> svn co
> http://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/tags/v2_8_2/ptlib
>
> export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig
> export LD_LIBRARY_PATH=/usr/local/lib
> cd ptlib
> ./configure
> make
> make install
>
> Step 2:
> ========
> cd ~/h323
> wget
> http://waix.dl.sourceforge.net/project/openh323gk/Sources/2.3.2/h323plus-20100525.tar.gz
>
> tar xzvf h323plus-20100525.tar.gz
> cd h323plus-20100525
> export PTLIBDIR=~/h323/ptlib
> ./configure
> make
> make install
>
>
>
> Step 3:
> ========
>
> cd /usr/local/src/freeswitch
>
> make mod_h323-clean
> make mod_h323
> make mod_h323-install
>
> -- Successfully installed
>
>
> Step 4:
> =======
> vi /usr/local/freeswitch/conf/autoload_configs/h323.conf.xml: put the
> following
>
> <configuration name="h323.conf" description="H323 Endpoints">
>  <settings>
>    <param name="trace-level" value="10"/>
>    <param name="context" value="default"/>
>    <param name="dialplan" value="XML"/>
>    <param name="codec-prefs" value="PCMA,PCMU,GSM,G729"/>
>    <param name="use-rtp-timer" value="true"/> <!-- enable RTP timer -
> should always be enabled -->
>    <param name="rtp-timer-name" value="soft"/> <!-- Timer name, soft is
> default -->
>    <!-- <param name="ptime-override-value" value="20"/> --> <!-- Override
> negotiated ptime value with this value -->
>    <param name="gk-address" value=""/>    <!-- empty to disable, "*" to
> search LAN -->
>    <param name="gk-identifer" value=""/>  <!-- optional name of gk -->
>    <param name="gk-interface" value=""/>  <!-- mandatory listener interface
> name -->
>    <param name="gk-retry" value="30"/>  <!-- optional GK register retry
> timer -->
>    <param name="faststart" value="true"/>  <!-- optional -->
>    <param name="h245tunneling" value="true"/>  <!-- optional -->
>    <param name="h245insetup" value="true"/>  <!-- optional -->
>    <param name="jitter-size" value="60"/>  <!-- optional -->
>    <param name="progress-indication" value="8"/> <!-- optional - PI value
> in progress message-->
>    <param name="alerting-indication" value="8"/> <!-- optional - PI value
> in alerting message-->
>    <param name="endpoint-name" value="fs"/>
>    <param name="fax-old-asn" value="true"/>
>  </settings>
>  <listeners>
>    <listener name="default">
>      <param name="h323-ip" value="$${local_ip_v4}"/>
>      <param name="h323-port" value="1720"/>
>    </listener>
>  </listeners>
> </configuration>
>
>
> Step 5:
> ======
>
>
>   _____              ______        _____ _____ ____ _   _
>   |  ___| __ ___  ___/ ___\ \      / /_ _|_   _/ ___| | | |
>   | |_ | '__/ _ \/ _ \___ \\ \ /\ / / | |  | || |   | |_| |
>   |  _|| | |  __/  __/___) |\ V  V /  | |  | || |___|  _  |
>   |_|  |_|  \___|\___|____/  \_/\_/  |___| |_| \____|_| |_|
>
> ************************************************************
> * Anthony Minessale II, Michael Jerris, Brian West, Others *
> * FreeSWITCH (http://www.freeswitch.org)                   *
> * Paypal Donations Appreciated: paypal at freeswitch.org      *
> * Brought to you by ClueCon http://www.cluecon.com/        *
> ************************************************************
>
> 2010-10-05 01:00:32.133303 [CONSOLE] switch_core.c:1649
> FreeSWITCH Version 1.0.head (git-828960a 2010-09-25 12-51-42 -0500)
> Started.
> Max Sessions[1000]
> Session Rate[30]
> SQL [Enabled]
> freeswitch at example> load mod_h323
> 2010-10-05 01:00:58.396229 [INFO] mod_enum.c:808 ENUM Reloaded
> 2010-10-05 01:00:58.397198 [INFO] switch_time.c:950 Timezone reloaded 530
> definitions
> 2010-10-05 01:00:58.505516 [CONSOLE] mod_h323.cpp:147 Starting loading
> mod_h323
> 2010-10-05 01:00:58.518634 [CONSOLE] mod_h323.cpp:164 H323 mod initialized
> and running
> 2010-10-05 01:00:58.518634 [CONSOLE] switch_loadable_module.c:944
> Successfully Loaded [mod_h323]
> 2010-10-05 01:00:58.518634 [NOTICE] switch_loadable_module.c:145 Adding
> Endpoint 'h323'
>
> +OK Reloading XML
> +OK
>
> Segmentation fault (core dumped)
> [root at example bin]#
>
>
>
>

are you sure you removed all previous installed libraries (ptlib and
h323plus) ??
Make sure the linker links the correct library.

when making mod_h323 .. the compiler looks into /usr/lib before
/usr/local/lib

T.
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From moises.silva at gmail.com  Tue Oct  5 06:59:03 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 5 Oct 2010 09:59:03 -0400
Subject: [Freeswitch-users] FreeSwitch Out of threads!
In-Reply-To: <1286271927.5790.33.camel@luna.tc.commsmundi.com>
References: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
	<AANLkTing67qXyj3msDQaxun2XOgocRDVYXvPh9hsZWFW@mail.gmail.com>
	<AANLkTimYA=XM4pPJRrUOA_er5WkhZfqhJQO0E4E6LKgj@mail.gmail.com>
	<1286271927.5790.33.camel@luna.tc.commsmundi.com>
Message-ID: <AANLkTi=Gis9k=pHQ1si5cXeOCjk6EskjN0ri3Nak9YVE@mail.gmail.com>

On Tue, Oct 5, 2010 at 5:45 AM, Fran?ois Delawarde
<fdelawarde at wirelessmundi.com> wrote:
> I maintain "only" 100 sessions at all times, with a duration of 20-30s,
> and no more than 10cps, so it is NOT a stress test, come on!
>
> I mean this would not even load an Asterisk server, and loading the
> server was not an objective, just to maintain low-medium load over a
> period of time. It worked very well during 5h.
>
> Why would I need to ulimit the crap out of my server then? :-)

Don't feel alone, you're not the only one.

We're getting the same pair of messages. We know what it means, we
have just not found the cause of it. In our case happens quite fast
(about an hour IIRC) when doing stress testing of about 900 calls at a
time lasting 10 seconds each at a rate of about 40cps.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From brian at freeswitch.org  Tue Oct  5 07:07:01 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 5 Oct 2010 09:07:01 -0500
Subject: [Freeswitch-users] FreeSWITCH overrides/dose no accept hangup
	cause
In-Reply-To: <AANLkTin6o0nsBM=PeMQKi+Cdv1iut_H-t0v6r1nNDkwR@mail.gmail.com>
References: <AANLkTin6o0nsBM=PeMQKi+Cdv1iut_H-t0v6r1nNDkwR@mail.gmail.com>
Message-ID: <D358829E-E260-460C-81B2-A4192B558C83@freeswitch.org>

Everyone running 1.0.6 should be running git head right now it will be 1.0.7 any day now.  Stop living in the past!  ;)

/b

On Oct 5, 2010, at 7:14 AM, ivdreg ivdreg wrote:

> SIP Cause of both NORMAL_TEMPORARY_FAILURE and NETWORK_OUT_OF_ORDER is 503 and is the same but Q.850 cause is different.
> Is it normal behavior for 1.0.6 ? Is it also possible first FS to respect Q.850 cause ?
> 
> Thanks



From brian at freeswitch.org  Tue Oct  5 07:09:38 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 5 Oct 2010 09:09:38 -0500
Subject: [Freeswitch-users] FreeSwitch Out of threads!
In-Reply-To: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
References: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
Message-ID: <FA4EFA56-4E28-4DA2-99FC-C9B7F3F74360@freeswitch.org>

what cpu?

/b

On Oct 5, 2010, at 3:19 AM, Fran?ois Delawarde wrote:

> Hello,
> 
> I decided to do a bit of "stability" testing leaving 100 sessions
> calling a playback "tone_stream://$${tetris}" on my dual-core CPU with
> 4G of RAM during a few hours.
> 
> 
> After 5 hours working flawelessly, I started getting lots of:
> 
> 2010-10-04 23:53:14.188596 [CRIT] switch_event.c:342 Out of threads!
> 
> 
> Later combined with:
> 
> 2010-10-04 23:59:02.692961 [CRIT] switch_event.c:325 Event system overloading
> 
> 
> Both messages (the second one at an increased rate) followed by a few:
> 
> 2010-10-05 00:25:43.711929 [CRIT] switch_time.c:755 Virtual Migration Detected! Syncing Clock
> 



From fdelawarde at wirelessmundi.com  Tue Oct  5 07:34:27 2010
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Tue, 05 Oct 2010 16:34:27 +0200
Subject: [Freeswitch-users] FreeSwitch Out of threads!
In-Reply-To: <FA4EFA56-4E28-4DA2-99FC-C9B7F3F74360@freeswitch.org>
References: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
	<FA4EFA56-4E28-4DA2-99FC-C9B7F3F74360@freeswitch.org>
Message-ID: <1286289267.5790.47.camel@luna.tc.commsmundi.com>

AMD Athlon(tm) 64 X2 Dual Core Processor 3800+

see:
http://pastebin.freeswitch.org/14135

Fran?ois.

On Tue, 2010-10-05 at 09:09 -0500, Brian West wrote:
> what cpu?
> 
> /b
> 
> On Oct 5, 2010, at 3:19 AM, Fran?ois Delawarde wrote:
> 
> > Hello,
> > 
> > I decided to do a bit of "stability" testing leaving 100 sessions
> > calling a playback "tone_stream://$${tetris}" on my dual-core CPU with
> > 4G of RAM during a few hours.
> > 
> > 
> > After 5 hours working flawelessly, I started getting lots of:
> > 
> > 2010-10-04 23:53:14.188596 [CRIT] switch_event.c:342 Out of threads!
> > 
> > 
> > Later combined with:
> > 
> > 2010-10-04 23:59:02.692961 [CRIT] switch_event.c:325 Event system overloading
> > 
> > 
> > Both messages (the second one at an increased rate) followed by a few:
> > 
> > 2010-10-05 00:25:43.711929 [CRIT] switch_time.c:755 Virtual Migration Detected! Syncing Clock
> > 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mnhassan at usa.net  Tue Oct  5 07:35:26 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Tue, 5 Oct 2010 20:35:26 +0600
Subject: [Freeswitch-users] FreeSwitch Out of threads!
In-Reply-To: <AANLkTi=Gis9k=pHQ1si5cXeOCjk6EskjN0ri3Nak9YVE@mail.gmail.com>
References: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
	<AANLkTing67qXyj3msDQaxun2XOgocRDVYXvPh9hsZWFW@mail.gmail.com>
	<AANLkTimYA=XM4pPJRrUOA_er5WkhZfqhJQO0E4E6LKgj@mail.gmail.com>
	<1286271927.5790.33.camel@luna.tc.commsmundi.com>
	<AANLkTi=Gis9k=pHQ1si5cXeOCjk6EskjN0ri3Nak9YVE@mail.gmail.com>
Message-ID: <AANLkTingwTJpYLtXChhGyzw6jPLq2ot_82bdptmRZR18@mail.gmail.com>

We learned from another thread about how Sangoma regularly tests 32 E1
setups in the lab, and were preparing a concept paper to show a
telecom that traditional vendors can be replaced using open source
even for large scale implementations. Does this mean we should rethink
our strategy?

Regards
HASSAN


On 2010-10-05, Moises Silva <moises.silva at gmail.com> wrote:
> On Tue, Oct 5, 2010 at 5:45 AM, Fran?ois Delawarde
> <fdelawarde at wirelessmundi.com> wrote:
>> I maintain "only" 100 sessions at all times, with a duration of 20-30s,
>> and no more than 10cps, so it is NOT a stress test, come on!
>>
>> I mean this would not even load an Asterisk server, and loading the
>> server was not an objective, just to maintain low-medium load over a
>> period of time. It worked very well during 5h.
>>
>> Why would I need to ulimit the crap out of my server then? :-)
>
> Don't feel alone, you're not the only one.
>
> We're getting the same pair of messages. We know what it means, we
> have just not found the cause of it. In our case happens quite fast
> (about an hour IIRC) when doing stress testing of about 900 calls at a
> time lasting 10 seconds each at a rate of about 40cps.
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
> L3R 9R6 Canada
> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device


From dave.redmore at spigotsystems.com  Tue Oct  5 07:50:12 2010
From: dave.redmore at spigotsystems.com (Dave Redmore)
Date: Tue, 5 Oct 2010 09:50:12 -0500 (CDT)
Subject: [Freeswitch-users] RTCP support
Message-ID: <6808999.01286290212435.JavaMail.root@zimbra1.spigotsystems.com>

Hi All, 


I've done some investigating into getting RTCP support working, but am finding a lot of conflicting information. 


The WIKI says RTCP can be configured on a per session bases or for an entire profile. 


I've added "<param name="rtcp-audio-interval-msec" value="1000"/> to my profile config and restarted FS, but am not seeing any RTCP packets in my captures. 


Is there something else that needs to be done? 




Thanks. 


Dave Redmore 
Spigot Networks, Inc. 



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From moises.silva at gmail.com  Tue Oct  5 07:53:31 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 5 Oct 2010 10:53:31 -0400
Subject: [Freeswitch-users] FreeSwitch Out of threads!
In-Reply-To: <AANLkTingwTJpYLtXChhGyzw6jPLq2ot_82bdptmRZR18@mail.gmail.com>
References: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
	<AANLkTing67qXyj3msDQaxun2XOgocRDVYXvPh9hsZWFW@mail.gmail.com>
	<AANLkTimYA=XM4pPJRrUOA_er5WkhZfqhJQO0E4E6LKgj@mail.gmail.com>
	<1286271927.5790.33.camel@luna.tc.commsmundi.com>
	<AANLkTi=Gis9k=pHQ1si5cXeOCjk6EskjN0ri3Nak9YVE@mail.gmail.com>
	<AANLkTingwTJpYLtXChhGyzw6jPLq2ot_82bdptmRZR18@mail.gmail.com>
Message-ID: <AANLkTinHEdrO9xvEpuX7SLCq+=5MJujjL0em3G6bp19i@mail.gmail.com>

On Tue, Oct 5, 2010 at 10:35 AM, Nyamul Hassan <mnhassan at usa.net> wrote:
> We learned from another thread about how Sangoma regularly tests 32 E1
> setups in the lab, and were preparing a concept paper to show a
> telecom that traditional vendors can be replaced using open source
> even for large scale implementations. Does this mean we should rethink
> our strategy?
>

FreeSWITCH is under constant development and the performance behavior
may change from the git of one month ago, to the latest one. In short,
stress testing is tricky, it's just matter of finding the right
balance.

Do not expect blindly checking out latest freeswitch git any day and
always have bug free software (if such thing exists I'd like to see
it). You need to do your homework for you particular implementation.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From fdelawarde at wirelessmundi.com  Tue Oct  5 07:55:51 2010
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Tue, 05 Oct 2010 16:55:51 +0200
Subject: [Freeswitch-users] FreeSwitch Out of threads!
In-Reply-To: <AANLkTingwTJpYLtXChhGyzw6jPLq2ot_82bdptmRZR18@mail.gmail.com>
References: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
	<AANLkTing67qXyj3msDQaxun2XOgocRDVYXvPh9hsZWFW@mail.gmail.com>
	<AANLkTimYA=XM4pPJRrUOA_er5WkhZfqhJQO0E4E6LKgj@mail.gmail.com>
	<1286271927.5790.33.camel@luna.tc.commsmundi.com>
	<AANLkTi=Gis9k=pHQ1si5cXeOCjk6EskjN0ri3Nak9YVE@mail.gmail.com>
	<AANLkTingwTJpYLtXChhGyzw6jPLq2ot_82bdptmRZR18@mail.gmail.com>
Message-ID: <1286290551.5790.58.camel@luna.tc.commsmundi.com>

I wouldn't jump to conclusions that fast, it's just a bug with current
git head, probably only happening in very specific cases.

If I learned anything from hollywood movies, it would be that bugs or
bad people always end up being tracked down and destroyed.

Fran?ois.


On Tue, 2010-10-05 at 20:35 +0600, Nyamul Hassan wrote:
> We learned from another thread about how Sangoma regularly tests 32 E1
> setups in the lab, and were preparing a concept paper to show a
> telecom that traditional vendors can be replaced using open source
> even for large scale implementations. Does this mean we should rethink
> our strategy?
> 
> Regards
> HASSAN
> 
> 
> On 2010-10-05, Moises Silva <moises.silva at gmail.com> wrote:
> > On Tue, Oct 5, 2010 at 5:45 AM, Fran?ois Delawarde
> > <fdelawarde at wirelessmundi.com> wrote:
> >> I maintain "only" 100 sessions at all times, with a duration of 20-30s,
> >> and no more than 10cps, so it is NOT a stress test, come on!
> >>
> >> I mean this would not even load an Asterisk server, and loading the
> >> server was not an objective, just to maintain low-medium load over a
> >> period of time. It worked very well during 5h.
> >>
> >> Why would I need to ulimit the crap out of my server then? :-)
> >
> > Don't feel alone, you're not the only one.
> >
> > We're getting the same pair of messages. We know what it means, we
> > have just not found the cause of it. In our case happens quite fast
> > (about an hour IIRC) when doing stress testing of about 900 calls at a
> > time lasting 10 seconds each at a rate of about 40cps.
> >
> > Moises Silva
> > Senior Software Engineer
> > Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
> > L3R 9R6 Canada
> > t. 1 905 474 1990 x128 | e. moy at sangoma.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 



From mario_fs at mgtech.com  Tue Oct  5 08:00:40 2010
From: mario_fs at mgtech.com (Mario)
Date: Tue, 05 Oct 2010 08:00:40 -0700
Subject: [Freeswitch-users] Caller ID lost on second bridge in extension
In-Reply-To: <AANLkTiksUFQuOQFSjFPaSDbOp28wUrb962p-+8PDd5sp@mail.gmail.com>
References: <4CAA60BA.9040601@mgtech.com>
	<AANLkTiksUFQuOQFSjFPaSDbOp28wUrb962p-+8PDd5sp@mail.gmail.com>
Message-ID: <4CAB3D98.7050802@mgtech.com>

Wring callerid, It's the inbound that's missing. I guess I should repost.

On 10/04/10 21:51, William Suffill wrote:
> All 0's is the default callerid from vars.xml from the stock configs.
> You can change it there or set it in the dialplan prior or pass it as
> part of the bridge command.
> 
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Setting_Outgoing_CallerID
> 
> should help.
> 
> 
> -- W
> 
> PS: Feel free to ask if any other questions just getting a lil late to
> give it too much thought atm.
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From hesser4900 at gmail.com  Tue Oct  5 07:19:41 2010
From: hesser4900 at gmail.com (Holger Esser)
Date: Tue, 5 Oct 2010 09:19:41 -0500
Subject: [Freeswitch-users] generate inband dtmf
Message-ID: <AANLkTikzbAv+kWfcmVPdq7FsFA+VAL9FRZqhF5hWietX@mail.gmail.com>

Hi guys,



I have a question about the start_dtmf_generate function. I have a dialplan
entry that is called from a socket call like this:

originate sofia/gateway/teliax/xxxxxxxxxx 5000



<condition field="destination_number" expression="^5000$">

        <action application="answer"/>

        <action application="sleep" data="1000"/>

        <action application="playback" data="voicemail/vm-goodbye.wav"/>

        <action application="sleep" data="2000"/>

        <action application="start_dtmf_generate"/>

        <action application="send_dtmf" data="0123456789"/>

        <action application="sleep" data="90000"/>

      </condition>





However, DTMF are still via the RFC and not inband. I am using the latest
GIT.



Any ideas?



Thx
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From ochere at gmail.com  Tue Oct  5 07:26:27 2010
From: ochere at gmail.com (Frank Ochere)
Date: Tue, 05 Oct 2010 17:26:27 +0300
Subject: [Freeswitch-users] limit_execute and distributor concurrent module
 usage
Message-ID: <1286288787.9945.10.camel@me>

Hello,

I have the following in distributor.conf.xml 

    <list name="aa_rr" total-weight="2">
      <node name="GW_001" weight="1"/>
      <node name="GW_002" weight="1"/>
    </list>



in dialplan

     <extension name="TESTGW">
       <condition field="destination_number" expression="^7777(\d+)$">
         <action application="limit_execute" data="hash TESTGW TESTGW 1
bridge sofia/gateway/${distributor(aa_rr)}/$1"/>
       </condition>
     </extension>

my question is, is it possible to use limit_execute and distributor
concurrently as in TESTGW above? ie limit calls to 1 per gateway but
being alternated as specified in list aa_rr


sofia debug below

Dialplan: sofia/external/00000000 at 1.1.1.1:5060 Action limit_execute(hash
TESTGW TESTGW 1 bridge
sofia/gateway/${distributor(sat_rr)}/00777717171) 
2010-10-05 17:13:15.616463 [DEBUG] switch_core_state_machine.c:119
(sofia/external/00000000 at 1.1.1.1:5060) State Change CS_ROUTING ->
CS_EXECUTE
2010-10-05 17:13:15.616463 [DEBUG] switch_core_session.c:1047 Send
signal sofia/external/00000000 at 1.1.1.1:5060 [BREAK]
2010-10-05 17:13:15.616463 [DEBUG] switch_core_state_machine.c:341
(sofia/external/00000000 at 1.1.1.1:5060) State ROUTING going to sleep
2010-10-05 17:13:15.616463 [DEBUG] switch_core_state_machine.c:314
(sofia/external/00000000 at 1.1.1.1:5060) Running State Change CS_EXECUTE
2010-10-05 17:13:15.616463 [DEBUG] switch_core_state_machine.c:348
(sofia/external/00000000 at 1.1.1.1:5060) State EXECUTE
2010-10-05 17:13:15.616463 [DEBUG] mod_sofia.c:239
sofia/external/00000000 at 1.1.1.1:5060 SOFIA EXECUTE
2010-10-05 17:13:15.616463 [DEBUG] switch_core_state_machine.c:157
sofia/external/00000000 at 1.1.1.1:5060 Standard EXECUTE
EXECUTE sofia/external/00000000 at 1.1.1.1:5060 limit_execute(hash TESTGW
TESTGW 1 bridge sofia/gateway/-err/00777717171)
2010-10-05 17:13:15.616463 [INFO] switch_limit.c:126 incr called:
TESTGW_TESTGW max:1, interval:0
2010-10-05 17:13:15.616463 [INFO] mod_hash.c:202 Usage for TESTGW_TESTGW
is now 1/1
EXECUTE sofia/external/00000000 at 1.1.1.1:5060
bridge(sofia/gateway/-err/00777717171)
2010-10-05 17:13:15.616463 [ERR] mod_sofia.c:3725 Invalid Gateway
2010-10-05 17:13:15.616463 [NOTICE] mod_sofia.c:4047 Close Channel N/A
[CS_NEW]
2010-10-05 17:13:15.616463 [DEBUG] switch_core_state_machine.c:430 ()
Running State Change CS_DESTROY
2010-10-05 17:13:15.616463 [DEBUG] switch_core_state_machine.c:440 (N/A)
State DESTROY
2010-10-05 17:13:15.616463 [DEBUG] mod_sofia.c:362 N/A SOFIA DESTROY
2010-10-05 17:13:15.616463 [DEBUG] switch_core_state_machine.c:440 (N/A)
State DESTROY going to sleep
2010-10-05 17:13:15.616463 [ERR] switch_ivr_originate.c:2648 Cannot
create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
2010-10-05 17:13:15.616463 [DEBUG] switch_ivr_originate.c:3456 Originate
Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
2010-10-05 17:13:15.616463 [INFO] mod_dptools.c:2411 Originate Failed.
Cause: INVALID_NUMBER_FORMAT
2010-10-05 17:13:15.616463 [DEBUG] switch_channel.c:2359
(sofia/external/00000000 at 1.1.1.1:5060) Callstate Change RINGING ->
HANGUP
2010-10-05 17:13:15.616463 [NOTICE] mod_dptools.c:2474 Hangup
sofia/external/00000000 at 1.1.1.1:5060 [CS_EXECUTE]
[INVALID_NUMBER_FORMAT]
2010-10-05 17:13:15.616463 [DEBUG] switch_channel.c:2375 Send signal
sofia/external/00000000 at 1.1.1.1:5060 [KILL]
2010-10-05 17:13:15.616463 [INFO] mod_hash.c:300 Usage for TESTGW_TESTGW
is now 0



From hesser4900 at gmail.com  Tue Oct  5 08:04:44 2010
From: hesser4900 at gmail.com (Holger Esser)
Date: Tue, 5 Oct 2010 10:04:44 -0500
Subject: [Freeswitch-users] test
Message-ID: <AANLkTimOeUGBkdA9MeAhone_neH55YSZk0YsjM2brRif@mail.gmail.com>

test
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From ivdreg at gmail.com  Tue Oct  5 08:05:48 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Tue, 5 Oct 2010 18:05:48 +0300
Subject: [Freeswitch-users] FreeSWITCH overrides/dose no accept hangup
	cause
In-Reply-To: <D358829E-E260-460C-81B2-A4192B558C83@freeswitch.org>
References: <AANLkTin6o0nsBM=PeMQKi+Cdv1iut_H-t0v6r1nNDkwR@mail.gmail.com>
	<D358829E-E260-460C-81B2-A4192B558C83@freeswitch.org>
Message-ID: <AANLkTikt=sOFih2fjOTUYRAXrX5s41nmPoOq7fR_6DBZ@mail.gmail.com>

Hi Brian,

Correct me if I'm wrong but I thing this is a bug in 1.0.6 that exist also
in 1.0.7.
In src/mod/endpoints/mod_sofia/sofia.c:

void sofia_handle_sip_i_bye(switch_core_session_t *session, int status  ....
.....
.....
        if (sip->sip_reason && sip->sip_reason->re_protocol &&
(!strcasecmp(sip->sip_reason->re_protocol, "Q.850")

|| !strcasecmp(sip->sip_reason->re_protocol, "FreeSWITCH")

|| !strcasecmp(sip->sip_reason->re_protocol, profile->username)) &&
sip->sip_reason->re_cause) {
                tech_pvt->q850_cause = atoi(sip->sip_reason->re_cause);
                cause = tech_pvt->q850_cause;
        } else {
                cause = sofia_glue_sip_cause_to_freeswitch(status);
        }
......

If this if is false default cause for 503 ( returned by
sofia_glue_sip_cause_to_freeswitch ) is NORMAL_TEMPORARY_FAILURE.

Regards


2010/10/5 Brian West <brian at freeswitch.org>

> Everyone running 1.0.6 should be running git head right now it will be
> 1.0.7 any day now.  Stop living in the past!  ;)
>
> /b
>
> On Oct 5, 2010, at 7:14 AM, ivdreg ivdreg wrote:
>
> > SIP Cause of both NORMAL_TEMPORARY_FAILURE and NETWORK_OUT_OF_ORDER is
> 503 and is the same but Q.850 cause is different.
> > Is it normal behavior for 1.0.6 ? Is it also possible first FS to respect
> Q.850 cause ?
> >
> > Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Joshua.Foshee at LogixCom.com  Tue Oct  5 08:12:35 2010
From: Joshua.Foshee at LogixCom.com (Joshua Foshee)
Date: Tue, 5 Oct 2010 10:12:35 -0500
Subject: [Freeswitch-users] Play_fsv application
In-Reply-To: <06502C073AD9394AADB3CA7FD94931BC0519DF25@okc1x1.Logixcom.com>
References: <06502C073AD9394AADB3CA7FD94931BC0519DF25@okc1x1.Logixcom.com>
Message-ID: <06502C073AD9394AADB3CA7FD94931BC0519DF28@okc1x1.Logixcom.com>

I have something strange happen when I play back a fsv file. It will
play super-fast. You can hear the audio go fast too then you get a long
busy. Any ideas what would cause the speed to increase on playback?

 

Thanks,

Josh

 

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From shamun.toha at gmail.com  Tue Oct  5 08:20:40 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Tue, 5 Oct 2010 17:20:40 +0200
Subject: [Freeswitch-users] test
In-Reply-To: <AANLkTimOeUGBkdA9MeAhone_neH55YSZk0YsjM2brRif@mail.gmail.com>
References: <AANLkTimOeUGBkdA9MeAhone_neH55YSZk0YsjM2brRif@mail.gmail.com>
Message-ID: <AANLkTinpwfU2SVng99k7925==ZYx-VTeFCorcedWgwwm@mail.gmail.com>

it works!


On Tue, Oct 5, 2010 at 5:04 PM, Holger Esser <hesser4900 at gmail.com> wrote:

> test
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From shamun.toha at gmail.com  Tue Oct  5 08:23:21 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Tue, 5 Oct 2010 17:23:21 +0200
Subject: [Freeswitch-users] FreeSwitch - mod_opal,
	mod_h323 never get installed
In-Reply-To: <AANLkTinOPDhyLcCcLh=W1y4Zr8zxws_YLPTHKtNiko9m@mail.gmail.com>
References: <AANLkTi=9Hw6dGtzdaKwYR+oA2R1y5P0vfuHJqJW8Fncd@mail.gmail.com>
	<AANLkTi=wvXA+mYPdCjSL07Wk=psBSXrYh58b7gkP0LTO@mail.gmail.com>
	<AANLkTi=gF1N13NMyGhmgmNw+mJTyhjO06g0o_Pr2Hg9_@mail.gmail.com>
	<AANLkTinj79-7xmwuoTwr72FLTNTf0_LHVEZvzSV2oGGM@mail.gmail.com>
	<AANLkTinOPDhyLcCcLh=W1y4Zr8zxws_YLPTHKtNiko9m@mail.gmail.com>
Message-ID: <AANLkTikn_hFh4Gr3NscX-VXcFscBT3S=m+FRc5KB53to@mail.gmail.com>

It works now, i can run it and use it, but another big issue, i will close
this email and open as new thread. Which is completely different.

Thanks for sharing..


> are you sure you removed all previous installed libraries (ptlib and
> h323plus) ??
> Make sure the linker links the correct library.
>
> when making mod_h323 .. the compiler looks into /usr/lib before
> /usr/local/lib
>
> T.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mario_fs at mgtech.com  Tue Oct  5 08:26:11 2010
From: mario_fs at mgtech.com (Mario)
Date: Tue, 05 Oct 2010 08:26:11 -0700
Subject: [Freeswitch-users] latest git pull has error, can't install
Message-ID: <4CAB4393.4010900@mgtech.com>

While working on a loop in FS I did a git pull and make current. I had
done this before with no problem but today I got the error below. Any
ideas? Thanks. Mario


cd libs/sofia-sip && make clean
make[1]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip'
 cd . && /bin/sh /usr/local/src/freeswitch/libs/sofia-sip/missing --run
automake-1.11 --foreign
configure.ac:47: warning: AC_CACHE_VAL(ac_cc_environment, ...):
suspicious cache-id, must contain _cv_ to be cached
../../lib/autoconf/general.m4:1998: AC_CACHE_VAL is expanded from...
../../lib/autoconf/general.m4:2019: AC_CACHE_CHECK is expanded from...
m4/sac-general.m4:351: AC_CHECK_COMPILATION_ENVIRONMENT is expanded from...
configure.ac:47: the top level
configure.ac:49: warning: AC_PROG_CPP was called before SAC_TOOL_CC
m4/sac-general.m4:90: SAC_TOOL_CC is expanded from...
configure.ac:49: the top level
configure.ac:56: required file `./compile' not found
configure.ac:56:   `automake --add-missing' can install `compile'
make[1]: *** [Makefile.in] Error 1
make[1]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip'
make: *** [update-clean] Error 2


From anthony.minessale at gmail.com  Tue Oct  5 08:26:28 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Oct 2010 10:26:28 -0500
Subject: [Freeswitch-users] FreeSwitch Out of threads!
In-Reply-To: <1286290551.5790.58.camel@luna.tc.commsmundi.com>
References: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
	<AANLkTing67qXyj3msDQaxun2XOgocRDVYXvPh9hsZWFW@mail.gmail.com>
	<AANLkTimYA=XM4pPJRrUOA_er5WkhZfqhJQO0E4E6LKgj@mail.gmail.com>
	<1286271927.5790.33.camel@luna.tc.commsmundi.com>
	<AANLkTi=Gis9k=pHQ1si5cXeOCjk6EskjN0ri3Nak9YVE@mail.gmail.com>
	<AANLkTingwTJpYLtXChhGyzw6jPLq2ot_82bdptmRZR18@mail.gmail.com>
	<1286290551.5790.58.camel@luna.tc.commsmundi.com>
Message-ID: <AANLkTinhiuJUQA26WevUPXiOtMHb=5bXKTR3N2dv3vrS@mail.gmail.com>

If there is a bug why are you not opening it on JIRA?
http://jira.freeswitch.org

We are doing work on improving the event system to be faster and use
less memory.

If it caused some problems, that is not uncommon, software development
requires testing.  The right thing to do is to report it not to make
underhanded comments about rethinking your strategy.

Ideally I would like to visit the machine while it's in that state or
otherwise I would like you to get a gcore while it's int that state,
open it in gdb and produce a "thread apply all bt" trace in text
format and attach it to the JIRA.




On Tue, Oct 5, 2010 at 9:55 AM, Fran?ois Delawarde
<fdelawarde at wirelessmundi.com> wrote:
> I wouldn't jump to conclusions that fast, it's just a bug with current
> git head, probably only happening in very specific cases.
>
> If I learned anything from hollywood movies, it would be that bugs or
> bad people always end up being tracked down and destroyed.
>
> Fran?ois.
>
>
> On Tue, 2010-10-05 at 20:35 +0600, Nyamul Hassan wrote:
>> We learned from another thread about how Sangoma regularly tests 32 E1
>> setups in the lab, and were preparing a concept paper to show a
>> telecom that traditional vendors can be replaced using open source
>> even for large scale implementations. Does this mean we should rethink
>> our strategy?
>>
>> Regards
>> HASSAN
>>
>>
>> On 2010-10-05, Moises Silva <moises.silva at gmail.com> wrote:
>> > On Tue, Oct 5, 2010 at 5:45 AM, Fran?ois Delawarde
>> > <fdelawarde at wirelessmundi.com> wrote:
>> >> I maintain "only" 100 sessions at all times, with a duration of 20-30s,
>> >> and no more than 10cps, so it is NOT a stress test, come on!
>> >>
>> >> I mean this would not even load an Asterisk server, and loading the
>> >> server was not an objective, just to maintain low-medium load over a
>> >> period of time. It worked very well during 5h.
>> >>
>> >> Why would I need to ulimit the crap out of my server then? :-)
>> >
>> > Don't feel alone, you're not the only one.
>> >
>> > We're getting the same pair of messages. We know what it means, we
>> > have just not found the cause of it. In our case happens quite fast
>> > (about an hour IIRC) when doing stress testing of about 900 calls at a
>> > time lasting 10 seconds each at a rate of about 40cps.
>> >
>> > Moises Silva
>> > Senior Software Engineer
>> > Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
>> > L3R 9R6 Canada
>> > t. 1 905 474 1990 x128 | e. moy at sangoma.com
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From anthony.minessale at gmail.com  Tue Oct  5 08:28:13 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Oct 2010 10:28:13 -0500
Subject: [Freeswitch-users] FreeSwitch Out of threads!
In-Reply-To: <AANLkTinhiuJUQA26WevUPXiOtMHb=5bXKTR3N2dv3vrS@mail.gmail.com>
References: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
	<AANLkTing67qXyj3msDQaxun2XOgocRDVYXvPh9hsZWFW@mail.gmail.com>
	<AANLkTimYA=XM4pPJRrUOA_er5WkhZfqhJQO0E4E6LKgj@mail.gmail.com>
	<1286271927.5790.33.camel@luna.tc.commsmundi.com>
	<AANLkTi=Gis9k=pHQ1si5cXeOCjk6EskjN0ri3Nak9YVE@mail.gmail.com>
	<AANLkTingwTJpYLtXChhGyzw6jPLq2ot_82bdptmRZR18@mail.gmail.com>
	<1286290551.5790.58.camel@luna.tc.commsmundi.com>
	<AANLkTinhiuJUQA26WevUPXiOtMHb=5bXKTR3N2dv3vrS@mail.gmail.com>
Message-ID: <AANLkTimr9mqzT6UUULWHvim5bWGUwLakYs_1=J8tHPFX@mail.gmail.com>

BTW,

If you are any less than this revision don't bother to report it until
you have tested latest GIT

commit b02c69bba9f46cad37225a0986cae068f24dcd81
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Wed Sep 22 18:14:24 2010 -0500



On Tue, Oct 5, 2010 at 10:26 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> If there is a bug why are you not opening it on JIRA?
> http://jira.freeswitch.org
>
> We are doing work on improving the event system to be faster and use
> less memory.
>
> If it caused some problems, that is not uncommon, software development
> requires testing. ?The right thing to do is to report it not to make
> underhanded comments about rethinking your strategy.
>
> Ideally I would like to visit the machine while it's in that state or
> otherwise I would like you to get a gcore while it's int that state,
> open it in gdb and produce a "thread apply all bt" trace in text
> format and attach it to the JIRA.
>
>
>
>
> On Tue, Oct 5, 2010 at 9:55 AM, Fran?ois Delawarde
> <fdelawarde at wirelessmundi.com> wrote:
>> I wouldn't jump to conclusions that fast, it's just a bug with current
>> git head, probably only happening in very specific cases.
>>
>> If I learned anything from hollywood movies, it would be that bugs or
>> bad people always end up being tracked down and destroyed.
>>
>> Fran?ois.
>>
>>
>> On Tue, 2010-10-05 at 20:35 +0600, Nyamul Hassan wrote:
>>> We learned from another thread about how Sangoma regularly tests 32 E1
>>> setups in the lab, and were preparing a concept paper to show a
>>> telecom that traditional vendors can be replaced using open source
>>> even for large scale implementations. Does this mean we should rethink
>>> our strategy?
>>>
>>> Regards
>>> HASSAN
>>>
>>>
>>> On 2010-10-05, Moises Silva <moises.silva at gmail.com> wrote:
>>> > On Tue, Oct 5, 2010 at 5:45 AM, Fran?ois Delawarde
>>> > <fdelawarde at wirelessmundi.com> wrote:
>>> >> I maintain "only" 100 sessions at all times, with a duration of 20-30s,
>>> >> and no more than 10cps, so it is NOT a stress test, come on!
>>> >>
>>> >> I mean this would not even load an Asterisk server, and loading the
>>> >> server was not an objective, just to maintain low-medium load over a
>>> >> period of time. It worked very well during 5h.
>>> >>
>>> >> Why would I need to ulimit the crap out of my server then? :-)
>>> >
>>> > Don't feel alone, you're not the only one.
>>> >
>>> > We're getting the same pair of messages. We know what it means, we
>>> > have just not found the cause of it. In our case happens quite fast
>>> > (about an hour IIRC) when doing stress testing of about 900 calls at a
>>> > time lasting 10 seconds each at a rate of about 40cps.
>>> >
>>> > Moises Silva
>>> > Senior Software Engineer
>>> > Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
>>> > L3R 9R6 Canada
>>> > t. 1 905 474 1990 x128 | e. moy at sangoma.com
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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
>



-- 
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


From shamun.toha at gmail.com  Tue Oct  5 08:42:00 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Tue, 5 Oct 2010 17:42:00 +0200
Subject: [Freeswitch-users] mod_h323 - worst audio quality compare to other
	h323
Message-ID: <AANLkTim0JR1Cghq_7m0E686DtRwEfBDShFHieuGTRgcb@mail.gmail.com>

Dear FS H323 author and experts,

With all my respect, I never saw till today H323 audio quality can be that
much bad compared to my past years audio experience with quintum/cisco
gateways.


Info:
===
After spending couple of days, reading the source and and make file sources,
i am now able to install mod_h323 in fedora/centos (32 bit or even 64 bit
system). But when it has to face real world calls scenario, it does not
really makes any sense the level of audio quality i am listening now.

- Non stop 3/4 hours i just played mod_h323 and British telecom premium call
termination route. (honestly mod_h323 really have some problem, somewhere)
- I just planning to switch back to SIP :o(


Highest priority, for mod_h323 Audio problem:
===================================

1. mod_h323 cant handle IVR audio/sound to other legs

2. When call is bridged with SIP user and H323 user ( both parties cant
communicate because there is drop of audio, latency is very low, very bad
audio quality)

3. H323 to H323 calls, have same audio issue, i just tried to tune to some
point, but i just cant make it work!


Less priority, Freeswitch instance problem, while using mod_h323:
===================================================

1. When mod_h323 is loaded, and i want to shutdown freeSwitch instance, i
cant, FS gets freezed.

2. I need to reboot server either physically or init 6 for every restart of
FS instance.

3. Risk of segmentation fault



Many thanks & Best regards
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20101005/6d528912/attachment.html 

From fdelawarde at wirelessmundi.com  Tue Oct  5 08:44:46 2010
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Tue, 05 Oct 2010 17:44:46 +0200
Subject: [Freeswitch-users] FreeSwitch Out of threads!
In-Reply-To: <AANLkTinhiuJUQA26WevUPXiOtMHb=5bXKTR3N2dv3vrS@mail.gmail.com>
References: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
	<AANLkTing67qXyj3msDQaxun2XOgocRDVYXvPh9hsZWFW@mail.gmail.com>
	<AANLkTimYA=XM4pPJRrUOA_er5WkhZfqhJQO0E4E6LKgj@mail.gmail.com>
	<1286271927.5790.33.camel@luna.tc.commsmundi.com>
	<AANLkTi=Gis9k=pHQ1si5cXeOCjk6EskjN0ri3Nak9YVE@mail.gmail.com>
	<AANLkTingwTJpYLtXChhGyzw6jPLq2ot_82bdptmRZR18@mail.gmail.com>
	<1286290551.5790.58.camel@luna.tc.commsmundi.com>
	<AANLkTinhiuJUQA26WevUPXiOtMHb=5bXKTR3N2dv3vrS@mail.gmail.com>
Message-ID: <1286293486.5790.69.camel@luna.tc.commsmundi.com>

Ok, I just wanted to ask before opening any JIRA issues, it could have
been some hidden config I missed. I will try to reproduce, learn what a
gcore is and how to extract it.

Moises, is there a Jira already open for this issue?

Fran?ois.


On Tue, 2010-10-05 at 10:26 -0500, Anthony Minessale wrote:
> If there is a bug why are you not opening it on JIRA?
> http://jira.freeswitch.org
> 
> We are doing work on improving the event system to be faster and use
> less memory.
> 
> If it caused some problems, that is not uncommon, software development
> requires testing.  The right thing to do is to report it not to make
> underhanded comments about rethinking your strategy.
> 
> Ideally I would like to visit the machine while it's in that state or
> otherwise I would like you to get a gcore while it's int that state,
> open it in gdb and produce a "thread apply all bt" trace in text
> format and attach it to the JIRA.
> 
> 
> 
> 
> On Tue, Oct 5, 2010 at 9:55 AM, Fran?ois Delawarde
> <fdelawarde at wirelessmundi.com> wrote:
> > I wouldn't jump to conclusions that fast, it's just a bug with current
> > git head, probably only happening in very specific cases.
> >
> > If I learned anything from hollywood movies, it would be that bugs or
> > bad people always end up being tracked down and destroyed.
> >
> > Fran?ois.
> >
> >
> > On Tue, 2010-10-05 at 20:35 +0600, Nyamul Hassan wrote:
> >> We learned from another thread about how Sangoma regularly tests 32 E1
> >> setups in the lab, and were preparing a concept paper to show a
> >> telecom that traditional vendors can be replaced using open source
> >> even for large scale implementations. Does this mean we should rethink
> >> our strategy?
> >>
> >> Regards
> >> HASSAN
> >>
> >>
> >> On 2010-10-05, Moises Silva <moises.silva at gmail.com> wrote:
> >> > On Tue, Oct 5, 2010 at 5:45 AM, Fran?ois Delawarde
> >> > <fdelawarde at wirelessmundi.com> wrote:
> >> >> I maintain "only" 100 sessions at all times, with a duration of 20-30s,
> >> >> and no more than 10cps, so it is NOT a stress test, come on!
> >> >>
> >> >> I mean this would not even load an Asterisk server, and loading the
> >> >> server was not an objective, just to maintain low-medium load over a
> >> >> period of time. It worked very well during 5h.
> >> >>
> >> >> Why would I need to ulimit the crap out of my server then? :-)
> >> >
> >> > Don't feel alone, you're not the only one.
> >> >
> >> > We're getting the same pair of messages. We know what it means, we
> >> > have just not found the cause of it. In our case happens quite fast
> >> > (about an hour IIRC) when doing stress testing of about 900 calls at a
> >> > time lasting 10 seconds each at a rate of about 40cps.
> >> >
> >> > Moises Silva
> >> > Senior Software Engineer
> >> > Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
> >> > L3R 9R6 Canada
> >> > t. 1 905 474 1990 x128 | e. moy at sangoma.com
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> 



From fdelawarde at wirelessmundi.com  Tue Oct  5 08:45:26 2010
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Tue, 05 Oct 2010 17:45:26 +0200
Subject: [Freeswitch-users] FreeSwitch Out of threads!
In-Reply-To: <AANLkTimr9mqzT6UUULWHvim5bWGUwLakYs_1=J8tHPFX@mail.gmail.com>
References: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
	<AANLkTing67qXyj3msDQaxun2XOgocRDVYXvPh9hsZWFW@mail.gmail.com>
	<AANLkTimYA=XM4pPJRrUOA_er5WkhZfqhJQO0E4E6LKgj@mail.gmail.com>
	<1286271927.5790.33.camel@luna.tc.commsmundi.com>
	<AANLkTi=Gis9k=pHQ1si5cXeOCjk6EskjN0ri3Nak9YVE@mail.gmail.com>
	<AANLkTingwTJpYLtXChhGyzw6jPLq2ot_82bdptmRZR18@mail.gmail.com>
	<1286290551.5790.58.camel@luna.tc.commsmundi.com>
	<AANLkTinhiuJUQA26WevUPXiOtMHb=5bXKTR3N2dv3vrS@mail.gmail.com>
	<AANLkTimr9mqzT6UUULWHvim5bWGUwLakYs_1=J8tHPFX@mail.gmail.com>
Message-ID: <1286293526.5790.71.camel@luna.tc.commsmundi.com>

GIT from 1st of October.

Fran?ois.

On Tue, 2010-10-05 at 10:28 -0500, Anthony Minessale wrote:
> BTW,
> 
> If you are any less than this revision don't bother to report it until
> you have tested latest GIT
> 
> commit b02c69bba9f46cad37225a0986cae068f24dcd81
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date:   Wed Sep 22 18:14:24 2010 -0500
> 
> 
> 
> On Tue, Oct 5, 2010 at 10:26 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > If there is a bug why are you not opening it on JIRA?
> > http://jira.freeswitch.org
> >
> > We are doing work on improving the event system to be faster and use
> > less memory.
> >
> > If it caused some problems, that is not uncommon, software development
> > requires testing.  The right thing to do is to report it not to make
> > underhanded comments about rethinking your strategy.
> >
> > Ideally I would like to visit the machine while it's in that state or
> > otherwise I would like you to get a gcore while it's int that state,
> > open it in gdb and produce a "thread apply all bt" trace in text
> > format and attach it to the JIRA.
> >
> >
> >
> >
> > On Tue, Oct 5, 2010 at 9:55 AM, Fran?ois Delawarde
> > <fdelawarde at wirelessmundi.com> wrote:
> >> I wouldn't jump to conclusions that fast, it's just a bug with current
> >> git head, probably only happening in very specific cases.
> >>
> >> If I learned anything from hollywood movies, it would be that bugs or
> >> bad people always end up being tracked down and destroyed.
> >>
> >> Fran?ois.
> >>
> >>
> >> On Tue, 2010-10-05 at 20:35 +0600, Nyamul Hassan wrote:
> >>> We learned from another thread about how Sangoma regularly tests 32 E1
> >>> setups in the lab, and were preparing a concept paper to show a
> >>> telecom that traditional vendors can be replaced using open source
> >>> even for large scale implementations. Does this mean we should rethink
> >>> our strategy?
> >>>
> >>> Regards
> >>> HASSAN
> >>>
> >>>
> >>> On 2010-10-05, Moises Silva <moises.silva at gmail.com> wrote:
> >>> > On Tue, Oct 5, 2010 at 5:45 AM, Fran?ois Delawarde
> >>> > <fdelawarde at wirelessmundi.com> wrote:
> >>> >> I maintain "only" 100 sessions at all times, with a duration of 20-30s,
> >>> >> and no more than 10cps, so it is NOT a stress test, come on!
> >>> >>
> >>> >> I mean this would not even load an Asterisk server, and loading the
> >>> >> server was not an objective, just to maintain low-medium load over a
> >>> >> period of time. It worked very well during 5h.
> >>> >>
> >>> >> Why would I need to ulimit the crap out of my server then? :-)
> >>> >
> >>> > Don't feel alone, you're not the only one.
> >>> >
> >>> > We're getting the same pair of messages. We know what it means, we
> >>> > have just not found the cause of it. In our case happens quite fast
> >>> > (about an hour IIRC) when doing stress testing of about 900 calls at a
> >>> > time lasting 10 seconds each at a rate of about 40cps.
> >>> >
> >>> > Moises Silva
> >>> > Senior Software Engineer
> >>> > Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
> >>> > L3R 9R6 Canada
> >>> > t. 1 905 474 1990 x128 | e. moy at sangoma.com
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> 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
> >
> 
> 
> 
> -- 
> 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
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Tue Oct  5 08:59:27 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Oct 2010 10:59:27 -0500
Subject: [Freeswitch-users] FreeSwitch Out of threads!
In-Reply-To: <1286293526.5790.71.camel@luna.tc.commsmundi.com>
References: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
	<AANLkTing67qXyj3msDQaxun2XOgocRDVYXvPh9hsZWFW@mail.gmail.com>
	<AANLkTimYA=XM4pPJRrUOA_er5WkhZfqhJQO0E4E6LKgj@mail.gmail.com>
	<1286271927.5790.33.camel@luna.tc.commsmundi.com>
	<AANLkTi=Gis9k=pHQ1si5cXeOCjk6EskjN0ri3Nak9YVE@mail.gmail.com>
	<AANLkTingwTJpYLtXChhGyzw6jPLq2ot_82bdptmRZR18@mail.gmail.com>
	<1286290551.5790.58.camel@luna.tc.commsmundi.com>
	<AANLkTinhiuJUQA26WevUPXiOtMHb=5bXKTR3N2dv3vrS@mail.gmail.com>
	<AANLkTimr9mqzT6UUULWHvim5bWGUwLakYs_1=J8tHPFX@mail.gmail.com>
	<1286293526.5790.71.camel@luna.tc.commsmundi.com>
Message-ID: <AANLkTi=hTTkfXY_o4ZKNHt21x4EJvq-bBXCU2YzkeyFc@mail.gmail.com>

if you are on centos or something like it.
you can run this from the FS build root.

./support-d/fscore_pb gcore <your nick>

while FS is running in this state.

That will give you a URL you can tell me, it will upload the trace I
need to see right to our pastebin.

If you can get it that way, and add our ssh key from
./support-d/shinzon.pub, install it and email or irc pm me the
hostname.  And I can examine it.


On Tue, Oct 5, 2010 at 10:45 AM, Fran?ois Delawarde
<fdelawarde at wirelessmundi.com> wrote:
> GIT from 1st of October.
>
> Fran?ois.
>
> On Tue, 2010-10-05 at 10:28 -0500, Anthony Minessale wrote:
>> BTW,
>>
>> If you are any less than this revision don't bother to report it until
>> you have tested latest GIT
>>
>> commit b02c69bba9f46cad37225a0986cae068f24dcd81
>> Author: Anthony Minessale <anthm at freeswitch.org>
>> Date: ? Wed Sep 22 18:14:24 2010 -0500
>>
>>
>>
>> On Tue, Oct 5, 2010 at 10:26 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > If there is a bug why are you not opening it on JIRA?
>> > http://jira.freeswitch.org
>> >
>> > We are doing work on improving the event system to be faster and use
>> > less memory.
>> >
>> > If it caused some problems, that is not uncommon, software development
>> > requires testing. ?The right thing to do is to report it not to make
>> > underhanded comments about rethinking your strategy.
>> >
>> > Ideally I would like to visit the machine while it's in that state or
>> > otherwise I would like you to get a gcore while it's int that state,
>> > open it in gdb and produce a "thread apply all bt" trace in text
>> > format and attach it to the JIRA.
>> >
>> >
>> >
>> >
>> > On Tue, Oct 5, 2010 at 9:55 AM, Fran?ois Delawarde
>> > <fdelawarde at wirelessmundi.com> wrote:
>> >> I wouldn't jump to conclusions that fast, it's just a bug with current
>> >> git head, probably only happening in very specific cases.
>> >>
>> >> If I learned anything from hollywood movies, it would be that bugs or
>> >> bad people always end up being tracked down and destroyed.
>> >>
>> >> Fran?ois.
>> >>
>> >>
>> >> On Tue, 2010-10-05 at 20:35 +0600, Nyamul Hassan wrote:
>> >>> We learned from another thread about how Sangoma regularly tests 32 E1
>> >>> setups in the lab, and were preparing a concept paper to show a
>> >>> telecom that traditional vendors can be replaced using open source
>> >>> even for large scale implementations. Does this mean we should rethink
>> >>> our strategy?
>> >>>
>> >>> Regards
>> >>> HASSAN
>> >>>
>> >>>
>> >>> On 2010-10-05, Moises Silva <moises.silva at gmail.com> wrote:
>> >>> > On Tue, Oct 5, 2010 at 5:45 AM, Fran?ois Delawarde
>> >>> > <fdelawarde at wirelessmundi.com> wrote:
>> >>> >> I maintain "only" 100 sessions at all times, with a duration of 20-30s,
>> >>> >> and no more than 10cps, so it is NOT a stress test, come on!
>> >>> >>
>> >>> >> I mean this would not even load an Asterisk server, and loading the
>> >>> >> server was not an objective, just to maintain low-medium load over a
>> >>> >> period of time. It worked very well during 5h.
>> >>> >>
>> >>> >> Why would I need to ulimit the crap out of my server then? :-)
>> >>> >
>> >>> > Don't feel alone, you're not the only one.
>> >>> >
>> >>> > We're getting the same pair of messages. We know what it means, we
>> >>> > have just not found the cause of it. In our case happens quite fast
>> >>> > (about an hour IIRC) when doing stress testing of about 900 calls at a
>> >>> > time lasting 10 seconds each at a rate of about 40cps.
>> >>> >
>> >>> > Moises Silva
>> >>> > Senior Software Engineer
>> >>> > Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
>> >>> > L3R 9R6 Canada
>> >>> > t. 1 905 474 1990 x128 | e. moy at sangoma.com
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> 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
>> >
>>
>>
>>
>> --
>> 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From bottleman at icf.org.ru  Tue Oct  5 09:03:23 2010
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Tue, 5 Oct 2010 20:03:23 +0400 (MSD)
Subject: [Freeswitch-users] mod_h323 - worst audio quality compare to
 other h323
In-Reply-To: <AANLkTim0JR1Cghq_7m0E686DtRwEfBDShFHieuGTRgcb@mail.gmail.com>
References: <AANLkTim0JR1Cghq_7m0E686DtRwEfBDShFHieuGTRgcb@mail.gmail.com>
Message-ID: <alpine.DEB.2.00.1010052002410.18429@icf.org.ru>

On 2010-10-05 17:42 +0200, Shamun toha md wrote FreeSWITCH Users Help:

Stm>Dear FS H323 author and experts,
Stm>
Stm>With all my respect, I never saw till today H323 audio quality can be that
Stm>much bad compared to my past years audio experience with quintum/cisco
Stm>gateways.
Stm>
Stm>
Stm>Info:
Stm>===
Stm>After spending couple of days, reading the source and and make file sources,
Stm>i am now able to install mod_h323 in fedora/centos (32 bit or even 64 bit
Stm>system). But when it has to face real world calls scenario, it does not
Stm>really makes any sense the level of audio quality i am listening now.
Stm>
Stm>- Non stop 3/4 hours i just played mod_h323 and British telecom premium call
Stm>termination route. (honestly mod_h323 really have some problem, somewhere)
Stm>- I just planning to switch back to SIP :o(
Stm>
Stm>
Stm>Highest priority, for mod_h323 Audio problem:
Stm>===================================

enable late negotiation and put fs in transconding mode.

Stm>1. mod_h323 cant handle IVR audio/sound to other legs
Stm>
Stm>2. When call is bridged with SIP user and H323 user ( both parties cant
Stm>communicate because there is drop of audio, latency is very low, very bad
Stm>audio quality)
Stm>
Stm>3. H323 to H323 calls, have same audio issue, i just tried to tune to some
Stm>point, but i just cant make it work!
Stm>
Stm>
Stm>Less priority, Freeswitch instance problem, while using mod_h323:
Stm>===================================================
Stm>
Stm>1. When mod_h323 is loaded, and i want to shutdown freeSwitch instance, i
Stm>cant, FS gets freezed.
Stm>
Stm>2. I need to reboot server either physically or init 6 for every restart of
Stm>FS instance.
Stm>
Stm>3. Risk of segmentation fault
Stm>
Stm>
Stm>
Stm>Many thanks & Best regards
Stm>

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From mario_fs at mgtech.com  Tue Oct  5 09:15:53 2010
From: mario_fs at mgtech.com (Mario)
Date: Tue, 05 Oct 2010 09:15:53 -0700
Subject: [Freeswitch-users] latest git pull has error, can't install
In-Reply-To: <4CAB4393.4010900@mgtech.com>
References: <4CAB4393.4010900@mgtech.com>
Message-ID: <4CAB4F39.8050705@mgtech.com>

I tried executing the 2 lines below but still get the same error. I
quess a full install is in order? I am puzzled why git pull && make
current only worked one time.
automake-1.11 --foreign
automake --add-missing


On 10/05/10 08:26, Mario wrote:
> While working on a loop in FS I did a git pull and make current. I had
> done this before with no problem but today I got the error below. Any
> ideas? Thanks. Mario
> 
> 
> cd libs/sofia-sip && make clean
> make[1]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip'
>  cd . && /bin/sh /usr/local/src/freeswitch/libs/sofia-sip/missing --run
> automake-1.11 --foreign
> configure.ac:47: warning: AC_CACHE_VAL(ac_cc_environment, ...):
> suspicious cache-id, must contain _cv_ to be cached
> ../../lib/autoconf/general.m4:1998: AC_CACHE_VAL is expanded from...
> ../../lib/autoconf/general.m4:2019: AC_CACHE_CHECK is expanded from...
> m4/sac-general.m4:351: AC_CHECK_COMPILATION_ENVIRONMENT is expanded from...
> configure.ac:47: the top level
> configure.ac:49: warning: AC_PROG_CPP was called before SAC_TOOL_CC
> m4/sac-general.m4:90: SAC_TOOL_CC is expanded from...
> configure.ac:49: the top level
> configure.ac:56: required file `./compile' not found
> configure.ac:56:   `automake --add-missing' can install `compile'
> make[1]: *** [Makefile.in] Error 1
> make[1]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip'
> make: *** [update-clean] Error 2


From anthony.minessale at gmail.com  Tue Oct  5 09:17:32 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Oct 2010 11:17:32 -0500
Subject: [Freeswitch-users] FreeSwitch Out of threads!
In-Reply-To: <AANLkTi=hTTkfXY_o4ZKNHt21x4EJvq-bBXCU2YzkeyFc@mail.gmail.com>
References: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
	<AANLkTing67qXyj3msDQaxun2XOgocRDVYXvPh9hsZWFW@mail.gmail.com>
	<AANLkTimYA=XM4pPJRrUOA_er5WkhZfqhJQO0E4E6LKgj@mail.gmail.com>
	<1286271927.5790.33.camel@luna.tc.commsmundi.com>
	<AANLkTi=Gis9k=pHQ1si5cXeOCjk6EskjN0ri3Nak9YVE@mail.gmail.com>
	<AANLkTingwTJpYLtXChhGyzw6jPLq2ot_82bdptmRZR18@mail.gmail.com>
	<1286290551.5790.58.camel@luna.tc.commsmundi.com>
	<AANLkTinhiuJUQA26WevUPXiOtMHb=5bXKTR3N2dv3vrS@mail.gmail.com>
	<AANLkTimr9mqzT6UUULWHvim5bWGUwLakYs_1=J8tHPFX@mail.gmail.com>
	<1286293526.5790.71.camel@luna.tc.commsmundi.com>
	<AANLkTi=hTTkfXY_o4ZKNHt21x4EJvq-bBXCU2YzkeyFc@mail.gmail.com>
Message-ID: <AANLkTino=xF7HAMXqaKmubswR26ytU4sNkGWB4Ch8ngz@mail.gmail.com>

Can you update first to this revision of HEAD or later?

commit 80883ebeb81c73a3dc5c4ee65050d57e2c6ccf00
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Tue Oct 5 11:11:21 2010 -0500

I added a small patch that I think may be related to your problem.


On Tue, Oct 5, 2010 at 10:59 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> if you are on centos or something like it.
> you can run this from the FS build root.
>
> ./support-d/fscore_pb gcore <your nick>
>
> while FS is running in this state.
>
> That will give you a URL you can tell me, it will upload the trace I
> need to see right to our pastebin.
>
> If you can get it that way, and add our ssh key from
> ./support-d/shinzon.pub, install it and email or irc pm me the
> hostname. ?And I can examine it.
>
>
> On Tue, Oct 5, 2010 at 10:45 AM, Fran?ois Delawarde
> <fdelawarde at wirelessmundi.com> wrote:
>> GIT from 1st of October.
>>
>> Fran?ois.
>>
>> On Tue, 2010-10-05 at 10:28 -0500, Anthony Minessale wrote:
>>> BTW,
>>>
>>> If you are any less than this revision don't bother to report it until
>>> you have tested latest GIT
>>>
>>> commit b02c69bba9f46cad37225a0986cae068f24dcd81
>>> Author: Anthony Minessale <anthm at freeswitch.org>
>>> Date: ? Wed Sep 22 18:14:24 2010 -0500
>>>
>>>
>>>
>>> On Tue, Oct 5, 2010 at 10:26 AM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>> > If there is a bug why are you not opening it on JIRA?
>>> > http://jira.freeswitch.org
>>> >
>>> > We are doing work on improving the event system to be faster and use
>>> > less memory.
>>> >
>>> > If it caused some problems, that is not uncommon, software development
>>> > requires testing. ?The right thing to do is to report it not to make
>>> > underhanded comments about rethinking your strategy.
>>> >
>>> > Ideally I would like to visit the machine while it's in that state or
>>> > otherwise I would like you to get a gcore while it's int that state,
>>> > open it in gdb and produce a "thread apply all bt" trace in text
>>> > format and attach it to the JIRA.
>>> >
>>> >
>>> >
>>> >
>>> > On Tue, Oct 5, 2010 at 9:55 AM, Fran?ois Delawarde
>>> > <fdelawarde at wirelessmundi.com> wrote:
>>> >> I wouldn't jump to conclusions that fast, it's just a bug with current
>>> >> git head, probably only happening in very specific cases.
>>> >>
>>> >> If I learned anything from hollywood movies, it would be that bugs or
>>> >> bad people always end up being tracked down and destroyed.
>>> >>
>>> >> Fran?ois.
>>> >>
>>> >>
>>> >> On Tue, 2010-10-05 at 20:35 +0600, Nyamul Hassan wrote:
>>> >>> We learned from another thread about how Sangoma regularly tests 32 E1
>>> >>> setups in the lab, and were preparing a concept paper to show a
>>> >>> telecom that traditional vendors can be replaced using open source
>>> >>> even for large scale implementations. Does this mean we should rethink
>>> >>> our strategy?
>>> >>>
>>> >>> Regards
>>> >>> HASSAN
>>> >>>
>>> >>>
>>> >>> On 2010-10-05, Moises Silva <moises.silva at gmail.com> wrote:
>>> >>> > On Tue, Oct 5, 2010 at 5:45 AM, Fran?ois Delawarde
>>> >>> > <fdelawarde at wirelessmundi.com> wrote:
>>> >>> >> I maintain "only" 100 sessions at all times, with a duration of 20-30s,
>>> >>> >> and no more than 10cps, so it is NOT a stress test, come on!
>>> >>> >>
>>> >>> >> I mean this would not even load an Asterisk server, and loading the
>>> >>> >> server was not an objective, just to maintain low-medium load over a
>>> >>> >> period of time. It worked very well during 5h.
>>> >>> >>
>>> >>> >> Why would I need to ulimit the crap out of my server then? :-)
>>> >>> >
>>> >>> > Don't feel alone, you're not the only one.
>>> >>> >
>>> >>> > We're getting the same pair of messages. We know what it means, we
>>> >>> > have just not found the cause of it. In our case happens quite fast
>>> >>> > (about an hour IIRC) when doing stress testing of about 900 calls at a
>>> >>> > time lasting 10 seconds each at a rate of about 40cps.
>>> >>> >
>>> >>> > Moises Silva
>>> >>> > Senior Software Engineer
>>> >>> > Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
>>> >>> > L3R 9R6 Canada
>>> >>> > t. 1 905 474 1990 x128 | e. moy at sangoma.com
>>> >>> >
>>> >>> > _______________________________________________
>>> >>> > FreeSWITCH-users mailing list
>>> >>> > FreeSWITCH-users at lists.freeswitch.org
>>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> > http://www.freeswitch.org
>>> >>> >
>>> >>>
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> 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
>>> >
>>>
>>>
>>>
>>> --
>>> 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
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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
>



-- 
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


From fdelawarde at wirelessmundi.com  Tue Oct  5 09:50:51 2010
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Tue, 05 Oct 2010 18:50:51 +0200
Subject: [Freeswitch-users] FreeSwitch Out of threads!
In-Reply-To: <AANLkTino=xF7HAMXqaKmubswR26ytU4sNkGWB4Ch8ngz@mail.gmail.com>
References: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
	<AANLkTing67qXyj3msDQaxun2XOgocRDVYXvPh9hsZWFW@mail.gmail.com>
	<AANLkTimYA=XM4pPJRrUOA_er5WkhZfqhJQO0E4E6LKgj@mail.gmail.com>
	<1286271927.5790.33.camel@luna.tc.commsmundi.com>
	<AANLkTi=Gis9k=pHQ1si5cXeOCjk6EskjN0ri3Nak9YVE@mail.gmail.com>
	<AANLkTingwTJpYLtXChhGyzw6jPLq2ot_82bdptmRZR18@mail.gmail.com>
	<1286290551.5790.58.camel@luna.tc.commsmundi.com>
	<AANLkTinhiuJUQA26WevUPXiOtMHb=5bXKTR3N2dv3vrS@mail.gmail.com>
	<AANLkTimr9mqzT6UUULWHvim5bWGUwLakYs_1=J8tHPFX@mail.gmail.com>
	<1286293526.5790.71.camel@luna.tc.commsmundi.com>
	<AANLkTi=hTTkfXY_o4ZKNHt21x4EJvq-bBXCU2YzkeyFc@mail.gmail.com>
	<AANLkTino=xF7HAMXqaKmubswR26ytU4sNkGWB4Ch8ngz@mail.gmail.com>
Message-ID: <1286297451.5850.0.camel@luna.tc.commsmundi.com>

I will certainly tomorrow first thing!

Thanks,
Fran?ois.

On Tue, 2010-10-05 at 11:17 -0500, Anthony Minessale wrote:
> Can you update first to this revision of HEAD or later?
> 
> commit 80883ebeb81c73a3dc5c4ee65050d57e2c6ccf00
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date:   Tue Oct 5 11:11:21 2010 -0500
> 
> I added a small patch that I think may be related to your problem.
> 
> 
> On Tue, Oct 5, 2010 at 10:59 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > if you are on centos or something like it.
> > you can run this from the FS build root.
> >
> > ./support-d/fscore_pb gcore <your nick>
> >
> > while FS is running in this state.
> >
> > That will give you a URL you can tell me, it will upload the trace I
> > need to see right to our pastebin.
> >
> > If you can get it that way, and add our ssh key from
> > ./support-d/shinzon.pub, install it and email or irc pm me the
> > hostname.  And I can examine it.
> >
> >
> > On Tue, Oct 5, 2010 at 10:45 AM, Fran?ois Delawarde
> > <fdelawarde at wirelessmundi.com> wrote:
> >> GIT from 1st of October.
> >>
> >> Fran?ois.
> >>
> >> On Tue, 2010-10-05 at 10:28 -0500, Anthony Minessale wrote:
> >>> BTW,
> >>>
> >>> If you are any less than this revision don't bother to report it until
> >>> you have tested latest GIT
> >>>
> >>> commit b02c69bba9f46cad37225a0986cae068f24dcd81
> >>> Author: Anthony Minessale <anthm at freeswitch.org>
> >>> Date:   Wed Sep 22 18:14:24 2010 -0500
> >>>
> >>>
> >>>
> >>> On Tue, Oct 5, 2010 at 10:26 AM, Anthony Minessale
> >>> <anthony.minessale at gmail.com> wrote:
> >>> > If there is a bug why are you not opening it on JIRA?
> >>> > http://jira.freeswitch.org
> >>> >
> >>> > We are doing work on improving the event system to be faster and use
> >>> > less memory.
> >>> >
> >>> > If it caused some problems, that is not uncommon, software development
> >>> > requires testing.  The right thing to do is to report it not to make
> >>> > underhanded comments about rethinking your strategy.
> >>> >
> >>> > Ideally I would like to visit the machine while it's in that state or
> >>> > otherwise I would like you to get a gcore while it's int that state,
> >>> > open it in gdb and produce a "thread apply all bt" trace in text
> >>> > format and attach it to the JIRA.
> >>> >
> >>> >
> >>> >
> >>> >
> >>> > On Tue, Oct 5, 2010 at 9:55 AM, Fran?ois Delawarde
> >>> > <fdelawarde at wirelessmundi.com> wrote:
> >>> >> I wouldn't jump to conclusions that fast, it's just a bug with current
> >>> >> git head, probably only happening in very specific cases.
> >>> >>
> >>> >> If I learned anything from hollywood movies, it would be that bugs or
> >>> >> bad people always end up being tracked down and destroyed.
> >>> >>
> >>> >> Fran?ois.
> >>> >>
> >>> >>
> >>> >> On Tue, 2010-10-05 at 20:35 +0600, Nyamul Hassan wrote:
> >>> >>> We learned from another thread about how Sangoma regularly tests 32 E1
> >>> >>> setups in the lab, and were preparing a concept paper to show a
> >>> >>> telecom that traditional vendors can be replaced using open source
> >>> >>> even for large scale implementations. Does this mean we should rethink
> >>> >>> our strategy?
> >>> >>>
> >>> >>> Regards
> >>> >>> HASSAN
> >>> >>>
> >>> >>>
> >>> >>> On 2010-10-05, Moises Silva <moises.silva at gmail.com> wrote:
> >>> >>> > On Tue, Oct 5, 2010 at 5:45 AM, Fran?ois Delawarde
> >>> >>> > <fdelawarde at wirelessmundi.com> wrote:
> >>> >>> >> I maintain "only" 100 sessions at all times, with a duration of 20-30s,
> >>> >>> >> and no more than 10cps, so it is NOT a stress test, come on!
> >>> >>> >>
> >>> >>> >> I mean this would not even load an Asterisk server, and loading the
> >>> >>> >> server was not an objective, just to maintain low-medium load over a
> >>> >>> >> period of time. It worked very well during 5h.
> >>> >>> >>
> >>> >>> >> Why would I need to ulimit the crap out of my server then? :-)
> >>> >>> >
> >>> >>> > Don't feel alone, you're not the only one.
> >>> >>> >
> >>> >>> > We're getting the same pair of messages. We know what it means, we
> >>> >>> > have just not found the cause of it. In our case happens quite fast
> >>> >>> > (about an hour IIRC) when doing stress testing of about 900 calls at a
> >>> >>> > time lasting 10 seconds each at a rate of about 40cps.
> >>> >>> >
> >>> >>> > Moises Silva
> >>> >>> > Senior Software Engineer
> >>> >>> > Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
> >>> >>> > L3R 9R6 Canada
> >>> >>> > t. 1 905 474 1990 x128 | e. moy at sangoma.com
> >>> >>> >
> >>> >>> > _______________________________________________
> >>> >>> > FreeSWITCH-users mailing list
> >>> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >>> > http://www.freeswitch.org
> >>> >>> >
> >>> >>>
> >>> >>
> >>> >>
> >>> >> _______________________________________________
> >>> >> FreeSWITCH-users mailing list
> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> 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
> >>> >
> >>>
> >>>
> >>>
> >>> --
> >>> 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
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> 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
> >
> 
> 
> 



From mario_fs at mgtech.com  Tue Oct  5 10:15:02 2010
From: mario_fs at mgtech.com (Mario)
Date: Tue, 05 Oct 2010 10:15:02 -0700
Subject: [Freeswitch-users] latest git pull has error, can't install
In-Reply-To: <4CAB4F39.8050705@mgtech.com>
References: <4CAB4393.4010900@mgtech.com> <4CAB4F39.8050705@mgtech.com>
Message-ID: <4CAB5D16.8020203@mgtech.com>

In case someone else runs across this I got it to work by:
./bootstrap.sh
./configure
make current

Still would like to know what could have caused this so I can avoid it.

On 10/05/10 09:15, Mario wrote:
> I tried executing the 2 lines below but still get the same error. I
> quess a full install is in order? I am puzzled why git pull && make
> current only worked one time.
> automake-1.11 --foreign
> automake --add-missing
> 
> 
> On 10/05/10 08:26, Mario wrote:
>> While working on a loop in FS I did a git pull and make current. I had
>> done this before with no problem but today I got the error below. Any
>> ideas? Thanks. Mario
>>
>>
>> cd libs/sofia-sip && make clean
>> make[1]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip'
>>  cd . && /bin/sh /usr/local/src/freeswitch/libs/sofia-sip/missing --run
>> automake-1.11 --foreign
>> configure.ac:47: warning: AC_CACHE_VAL(ac_cc_environment, ...):
>> suspicious cache-id, must contain _cv_ to be cached
>> ../../lib/autoconf/general.m4:1998: AC_CACHE_VAL is expanded from...
>> ../../lib/autoconf/general.m4:2019: AC_CACHE_CHECK is expanded from...
>> m4/sac-general.m4:351: AC_CHECK_COMPILATION_ENVIRONMENT is expanded from...
>> configure.ac:47: the top level
>> configure.ac:49: warning: AC_PROG_CPP was called before SAC_TOOL_CC
>> m4/sac-general.m4:90: SAC_TOOL_CC is expanded from...
>> configure.ac:49: the top level
>> configure.ac:56: required file `./compile' not found
>> configure.ac:56:   `automake --add-missing' can install `compile'
>> make[1]: *** [Makefile.in] Error 1
>> make[1]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip'
>> make: *** [update-clean] Error 2


From rupa at rupa.com  Tue Oct  5 10:28:35 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 5 Oct 2010 12:28:35 -0500
Subject: [Freeswitch-users] limit_execute and distributor concurrent
 module usage
In-Reply-To: <1286288787.9945.10.camel@me>
References: <1286288787.9945.10.camel@me>
Message-ID: <AANLkTinff5ugbAZ2BB_CB5b7sbjdpGLbT-YMNn_EvREo@mail.gmail.com>

That won't work 'cause you are limiting on the general TESTGW for realm and
resource.  You need to find out which resource distributor gave you and then
use limit_execute and the actual resource name (GW_001 or GW_002).

As you can see, without changing how distributor works this will be full of
race conditions.

ie: you can get the resource by expanding ${distributor(aa_rr)} and then
using that in limit_execute, but since distributor has no knowledge of
limits it may continue serving up a resource that has reached it's limit.
 And even if distributor knew about limits it would have to reserve the
resource and then *use* the resource right then not pass it back as a
string.  That changes how you use distributor.

Kind of a pain.

A bit more heavy weight, but maybe look at using mod_lcr with limit support?

On Tue, Oct 5, 2010 at 9:26 AM, Frank Ochere <ochere at gmail.com> wrote:

> Hello,
>
> I have the following in distributor.conf.xml
>
>    <list name="aa_rr" total-weight="2">
>      <node name="GW_001" weight="1"/>
>      <node name="GW_002" weight="1"/>
>    </list>
>
>
>
> in dialplan
>
>     <extension name="TESTGW">
>       <condition field="destination_number" expression="^7777(\d+)$">
>         <action application="limit_execute" data="hash TESTGW TESTGW 1
> bridge sofia/gateway/${distributor(aa_rr)}/$1"/>
>       </condition>
>     </extension>
>
> my question is, is it possible to use limit_execute and distributor
> concurrently as in TESTGW above? ie limit calls to 1 per gateway but
> being alternated as specified in list aa_rr
>
>
> sofia debug below
>
> Dialplan: sofia/external/00000000 at 1.1.1.1:5060 Action limit_execute(hash
> TESTGW TESTGW 1 bridge
> sofia/gateway/${distributor(sat_rr)}/00777717171)
> 2010-10-05 17:13:15.616463 [DEBUG] switch_core_state_machine.c:119
> (sofia/external/00000000 at 1.1.1.1:5060) State Change CS_ROUTING ->
> CS_EXECUTE
> 2010-10-05 17:13:15.616463 [DEBUG] switch_core_session.c:1047 Send
> signal sofia/external/00000000 at 1.1.1.1:5060 [BREAK]
> 2010-10-05 17:13:15.616463 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/00000000 at 1.1.1.1:5060) State ROUTING going to sleep
> 2010-10-05 17:13:15.616463 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/00000000 at 1.1.1.1:5060) Running State Change CS_EXECUTE
> 2010-10-05 17:13:15.616463 [DEBUG] switch_core_state_machine.c:348
> (sofia/external/00000000 at 1.1.1.1:5060) State EXECUTE
> 2010-10-05 17:13:15.616463 [DEBUG] mod_sofia.c:239
> sofia/external/00000000 at 1.1.1.1:5060 SOFIA EXECUTE
> 2010-10-05 17:13:15.616463 [DEBUG] switch_core_state_machine.c:157
> sofia/external/00000000 at 1.1.1.1:5060 Standard EXECUTE
> EXECUTE sofia/external/00000000 at 1.1.1.1:5060 limit_execute(hash TESTGW
> TESTGW 1 bridge sofia/gateway/-err/00777717171)
> 2010-10-05 17:13:15.616463 [INFO] switch_limit.c:126 incr called:
> TESTGW_TESTGW max:1, interval:0
> 2010-10-05 17:13:15.616463 [INFO] mod_hash.c:202 Usage for TESTGW_TESTGW
> is now 1/1
> EXECUTE sofia/external/00000000 at 1.1.1.1:5060
> bridge(sofia/gateway/-err/00777717171)
> 2010-10-05 17:13:15.616463 [ERR] mod_sofia.c:3725 Invalid Gateway
> 2010-10-05 17:13:15.616463 [NOTICE] mod_sofia.c:4047 Close Channel N/A
> [CS_NEW]
> 2010-10-05 17:13:15.616463 [DEBUG] switch_core_state_machine.c:430 ()
> Running State Change CS_DESTROY
> 2010-10-05 17:13:15.616463 [DEBUG] switch_core_state_machine.c:440 (N/A)
> State DESTROY
> 2010-10-05 17:13:15.616463 [DEBUG] mod_sofia.c:362 N/A SOFIA DESTROY
> 2010-10-05 17:13:15.616463 [DEBUG] switch_core_state_machine.c:440 (N/A)
> State DESTROY going to sleep
> 2010-10-05 17:13:15.616463 [ERR] switch_ivr_originate.c:2648 Cannot
> create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
> 2010-10-05 17:13:15.616463 [DEBUG] switch_ivr_originate.c:3456 Originate
> Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
> 2010-10-05 17:13:15.616463 [INFO] mod_dptools.c:2411 Originate Failed.
> Cause: INVALID_NUMBER_FORMAT
> 2010-10-05 17:13:15.616463 [DEBUG] switch_channel.c:2359
> (sofia/external/00000000 at 1.1.1.1:5060) Callstate Change RINGING ->
> HANGUP
> 2010-10-05 17:13:15.616463 [NOTICE] mod_dptools.c:2474 Hangup
> sofia/external/00000000 at 1.1.1.1:5060 [CS_EXECUTE]
> [INVALID_NUMBER_FORMAT]
> 2010-10-05 17:13:15.616463 [DEBUG] switch_channel.c:2375 Send signal
> sofia/external/00000000 at 1.1.1.1:5060 [KILL]
> 2010-10-05 17:13:15.616463 [INFO] mod_hash.c:300 Usage for TESTGW_TESTGW
> is now 0
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From msc at freeswitch.org  Tue Oct  5 10:30:58 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Oct 2010 10:30:58 -0700
Subject: [Freeswitch-users] latest git pull has error, can't install
In-Reply-To: <4CAB5D16.8020203@mgtech.com>
References: <4CAB4393.4010900@mgtech.com> <4CAB4F39.8050705@mgtech.com>
	<4CAB5D16.8020203@mgtech.com>
Message-ID: <AANLkTim=xKrSo8vhjLon9D=tQO27HezCpdEqVZN20vjZ@mail.gmail.com>

Mario,

Let the build system work for you. Usually just "make current" is all you
need. Occasionally you need the bootstrap and ./configure thing, but that's
only when the boys make some deeper changes. Here's a script that I run
every single day on 3 different CentOS machines (2 32-bit, 1 64-bit) and it
never fails me - except in those few instances that I need the
bootstrap/configure thing.

<563>:cat rebuild_freeswitch.sh
#!/bin/sh
freeswitch -stop
make current
cd libs/esl
make
make perlmod
cd ../..
sleep 10
freeswitch -nc -nonat

(I always rebuild ESL and perlmod cuz I tend to tinker with that. Also, I
have symlinks to fs_cli and freeswitch bin files.)
-MC

On Tue, Oct 5, 2010 at 10:15 AM, Mario <mario_fs at mgtech.com> wrote:

> In case someone else runs across this I got it to work by:
> ./bootstrap.sh
> ./configure
> make current
>
> Still would like to know what could have caused this so I can avoid it.
>
> On 10/05/10 09:15, Mario wrote:
> > I tried executing the 2 lines below but still get the same error. I
> > quess a full install is in order? I am puzzled why git pull && make
> > current only worked one time.
> > automake-1.11 --foreign
> > automake --add-missing
> >
> >
> > On 10/05/10 08:26, Mario wrote:
> >> While working on a loop in FS I did a git pull and make current. I had
> >> done this before with no problem but today I got the error below. Any
> >> ideas? Thanks. Mario
> >>
> >>
> >> cd libs/sofia-sip && make clean
> >> make[1]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip'
> >>  cd . && /bin/sh /usr/local/src/freeswitch/libs/sofia-sip/missing --run
> >> automake-1.11 --foreign
> >> configure.ac:47: warning: AC_CACHE_VAL(ac_cc_environment, ...):
> >> suspicious cache-id, must contain _cv_ to be cached
> >> ../../lib/autoconf/general.m4:1998: AC_CACHE_VAL is expanded from...
> >> ../../lib/autoconf/general.m4:2019: AC_CACHE_CHECK is expanded from...
> >> m4/sac-general.m4:351: AC_CHECK_COMPILATION_ENVIRONMENT is expanded
> from...
> >> configure.ac:47: the top level
> >> configure.ac:49: warning: AC_PROG_CPP was called before SAC_TOOL_CC
> >> m4/sac-general.m4:90: SAC_TOOL_CC is expanded from...
> >> configure.ac:49: the top level
> >> configure.ac:56: required file `./compile' not found
> >> configure.ac:56:   `automake --add-missing' can install `compile'
> >> make[1]: *** [Makefile.in] Error 1
> >> make[1]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip'
> >> make: *** [update-clean] Error 2
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mario_fs at mgtech.com  Tue Oct  5 10:55:10 2010
From: mario_fs at mgtech.com (Mario)
Date: Tue, 05 Oct 2010 10:55:10 -0700
Subject: [Freeswitch-users] FS disk thrashing after 1440 Auto-Adding Alias +
	segmentation fault
Message-ID: <4CAB667E.4060405@mgtech.com>

Starting last night at 2am (it had been up for 10 hours) FS issued:
1440 Auto-Adding Alias [208.2.3.4] for profile [internal].
Now, when I start FS I get the message right away and heavy disk
thrashing. For the last couple of weeks while working on FS this
messages never came up (that I noticed). The machine goes to 100 percent
processor and the disk thrashes. I removed all external SIP accounts and
the ip-v6 profile to test and it still happens. This came out of the
blue and stopped everything I was working on. Looking into this all I
could find was it may be related to UPnP which the router has. Had been
working fine until now. Any help greatly appreciated. Mario

When I issue the shutdown command FS end with an error:

2010-10-05 10:22:44.889143 [CONSOLE] switch_loadable_module.c:1401
mod_siren has no shutdown routine
2010-10-05 10:22:44.889167 [NOTICE] switch_loadable_module.c:518
Deleting Codec SPEEX 99 Speex 32000hz 20ms
2010-10-05 10:22:44.889213 [NOTICE] switch_loadable_module.c:518
Deleting Codec SPEEX 99 Speex 16000hz 20ms
2010-10-05 10:22:44.889257 [NOTICE] switch_loadable_module.c:518
Deleting Codec SPEEX 99 Speex 8000hz 20ms
2010-10-05 10:22:44.889330 [CONSOLE] switch_loadable_module.c:1401
mod_speex has no shutdown routine
Segmentation fault (core dumped)

                     Name          Type
      Data      State
=================================================================================================
          210.17.234.127         alias
 internal      ALIASED
                 internal       profile
sip:mod_sofia at 100.24.1.37:5060      RUNNING (0)
                 external       profile
sip:mod_sofia at 100.24.1.37:5080      RUNNING (0)
              100.24.1.37         alias
  internal      ALIASED
=================================================================================================
2 profiles 2 aliases


From robert.hadley at teotech.com  Tue Oct  5 11:37:52 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Tue, 5 Oct 2010 11:37:52 -0700
Subject: [Freeswitch-users] Cannot email voicemails running FS as daemon user
Message-ID: <008c01cb64bc$6b40f700$41c2e500$@teotech.com>

Hi FS Users,

 

How do I configure FS or sendmail to allow emailing voicemails when running
as service with freeswitch user instead of running as root?  

 

These statements work (on CentOS 5.3) running as root but not running as
service and freeswitch user.

2010-10-05 11:19:30.406386 [DEBUG] mod_voicemail.c:2412 Deliver VM to
1018 at 192.168.72.138

2010-10-05 11:19:30.528573 [DEBUG] switch_utils.c:631 Emailed file
[/tmp/mail.12863027703a8c] to [robert.hadley at teotech.com]

2010-10-05 11:19:30.530594 [DEBUG] mod_voicemail.c:2580 Sending message to
robert.hadley at teotech.com

 

Thanks,

Robert

 

 

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From anthony.minessale at gmail.com  Tue Oct  5 11:45:21 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Oct 2010 13:45:21 -0500
Subject: [Freeswitch-users] FS disk thrashing after 1440 Auto-Adding
 Alias + segmentation fault
In-Reply-To: <4CAB667E.4060405@mgtech.com>
References: <4CAB667E.4060405@mgtech.com>
Message-ID: <AANLkTin0XbC2iLb=-YWvQgG_citJGbKKEGVRi5m9s7Py@mail.gmail.com>

Are you perhaps getting ddos'd by some sip scanner?


On Tue, Oct 5, 2010 at 12:55 PM, Mario <mario_fs at mgtech.com> wrote:
> Starting last night at 2am (it had been up for 10 hours) FS issued:
> 1440 Auto-Adding Alias [208.2.3.4] for profile [internal].
> Now, when I start FS I get the message right away and heavy disk
> thrashing. For the last couple of weeks while working on FS this
> messages never came up (that I noticed). The machine goes to 100 percent
> processor and the disk thrashes. I removed all external SIP accounts and
> the ip-v6 profile to test and it still happens. This came out of the
> blue and stopped everything I was working on. Looking into this all I
> could find was it may be related to UPnP which the router has. Had been
> working fine until now. Any help greatly appreciated. Mario
>
> When I issue the shutdown command FS end with an error:
>
> 2010-10-05 10:22:44.889143 [CONSOLE] switch_loadable_module.c:1401
> mod_siren has no shutdown routine
> 2010-10-05 10:22:44.889167 [NOTICE] switch_loadable_module.c:518
> Deleting Codec SPEEX 99 Speex 32000hz 20ms
> 2010-10-05 10:22:44.889213 [NOTICE] switch_loadable_module.c:518
> Deleting Codec SPEEX 99 Speex 16000hz 20ms
> 2010-10-05 10:22:44.889257 [NOTICE] switch_loadable_module.c:518
> Deleting Codec SPEEX 99 Speex 8000hz 20ms
> 2010-10-05 10:22:44.889330 [CONSOLE] switch_loadable_module.c:1401
> mod_speex has no shutdown routine
> Segmentation fault (core dumped)
>
> ? ? ? ? ? ? ? ? ? ? Name ? ? ? ? ?Type
> ? ? ?Data ? ? ?State
> =================================================================================================
> ? ? ? ? ?210.17.234.127 ? ? ? ? alias
> ?internal ? ? ?ALIASED
> ? ? ? ? ? ? ? ? internal ? ? ? profile
> sip:mod_sofia at 100.24.1.37:5060 ? ? ?RUNNING (0)
> ? ? ? ? ? ? ? ? external ? ? ? profile
> sip:mod_sofia at 100.24.1.37:5080 ? ? ?RUNNING (0)
> ? ? ? ? ? ? ?100.24.1.37 ? ? ? ? alias
> ?internal ? ? ?ALIASED
> =================================================================================================
> 2 profiles 2 aliases
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From mario_fs at mgtech.com  Tue Oct  5 11:49:32 2010
From: mario_fs at mgtech.com (Mario)
Date: Tue, 05 Oct 2010 11:49:32 -0700
Subject: [Freeswitch-users] FS disk thrashing after 1440 Auto-Adding
 Alias + segmentation fault
In-Reply-To: <4CAB667E.4060405@mgtech.com>
References: <4CAB667E.4060405@mgtech.com>
Message-ID: <4CAB733C.8090407@mgtech.com>

This turned out to be a hack attempt from outside! Once I blocked the
offending address things started working. FYI the hacker is 208.109.87.234.

On 10/05/10 10:55, Mario wrote:
> Starting last night at 2am (it had been up for 10 hours) FS issued:
> 1440 Auto-Adding Alias [208.2.3.4] for profile [internal].
> Now, when I start FS I get the message right away and heavy disk
> thrashing. For the last couple of weeks while working on FS this
> messages never came up (that I noticed). The machine goes to 100 percent
> processor and the disk thrashes. I removed all external SIP accounts and
> the ip-v6 profile to test and it still happens. This came out of the
> blue and stopped everything I was working on. Looking into this all I
> could find was it may be related to UPnP which the router has. Had been
> working fine until now. Any help greatly appreciated. Mario
> 
> When I issue the shutdown command FS end with an error:
> 
> 2010-10-05 10:22:44.889143 [CONSOLE] switch_loadable_module.c:1401
> mod_siren has no shutdown routine
> 2010-10-05 10:22:44.889167 [NOTICE] switch_loadable_module.c:518
> Deleting Codec SPEEX 99 Speex 32000hz 20ms
> 2010-10-05 10:22:44.889213 [NOTICE] switch_loadable_module.c:518
> Deleting Codec SPEEX 99 Speex 16000hz 20ms
> 2010-10-05 10:22:44.889257 [NOTICE] switch_loadable_module.c:518
> Deleting Codec SPEEX 99 Speex 8000hz 20ms
> 2010-10-05 10:22:44.889330 [CONSOLE] switch_loadable_module.c:1401
> mod_speex has no shutdown routine
> Segmentation fault (core dumped)
> 
>                      Name          Type
>       Data      State
> =================================================================================================
>           210.17.234.127         alias
>  internal      ALIASED
>                  internal       profile
> sip:mod_sofia at 100.24.1.37:5060      RUNNING (0)
>                  external       profile
> sip:mod_sofia at 100.24.1.37:5080      RUNNING (0)
>               100.24.1.37         alias
>   internal      ALIASED
> =================================================================================================
> 2 profiles 2 aliases


From msc at freeswitch.org  Tue Oct  5 11:49:25 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Oct 2010 11:49:25 -0700
Subject: [Freeswitch-users] generate inband dtmf
In-Reply-To: <AANLkTikzbAv+kWfcmVPdq7FsFA+VAL9FRZqhF5hWietX@mail.gmail.com>
References: <AANLkTikzbAv+kWfcmVPdq7FsFA+VAL9FRZqhF5hWietX@mail.gmail.com>
Message-ID: <AANLkTinKtDWHb5BpCmFWv6+7L-D77g_VyszhU=qB3Svm@mail.gmail.com>

Get a console debug of a call in progress and put it in pastebin. Be sure to
use console loglevel 7. Hopefully the debug output will tell you what's
going on when it tries to do inband DTMF sending.

Also, check this handy page for tips on collecting information for
troubleshooting:
http://wiki.freeswitch.org/wiki/Reporting_Bugs

-MC

On Tue, Oct 5, 2010 at 7:19 AM, Holger Esser <hesser4900 at gmail.com> wrote:

> Hi guys,
>
>
>
> I have a question about the start_dtmf_generate function. I have a
> dialplan entry that is called from a socket call like this:
>
> originate sofia/gateway/teliax/xxxxxxxxxx 5000
>
>
>
> <condition field="destination_number" expression="^5000$">
>
>         <action application="answer"/>
>
>         <action application="sleep" data="1000"/>
>
>         <action application="playback" data="voicemail/vm-goodbye.wav"/>
>
>         <action application="sleep" data="2000"/>
>
>         <action application="start_dtmf_generate"/>
>
>         <action application="send_dtmf" data="0123456789"/>
>
>         <action application="sleep" data="90000"/>
>
>       </condition>
>
>
>
>
>
> However, DTMF are still via the RFC and not inband. I am using the latest
> GIT.
>
>
>
> Any ideas?
>
>
>
> Thx
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mario_fs at mgtech.com  Tue Oct  5 11:55:51 2010
From: mario_fs at mgtech.com (Mario)
Date: Tue, 05 Oct 2010 11:55:51 -0700
Subject: [Freeswitch-users] Been hacked - what's the best way to prevent sip
	scanner?
Message-ID: <4CAB74B7.6050805@mgtech.com>

After working 4 hours I found that FS was hosed due to someone from
208.109.87.234 sending tons of traffic to FS. I blocked the IP address.
Not only did it overload the connection but stopped FS from working,
meaning no phones. This had not happened with the SPA9000. I listed on
on a FS conference call discussing this issue. Is there someplace that
has a list of things to do to prevent/reduce this? I did have ports,
etc. blocked in the firewall. Thanks. Mario


From tgraziano at myitdepartment.net  Tue Oct  5 12:01:36 2010
From: tgraziano at myitdepartment.net (Tony Graziano)
Date: Tue, 5 Oct 2010 15:01:36 -0400
Subject: [Freeswitch-users] Been hacked - what's the best way to prevent
 sip scanner?
In-Reply-To: <4CAB74B7.6050805@mgtech.com>
References: <4CAB74B7.6050805@mgtech.com>
Message-ID: <AANLkTi=5oOOQLnpPy5+ghr0EJgnKNmLMDAOfYeDVsptg@mail.gmail.com>

Depends on your config. If the attack is vicious enough, the best defense it
to block it upstream, if possible.

I always put my servers behind firewalls, but that's me.

My firewall is helpful in that is limits the connections per second from any
ip address. This is not something that can be helpful if the attack is
distributed though.

There are some good scripts out there to help identify a potential attack
and help alert your firewall, etc.



On Tue, Oct 5, 2010 at 2:55 PM, Mario <mario_fs at mgtech.com> wrote:

> After working 4 hours I found that FS was hosed due to someone from
> 208.109.87.234 sending tons of traffic to FS. I blocked the IP address.
> Not only did it overload the connection but stopped FS from working,
> meaning no phones. This had not happened with the SPA9000. I listed on
> on a FS conference call discussing this issue. Is there someplace that
> has a list of things to do to prevent/reduce this? I did have ports,
> etc. blocked in the firewall. Thanks. Mario
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgraziano at voice.myitdepartment.net
Fax: 434.984.8431

Email: tgraziano at myitdepartment.net

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpdesk at voice.myitdepartment.net
Fax: 434.984.8427

Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/

Why do mathematicians always confuse Halloween and Christmas?
Because 31 Oct = 25 Dec.
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From william.suffill at gmail.com  Tue Oct  5 12:04:51 2010
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 5 Oct 2010 15:04:51 -0400
Subject: [Freeswitch-users] Been hacked - what's the best way to prevent
 sip scanner?
In-Reply-To: <4CAB74B7.6050805@mgtech.com>
References: <4CAB74B7.6050805@mgtech.com>
Message-ID: <AANLkTi=V7gu0f4n_aa7S_M2t8GruZLqONPnG26N7tT++@mail.gmail.com>

fail2ban is 1 option.
http://wiki.freeswitch.org/wiki/Fail2ban

<http://wiki.freeswitch.org/wiki/Fail2ban>after X # of failed attempts it
will be auto added to the firewall.

Restricting the external ports to only trusted hosts can be an option as
well if you don't have any incoming sip calls from unknown sources.
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From shamun.toha at gmail.com  Tue Oct  5 12:14:07 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Tue, 5 Oct 2010 21:14:07 +0200
Subject: [Freeswitch-users] Been hacked - what's the best way to prevent
 sip scanner?
In-Reply-To: <4CAB74B7.6050805@mgtech.com>
References: <4CAB74B7.6050805@mgtech.com>
Message-ID: <AANLkTi=+wvhAjTdJ0AgPi31m7c4Z8RT5eQ77rPd-hSGY@mail.gmail.com>

I agree with @William Suffill mentioned. very good advise. Because putting
third party router in front of your FS switch will put your back to NAT
issues.

So, putting iptables as suggested can protect + keep the voice quality
stable.

Thanks  & Regards


On Tue, Oct 5, 2010 at 8:55 PM, Mario <mario_fs at mgtech.com> wrote:

> After working 4 hours I found that FS was hosed due to someone from
> 208.109.87.234 sending tons of traffic to FS. I blocked the IP address.
> Not only did it overload the connection but stopped FS from working,
> meaning no phones. This had not happened with the SPA9000. I listed on
> on a FS conference call discussing this issue. Is there someplace that
> has a list of things to do to prevent/reduce this? I did have ports,
> etc. blocked in the firewall. Thanks. Mario
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mario_fs at mgtech.com  Tue Oct  5 12:41:17 2010
From: mario_fs at mgtech.com (Mario)
Date: Tue, 05 Oct 2010 12:41:17 -0700
Subject: [Freeswitch-users] Been hacked - what's the best way to prevent
 sip scanner?
In-Reply-To: <AANLkTi=+wvhAjTdJ0AgPi31m7c4Z8RT5eQ77rPd-hSGY@mail.gmail.com>
References: <4CAB74B7.6050805@mgtech.com>
	<AANLkTi=+wvhAjTdJ0AgPi31m7c4Z8RT5eQ77rPd-hSGY@mail.gmail.com>
Message-ID: <4CAB7F5D.3020308@mgtech.com>

I have a Linksys firewall and use NAT. The thing that's puzzling is even
with the natting how someone could get through to the natted address of
FS. It worries me that the phone system could be killed this way.
Anyway, this was a GoDaddy address and I reported it, they finally
killed it a little while ago. BTW, never could get Asterisk to work with
NAT but FS worked first time which is why I went with it.

On 10/05/2010 12:14 PM, Shamun toha md wrote:
> I agree with @William Suffill mentioned. very good advise. Because
> putting third party router in front of your FS switch will put your back
> to NAT issues.
> 
> So, putting iptables as suggested can protect + keep the voice quality
> stable.
> 
> Thanks  & Regards
> 
> 
> On Tue, Oct 5, 2010 at 8:55 PM, Mario <mario_fs at mgtech.com
> <mailto:mario_fs at mgtech.com>> wrote:
> 
>     After working 4 hours I found that FS was hosed due to someone from
>     208.109.87.234 sending tons of traffic to FS. I blocked the IP address.
>     Not only did it overload the connection but stopped FS from working,
>     meaning no phones. This had not happened with the SPA9000. I listed on
>     on a FS conference call discussing this issue. Is there someplace that
>     has a list of things to do to prevent/reduce this? I did have ports,
>     etc. blocked in the firewall. Thanks. Mario
> 
>> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
*Mario*


From peter.olsson at visionutveckling.se  Tue Oct  5 12:50:01 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 5 Oct 2010 21:50:01 +0200
Subject: [Freeswitch-users] Been hacked - what's the best way to prevent
 sip scanner?
In-Reply-To: <4CAB7F5D.3020308@mgtech.com>
References: <4CAB74B7.6050805@mgtech.com>
	<AANLkTi=+wvhAjTdJ0AgPi31m7c4Z8RT5eQ77rPd-hSGY@mail.gmail.com>,
	<4CAB7F5D.3020308@mgtech.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57E092D09B@cooper>

Well, if they couldn't reach FS, it couldn't be used at all, so it's not that strange :)

/Peter

________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Mario [mario_fs at mgtech.com]
Skickat: den 5 oktober 2010 21:41
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Been hacked - what's the best way to prevent sip scanner?

I have a Linksys firewall and use NAT. The thing that's puzzling is even
with the natting how someone could get through to the natted address of
FS. It worries me that the phone system could be killed this way.
Anyway, this was a GoDaddy address and I reported it, they finally
killed it a little while ago. BTW, never could get Asterisk to work with
NAT but FS worked first time which is why I went with it.

On 10/05/2010 12:14 PM, Shamun toha md wrote:
> I agree with @William Suffill mentioned. very good advise. Because
> putting third party router in front of your FS switch will put your back
> to NAT issues.
>
> So, putting iptables as suggested can protect + keep the voice quality
> stable.
>
> Thanks  & Regards
>
>
> On Tue, Oct 5, 2010 at 8:55 PM, Mario <mario_fs at mgtech.com
> <mailto:mario_fs at mgtech.com>> wrote:
>
>     After working 4 hours I found that FS was hosed due to someone from
>     208.109.87.234 sending tons of traffic to FS. I blocked the IP address.
>     Not only did it overload the connection but stopped FS from working,
>     meaning no phones. This had not happened with the SPA9000. I listed on
>     on a FS conference call discussing this issue. Is there someplace that
>     has a list of things to do to prevent/reduce this? I did have ports,
>     etc. blocked in the firewall. Thanks. Mario
>
>>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

--
*Mario*

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

!DSPAM:4cab810e32931924414817!



From brian at freeswitch.org  Tue Oct  5 13:10:18 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 5 Oct 2010 15:10:18 -0500
Subject: [Freeswitch-users] Been hacked - what's the best way to prevent
	sip scanner?
In-Reply-To: <AANLkTi=V7gu0f4n_aa7S_M2t8GruZLqONPnG26N7tT++@mail.gmail.com>
References: <4CAB74B7.6050805@mgtech.com>
	<AANLkTi=V7gu0f4n_aa7S_M2t8GruZLqONPnG26N7tT++@mail.gmail.com>
Message-ID: <D6827951-E07D-4AA5-8AFC-E7060A91FAFB@freeswitch.org>

see also scripts/perl/blacklist.pl along with scripts/perl/honeypot.pl

NEXT!!!

/b

On Oct 5, 2010, at 2:04 PM, William Suffill wrote:

> fail2ban is 1 option. 
> http://wiki.freeswitch.org/wiki/Fail2ban
> 
> after X # of failed attempts it will be auto added to the firewall.
> 
> Restricting the external ports to only trusted hosts can be an option as well if you don't have any incoming sip calls from unknown sources.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From hesser4900 at gmail.com  Tue Oct  5 08:32:11 2010
From: hesser4900 at gmail.com (Holger Esser)
Date: Tue, 5 Oct 2010 10:32:11 -0500
Subject: [Freeswitch-users] latest git pull has error, can't install
In-Reply-To: <4CAB4393.4010900@mgtech.com>
References: <4CAB4393.4010900@mgtech.com>
Message-ID: <AANLkTin4Y902nsYczPCK4dr_DpKe_bTsW9s+RvJe10u_@mail.gmail.com>

Run the ./bootstrap.sh again and it will work after that.

On Tue, Oct 5, 2010 at 10:26 AM, Mario <mario_fs at mgtech.com> wrote:

> While working on a loop in FS I did a git pull and make current. I had
> done this before with no problem but today I got the error below. Any
> ideas? Thanks. Mario
>
>
> cd libs/sofia-sip && make clean
> make[1]: Entering directory `/usr/local/src/freeswitch/libs/sofia-sip'
>  cd . && /bin/sh /usr/local/src/freeswitch/libs/sofia-sip/missing --run
> automake-1.11 --foreign
> configure.ac:47: warning: AC_CACHE_VAL(ac_cc_environment, ...):
> suspicious cache-id, must contain _cv_ to be cached
> ../../lib/autoconf/general.m4:1998: AC_CACHE_VAL is expanded from...
> ../../lib/autoconf/general.m4:2019: AC_CACHE_CHECK is expanded from...
> m4/sac-general.m4:351: AC_CHECK_COMPILATION_ENVIRONMENT is expanded from...
> configure.ac:47: the top level
> configure.ac:49: warning: AC_PROG_CPP was called before SAC_TOOL_CC
> m4/sac-general.m4:90: SAC_TOOL_CC is expanded from...
> configure.ac:49: the top level
> configure.ac:56: required file `./compile' not found
> configure.ac:56:   `automake --add-missing' can install `compile'
> make[1]: *** [Makefile.in] Error 1
> make[1]: Leaving directory `/usr/local/src/freeswitch/libs/sofia-sip'
> make: *** [update-clean] Error 2
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From hesser4900 at gmail.com  Tue Oct  5 12:49:47 2010
From: hesser4900 at gmail.com (Holger Esser)
Date: Tue, 5 Oct 2010 14:49:47 -0500
Subject: [Freeswitch-users] generate inband dtmf
In-Reply-To: <AANLkTinKtDWHb5BpCmFWv6+7L-D77g_VyszhU=qB3Svm@mail.gmail.com>
References: <AANLkTikzbAv+kWfcmVPdq7FsFA+VAL9FRZqhF5hWietX@mail.gmail.com>
	<AANLkTinKtDWHb5BpCmFWv6+7L-D77g_VyszhU=qB3Svm@mail.gmail.com>
Message-ID: <AANLkTin9ke5JJf2HV91Qh_4yc9M_vLc3C+k05evCW+yr@mail.gmail.com>

Hi Mike,

Thank you very much for your input. I figured out what I did wrong. I went
through the code and saw that you attach a bug. It in turn will listen for
the incoming RTP events and then generate the appropriate DTMF back out. I
only needed to use gentones for my testing.
In any case, many thanks for putting me on the right path with the console
logging.

Holger

On Tue, Oct 5, 2010 at 1:49 PM, Michael Collins <msc at freeswitch.org> wrote:

> Get a console debug of a call in progress and put it in pastebin. Be sure
> to use console loglevel 7. Hopefully the debug output will tell you what's
> going on when it tries to do inband DTMF sending.
>
> Also, check this handy page for tips on collecting information for
> troubleshooting:
> http://wiki.freeswitch.org/wiki/Reporting_Bugs
>
> -MC
>
> On Tue, Oct 5, 2010 at 7:19 AM, Holger Esser <hesser4900 at gmail.com> wrote:
>
>> Hi guys,
>>
>>
>>
>> I have a question about the start_dtmf_generate function. I have a
>> dialplan entry that is called from a socket call like this:
>>
>> originate sofia/gateway/teliax/xxxxxxxxxx 5000
>>
>>
>>
>> <condition field="destination_number" expression="^5000$">
>>
>>         <action application="answer"/>
>>
>>         <action application="sleep" data="1000"/>
>>
>>         <action application="playback" data="voicemail/vm-goodbye.wav"/>
>>
>>         <action application="sleep" data="2000"/>
>>
>>         <action application="start_dtmf_generate"/>
>>
>>         <action application="send_dtmf" data="0123456789"/>
>>
>>         <action application="sleep" data="90000"/>
>>
>>       </condition>
>>
>>
>>
>>
>>
>> However, DTMF are still via the RFC and not inband. I am using the latest
>> GIT.
>>
>>
>>
>> Any ideas?
>>
>>
>>
>> Thx
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Nabble at slickdeals.endjunk.com  Tue Oct  5 13:13:31 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 5 Oct 2010 13:13:31 -0700 (PDT)
Subject: [Freeswitch-users] Been hacked - what's the best way to prevent
	sip scanner?
In-Reply-To: <4CAB74B7.6050805@mgtech.com>
References: <4CAB74B7.6050805@mgtech.com>
Message-ID: <1286309611314-5604670.post@n2.nabble.com>



Mario wrote:
> After working 4 hours I found that FS was hosed due to someone from
> 208.109.87.234 sending tons of traffic to FS.
I just hope you don't have ports forwarding set on your router to your FS.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Been-hacked-what-s-the-best-way-to-prevent-sip-scanner-tp5604432p5604670.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mario_fs at mgtech.com  Tue Oct  5 13:48:23 2010
From: mario_fs at mgtech.com (Mario)
Date: Tue, 05 Oct 2010 13:48:23 -0700
Subject: [Freeswitch-users] Been hacked - what's the best way to prevent
 sip scanner?
In-Reply-To: <1286309611314-5604670.post@n2.nabble.com>
References: <4CAB74B7.6050805@mgtech.com>
	<1286309611314-5604670.post@n2.nabble.com>
Message-ID: <4CAB8F17.1070204@mgtech.com>

No I don't, thanks.

On 10/05/2010 01:13 PM, mazilo wrote:
> 
> 
> Mario wrote:
>> After working 4 hours I found that FS was hosed due to someone from
>> 208.109.87.234 sending tons of traffic to FS.
> I just hope you don't have ports forwarding set on your router to your FS.
> 
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.

-- 
*Mario*


From david.ponzone at ipeva.fr  Tue Oct  5 14:25:36 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 5 Oct 2010 23:25:36 +0200
Subject: [Freeswitch-users] Been hacked - what's the best way to prevent
	sip scanner?
In-Reply-To: <4CAB74B7.6050805@mgtech.com>
References: <4CAB74B7.6050805@mgtech.com>
Message-ID: <2BAF7C30-3F85-4427-9322-DF73EA64BDE4@ipeva.fr>

Mario,

personnally, following a DoS REGISTER attack I had recently, I  
configured some rate-limiting on REGISTER attempts.
Here is the result, in "iptables-save" format:

-A INPUT -d YOUR_FS_IP -p udp -m udp --dport YOUR_FS_PORT -m string -- 
string "REGISTER" --algo kmp --from 20 --to 60 -j dos-filter-register- 
external
-A dos-filter-register-external -m hashlimit --hashlimit 5/sec -- 
hashlimit-burst 8 --hashlimit-mode srcip --hashlimit-name REGISTER -- 
hashlimit-htable-size 24593 --hashlimit-htable-expire 90000 -j RETURN
-A dos-filter-register-external -j REJECT --reject-with icmp-admin- 
prohibited

This will ratelimit REGISTER packets coming to YOUR_FS_IP:YOUR_FS_PORT  
to 5 per second for each source IP.

PS: thanks to the experienced people on #freeswitch for the help  
provided to setup this filter.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 05/10/2010 ? 20:55, Mario a ?crit :

> After working 4 hours I found that FS was hosed due to someone from
> 208.109.87.234 sending tons of traffic to FS. I blocked the IP  
> address.
> Not only did it overload the connection but stopped FS from working,
> meaning no phones. This had not happened with the SPA9000. I listed on
> on a FS conference call discussing this issue. Is there someplace that
> has a list of things to do to prevent/reduce this? I did have ports,
> etc. blocked in the firewall. Thanks. Mario
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jstricker at lightnex.com  Tue Oct  5 11:10:56 2010
From: jstricker at lightnex.com (Jeremy Stricker)
Date: Tue, 05 Oct 2010 12:10:56 -0600
Subject: [Freeswitch-users] valet_park timeout and spot announcement
Message-ID: <4CAB6A30.7070608@lightnex.com>

Hello all,

Is there anyway to set a call timeout when a call is valet parked so 
that it will ring back to the extension that parked it if it isn't 
answered by the intended recipient?  We have an user who parks calls and 
then blindly announces the parking spot over an intercom.  With this 
setup there is a great risk of a caller getting stuck in park.  If there 
isn't a timeout method, can anyone recommend any alternate setup that 
would allow at least two parking spots with visual indication of calls 
parked on the handsets (Aastra 6730i and 6757i).

Secondly, is there anyway to stop read-back of the parking spot on the 
parkee's side of the call?  Currently, the parker performs an attended 
transfer, waits for and hears the spot read-back.  The parkee hears MOH 
while the transfer is being performed, but then the spot announcement 
themselves as soon as the parker finishes the attended transfer by 
hanging up.

Thanks for any help available.

Jeremy
-- 
Jeremy Stricker
LightNex Communications

e: jstricker at lightnex.com
p: 877-342-3768
w: www.lightnex.com


From Peter.Hinman at ParcelPool.com  Tue Oct  5 14:57:46 2010
From: Peter.Hinman at ParcelPool.com (Peter Hinman)
Date: Tue, 05 Oct 2010 15:57:46 -0600
Subject: [Freeswitch-users] ODBC and MSSQL
Message-ID: <4CAB9F5A.2090503@ParcelPool.com>

  We've got a MSSQL server that I was hoping to put to use as an ODBC 
data source for FS.  I can connect via ODBS to the database from the FS 
box and run queries using tsql.

FS also appears to be able to connect to the database, as it creates 
several tables when it starts up.  After successfully creating tables, 
it appears to run a test query, receives an error from which it 
concludes that transactions are not supported and then disables ODBC.

Is MSSQL expected to work with ODBC and FS?
It does successfully create tables. Does this mean that FS disabling 
ODBC is a bug that should be reported in jira?
Is there something I should be setting in the config other than
<param name="core-db-dsn" value="DSN:USERID:PASSWORD" />

Thanks,
Peter


Log snippet below:

2010-10-05 15:37:33.169737 [INFO] switch_core_sqldb.c:1275 Opening DB
2010-10-05 15:37:33.234702 [ERR] switch_odbc.c:427 ERR: [begin;delete 
from channels where hostname='';delete from channels where 
hostname='';commit;]
[STATE: 42000 CODE 102 ERROR: [FreeTDS][SQL Server]Incorrect syntax near 
';'.
]
2010-10-05 15:37:33.234775 [ERR] switch_core_sqldb.c:404 SQL ERR [STATE: 
42000 CODE 8180 ERROR: [FreeTDS][SQL Server]Statement(s) could not be 
prepared.
]
begin;delete from channels where hostname='';delete from channels where 
hostname='';commit;
2010-10-05 15:37:33.234796 [ERR] switch_core_sqldb.c:1326 Transactions 
not supported on your DB, disabling ODBC
2010-10-05 15:37:33.235916 [INFO] switch_core_sqldb.c:1275 Opening DB

Thanks,
Peter



From kris at kriskinc.com  Tue Oct  5 15:09:40 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 5 Oct 2010 18:09:40 -0400
Subject: [Freeswitch-users] Been hacked - what's the best way to prevent
 sip scanner?
In-Reply-To: <2BAF7C30-3F85-4427-9322-DF73EA64BDE4@ipeva.fr>
References: <4CAB74B7.6050805@mgtech.com>
	<2BAF7C30-3F85-4427-9322-DF73EA64BDE4@ipeva.fr>
Message-ID: <AANLkTi=Udm6xj7HOq7ZGJUwfoKDkpCVzpUaQ3Ufb7885@mail.gmail.com>

It may have started here:

http://blog.krisk.org/2008/07/sip-dosddos-mitigation.html

If other people have made improvements I'd love to hear about them and
maintain the script somewhere :).

On Tue, Oct 5, 2010 at 5:25 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
> Mario,
> personnally, following a DoS REGISTER attack I had recently, I configured
> some rate-limiting on REGISTER attempts.
> Here is the result, in "iptables-save" format:
> -A INPUT -d YOUR_FS_IP -p udp -m udp --dport YOUR_FS_PORT -m string --string
> "REGISTER" --algo kmp --from 20 --to 60 -j dos-filter-register-external
> -A dos-filter-register-external -m hashlimit --hashlimit 5/sec
> --hashlimit-burst 8 --hashlimit-mode srcip --hashlimit-name REGISTER
> --hashlimit-htable-size 24593 --hashlimit-htable-expire 90000 -j RETURN
> -A dos-filter-register-external -j REJECT --reject-with
> icmp-admin-prohibited
> This will ratelimit REGISTER packets coming to YOUR_FS_IP:YOUR_FS_PORT to 5
> per second for each source IP.
> PS: thanks to the experienced people on #freeswitch for the help provided to
> setup this filter.
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From covici at ccs.covici.com  Tue Oct  5 16:31:02 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 05 Oct 2010 19:31:02 -0400
Subject: [Freeswitch-users] mod_cepstral appears not to be working
Message-ID: <2046.1286321462@ccs.covici.com>

I was trying to put a tts engine into a conference and I made sure
mod_cepstral was loaded, however its not working.  I have Allison-8kHz
as the voice and its listed in the profile for the conference.  One
thing, if I type say and some text after the conference name in the cli,
it says ERR with no explanation.  Now I wonder whether it knows about
the voice version I have which is 5.x?
So how do I debug such a thing?

Thanks for any suggestions.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From stephen at stephenjc.com  Tue Oct  5 16:46:48 2010
From: stephen at stephenjc.com (stephen at stephenjc)
Date: Tue, 5 Oct 2010 19:46:48 -0400
Subject: [Freeswitch-users] mod_cepstral appears not to be working
In-Reply-To: <2046.1286321462@ccs.covici.com>
References: <2046.1286321462@ccs.covici.com>
Message-ID: <AANLkTimQP_e63iSpbc3W+pyYCFbJO4jzX4qxXxz83iXh@mail.gmail.com>

Test from the command line with swift and make sure localhost resolves to
127.0.0.1 in your host file.

Thanks,
Stephen C
-All of my email addresses go to the same place
-Save Paper, think before you print



On Tue, Oct 5, 2010 at 7:31 PM, <covici at ccs.covici.com> wrote:

> I was trying to put a tts engine into a conference and I made sure
> mod_cepstral was loaded, however its not working.  I have Allison-8kHz
> as the voice and its listed in the profile for the conference.  One
> thing, if I type say and some text after the conference name in the cli,
> it says ERR with no explanation.  Now I wonder whether it knows about
> the voice version I have which is 5.x?
> So how do I debug such a thing?
>
> Thanks for any suggestions.
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From stephen at stephenjc.com  Tue Oct  5 17:08:06 2010
From: stephen at stephenjc.com (stephen at stephenjc)
Date: Tue, 5 Oct 2010 20:08:06 -0400
Subject: [Freeswitch-users] firewall suggestions
Message-ID: <AANLkTimaoZDMCQcc5SxZ+zvtMmvYNJDW5gn7LjpCNZkX@mail.gmail.com>

I am thinking of putting a pfsense firewall as a transparent bridge between
my colo boxes (including freeswitch) between me and the world. Does any one
see any issues with this?


Thanks,
Stephen C
-All of my email addresses go to the same place
-Save Paper, think before you print
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From covici at ccs.covici.com  Tue Oct  5 17:26:10 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 05 Oct 2010 20:26:10 -0400
Subject: [Freeswitch-users] mod_cepstral appears not to be working
In-Reply-To: <AANLkTimQP_e63iSpbc3W+pyYCFbJO4jzX4qxXxz83iXh@mail.gmail.com>
References: <2046.1286321462@ccs.covici.com>
	<AANLkTimQP_e63iSpbc3W+pyYCFbJO4jzX4qxXxz83iXh@mail.gmail.com>
Message-ID: <6127.1286324770@ccs.covici.com>

Yep, both those tests work.

stephen at stephenjc <stephen at stephenjc.com> wrote:

> Test from the command line with swift and make sure localhost resolves to
> 127.0.0.1 in your host file.
> 
> Thanks,
> Stephen C
> -All of my email addresses go to the same place
> -Save Paper, think before you print
> 
> 
> 
> On Tue, Oct 5, 2010 at 7:31 PM, <covici at ccs.covici.com> wrote:
> 
> > I was trying to put a tts engine into a conference and I made sure
> > mod_cepstral was loaded, however its not working.  I have Allison-8kHz
> > as the voice and its listed in the profile for the conference.  One
> > thing, if I type say and some text after the conference name in the cli,
> > it says ERR with no explanation.  Now I wonder whether it knows about
> > the voice version I have which is 5.x?
> > So how do I debug such a thing?
> >
> > Thanks for any suggestions.
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From mario_fs at mgtech.com  Tue Oct  5 17:46:11 2010
From: mario_fs at mgtech.com (Mario)
Date: Tue, 05 Oct 2010 17:46:11 -0700
Subject: [Freeswitch-users] Been hacked - what's the best way to prevent
 sip scanner?
In-Reply-To: <AANLkTi=Udm6xj7HOq7ZGJUwfoKDkpCVzpUaQ3Ufb7885@mail.gmail.com>
References: <4CAB74B7.6050805@mgtech.com>	<2BAF7C30-3F85-4427-9322-DF73EA64BDE4@ipeva.fr>
	<AANLkTi=Udm6xj7HOq7ZGJUwfoKDkpCVzpUaQ3Ufb7885@mail.gmail.com>
Message-ID: <4CABC6D3.4020309@mgtech.com>

Thanks to all, I will look into and use some or all options presented
here. Amazing, this happened so fast, I didn't even have the machine in
full use, just in testing on/off since 9/20 and I'm only a SOHO. Still
have 2 things to figure out before the switch. Thanks again!

On 10/05/2010 03:09 PM, Kristian Kielhofner wrote:
> It may have started here:
> 
> http://blog.krisk.org/2008/07/sip-dosddos-mitigation.html
> 
> If other people have made improvements I'd love to hear about them and
> maintain the script somewhere :).
> 
> On Tue, Oct 5, 2010 at 5:25 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
>> Mario,
>> personnally, following a DoS REGISTER attack I had recently, I configured
>> some rate-limiting on REGISTER attempts.
>> Here is the result, in "iptables-save" format:
>> -A INPUT -d YOUR_FS_IP -p udp -m udp --dport YOUR_FS_PORT -m string --string
>> "REGISTER" --algo kmp --from 20 --to 60 -j dos-filter-register-external
>> -A dos-filter-register-external -m hashlimit --hashlimit 5/sec
>> --hashlimit-burst 8 --hashlimit-mode srcip --hashlimit-name REGISTER
>> --hashlimit-htable-size 24593 --hashlimit-htable-expire 90000 -j RETURN
>> -A dos-filter-register-external -j REJECT --reject-with
>> icmp-admin-prohibited
>> This will ratelimit REGISTER packets coming to YOUR_FS_IP:YOUR_FS_PORT to 5
>> per second for each source IP.
>> PS: thanks to the experienced people on #freeswitch for the help provided to
>> setup this filter.
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
> 

-- 
*Mario*


From jeff at jefflenk.com  Tue Oct  5 19:25:31 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 5 Oct 2010 19:25:31 -0700 (PDT)
Subject: [Freeswitch-users] ODBC and MSSQL
In-Reply-To: <4CAB9F5A.2090503@ParcelPool.com>
References: <4CAB9F5A.2090503@ParcelPool.com>
Message-ID: <1286331931285-5605569.post@n2.nabble.com>


This is a known issue, and is waiting for submittal.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ODBC-and-MSSQL-tp5605076p5605569.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From woodydickson at gmail.com  Tue Oct  5 21:20:43 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Wed, 6 Oct 2010 12:20:43 +0800
Subject: [Freeswitch-users] compare bill rate of different increment
Message-ID: <AANLkTi=vb8mne+rQiNzGKTpRRq5LCXSVXYX-1FP0wXk6@mail.gmail.com>

Hi,

I would like to use mod_lcr but I am wondering how I should due with
rate decks from different providers that have different bill
increment?

If one deck's increment is 6 s and another one is 1 s, is it fair to
compare the two deck just purely based on the rate or is there a
better way of handling that?

Thank you for your input.

Thanks,

Woody


From abubacker at bksystems.co.in  Tue Oct  5 21:23:49 2010
From: abubacker at bksystems.co.in (abubacker)
Date: Wed, 06 Oct 2010 09:53:49 +0530
Subject: [Freeswitch-users] Is anyone familiar with Telemaque?
In-Reply-To: <4CAAE5DE.6080407@telemaque.fr>
References: <AANLkTimib+3T7L9sP28HN6BREJF5ca2j2nou-84YU=Ln@mail.gmail.com>
	<4CAAE5DE.6080407@telemaque.fr>
Message-ID: <4CABF9D5.7000501@bksys.co.in>

On Tuesday 05 October 2010 02:16 PM, Tristan Mah? wrote:
> There's someone working for them on list :)
>
> Feel free to ask me questions Michael.
>
> Regards,
>
> Gled.
>
> Le 10/04/2010 08:39 PM, Michael Collins a ?crit :
>> Dear FreeSWITCH Community,
>>
>> Some of you may have stumbled upon this news item:
>>
>> http://blogs.oracle.com/mysql/2010/10/innovating_with_open_source_call_center_services.html
>>
>> Evidently there is a company in France (Telemaque) who is using MySQL 
>> + Kamailio + FreeSWITCH for some heavy duty call processing. If you 
>> are at all familiar with this company please let me know. I'd like to 
>> learn more about what they are doing. (Je ne parle pas boucoup de 
>> Francais :( )
>>
>> Thanks for your help,
>> MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>      
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

"Feel free to ask me questions Michael"

I think I must use this opportunity ,
please specify the software ( OS ) and the hardwares required to handle 
the heavy duty call
processing and also specify the FreeSWITCH version.
I guess you could answer this very precisely.

Thanks in Advance !

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

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

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From adminjew at gmail.com  Tue Oct  5 22:20:56 2010
From: adminjew at gmail.com (Yitzchok)
Date: Wed, 6 Oct 2010 01:20:56 -0400
Subject: [Freeswitch-users] mod_managed on linux centos
In-Reply-To: <AANLkTinKemVBa5HNSrJk6EOfWYM5wd7CyL4L-JX-BawQ@mail.gmail.com>
References: <AANLkTikg4bxav+e04fAmJ_dKTHKERAPk715h9oHeYMRh@mail.gmail.com>
	<1283347083542-5486890.post@n2.nabble.com>
	<AANLkTi=xt9CZBPZLnBX56e2Rjdc_FRf_=er6Yy7w7ZuN@mail.gmail.com>
	<AANLkTi=S=tpzczDM8WSm6N8AeTkN3XNTPda6048sRoJ9@mail.gmail.com>
	<4C801B6E.4050103@puzzled.xs4all.nl>
	<AANLkTimzjbLHbObgsQKqX3t-hfaZWr95UGMi17YxZFHS@mail.gmail.com>
	<4C838C3B.3070202@puzzled.xs4all.nl>
	<AANLkTinP+UfvkM0dmBycy5GqivAXNHGF9RnTBNvV0gV4@mail.gmail.com>
	<AANLkTik1CY5-6DUvsLg512bGRr66xLF5H8Gi7WYRenHx@mail.gmail.com>
	<1286205151753-5599563.post@n2.nabble.com>
	<AANLkTinKemVBa5HNSrJk6EOfWYM5wd7CyL4L-JX-BawQ@mail.gmail.com>
Message-ID: <AANLkTinF2yvyA6ZerxjFoOi+ti8DAGHAyEYfBFrRGiR1@mail.gmail.com>

--Bug Fix--

Here is a diff file for *src/mod/languages/mod_managed/mod_managed.cpp *it
seems to work but I don't know if this has some side effects.


----

@@ -377,17 +377,13 @@ SWITCH_STANDARD_API(managedrun_api_function)
  stream->write_function(stream, "-ERR no args specified!\n");
  return SWITCH_STATUS_SUCCESS;
  }
-#ifndef _MANAGED
- mono_thread_attach(globals.domain);
-#endif
+
  if (executeBackgroundDelegate(cmd)) {
  stream->write_function(stream, "+OK\n");
  } else {
  stream->write_function(stream, "-ERR ExecuteBackground returned false
(unknown module or exception?).\n");
  }
-#ifndef _MANAGED
- mono_thread_detach(mono_thread_current());
-#endif
+
  return SWITCH_STATUS_SUCCESS;
 }

@@ -397,15 +393,11 @@ SWITCH_STANDARD_API(managed_api_function)
  stream->write_function(stream, "-ERR no args specified!\n");
  return SWITCH_STATUS_SUCCESS;
  }
-#ifndef _MANAGED
- mono_thread_attach(globals.domain);
-#endif
+
  if (!(executeDelegate(cmd, stream, stream->param_event))) {
  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Execute failed
for %s (unknown module or exception).\n", cmd);
  }
-#ifndef _MANAGED
- mono_thread_detach(mono_thread_current());
-#endif
+
  return SWITCH_STATUS_SUCCESS;
 }

@@ -415,15 +407,11 @@ SWITCH_STANDARD_APP(managed_app_function)
  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "No args
specified!\n");
  return;
  }
-#ifndef _MANAGED
- mono_thread_attach(globals.domain);
-#endif
+
  if (!(runDelegate(data, session))) {
  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Application run
failed for %s (unknown module or exception).\n", data);
  }
-#ifndef _MANAGED
- mono_thread_detach(mono_thread_current());
-#endif
+
 }

 SWITCH_STANDARD_API(managedreload_api_function)
@@ -432,15 +420,11 @@ SWITCH_STANDARD_API(managedreload_api_function)
  stream->write_function(stream, "-ERR no args specified!\n");
  return SWITCH_STATUS_SUCCESS;
  }
-#ifndef _MANAGED
- mono_thread_attach(globals.domain);
-#endif
+
  if (!(reloadDelegate(cmd))) {
  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Execute failed
for %s (unknown module or exception).\n", cmd);
  }
-#ifndef _MANAGED
- mono_thread_detach(mono_thread_current());
-#endif
+
  return SWITCH_STATUS_SUCCESS;
 }



Yitzchok


On Mon, Oct 4, 2010 at 12:04 PM, Yitzchok <adminjew at gmail.com> wrote:

> I have already done that but it doesn't help also note that on the first
> call it works fine.
>
>
> Yitzchok
>
>
>
> On Mon, Oct 4, 2010 at 11:12 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
>
>>
>> Have you seen:
>> http://wiki.freeswitch.org/wiki/Mod_mono
>>
>> <dllmap dll="mod_managed.dll" target="mod_managed.so" os="!windows"/>
>>
>> regarding null exceptions?
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/mod-managed-on-linux-centos-tp5485480p5599563.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From abid_freeswitch at live.com  Tue Oct  5 22:44:37 2010
From: abid_freeswitch at live.com (Abid Saleem)
Date: Wed, 6 Oct 2010 11:44:37 +0600
Subject: [Freeswitch-users] Radius AAA
Message-ID: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>


Hi All,
Can someone provide a clear guideline on how to setup Radius AAA to work with FS. Please send complete installation & configuration instructions. Thanks for help in advance.
Regards--------------Abid SaleemSr. Product ManagerTerminus Technologies 		 	   		  
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From stas at khirman.com  Tue Oct  5 23:11:38 2010
From: stas at khirman.com (Stas Khirman)
Date: Tue, 5 Oct 2010 23:11:38 -0700
Subject: [Freeswitch-users] FreeSWITCH CLI description?
Message-ID: <02db01cb651d$576a1ca0$063e55e0$@khirman.com>

Hi,

 

Sorry for probably dumb  questions, but I can't find answer in the
docs/wiki:

 

I'm looking for a good list of major CLI commands and their descriptions -
any pointers are deeply appreciated.

 

Regards

Stas

 

 

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From nagalenoj at gmail.com  Tue Oct  5 23:43:03 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Wed, 6 Oct 2010 12:13:03 +0530
Subject: [Freeswitch-users] FreeSWITCH CLI description?
In-Reply-To: <02db01cb651d$576a1ca0$063e55e0$@khirman.com>
References: <02db01cb651d$576a1ca0$063e55e0$@khirman.com>
Message-ID: <AANLkTikU=EwkTd5+e7ZLL9MTae5+aUO-DasSv43VNiA_@mail.gmail.com>

I don't understand what do you mean by major CLI commands, but here is the
list.,

http://wiki.freeswitch.org/wiki/Mod_commands

On Wed, Oct 6, 2010 at 11:41 AM, Stas Khirman <stas at khirman.com> wrote:

> Hi,
>
>
>
> Sorry for probably dumb  questions, but I can?t find answer in the
> docs/wiki:
>
>
>
> I?m looking for a good list of major CLI commands and their descriptions ?
> any pointers are deeply appreciated.
>
>
>
> Regards
>
> Stas
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Nagalenoj H.
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From devel at thom.fr.eu.org  Wed Oct  6 01:26:52 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Wed, 06 Oct 2010 10:26:52 +0200
Subject: [Freeswitch-users] Cannot email voicemails running FS as daemon
 user
In-Reply-To: <008c01cb64bc$6b40f700$41c2e500$@teotech.com>
References: <008c01cb64bc$6b40f700$41c2e500$@teotech.com>
Message-ID: <6e044cdefb41641d5ac4a34237e6ce90@thom.fr.eu.org>



I don't know if you did make furhter investigation, but I got this
problem on Debian with sendmail where sendmail was doing segfault (I
don't remember the reason, but this must be somewhere in the list
archive). 

However, I ended up adding msmtp for doing the mail processing for
freeswitch. 

Fran?ois 

On Tue, 5 Oct 2010 11:37:52 -0700, "Robert Hadley"  wrote:  

Hi FS Users, 

How do I configure FS or sendmail to allow emailing voicemails when
running as service with freeswitch user instead of running as root?  

These statements work (on CentOS 5.3) running as root but not running
as service and freeswitch user. 

2010-10-05 11:19:30.406386 [DEBUG] mod_voicemail.c:2412 Deliver VM to
1018 at 192.168.72.138 

2010-10-05 11:19:30.528573 [DEBUG] switch_utils.c:631 Emailed file
[/tmp/mail.12863027703a8c] to [robert.hadley at teotech.com] 

2010-10-05 11:19:30.530594 [DEBUG] mod_voicemail.c:2580 Sending message
to robert.hadley at teotech.com 

Thanks, 

Robert 

   
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From pkelly at gmail.com  Wed Oct  6 02:18:11 2010
From: pkelly at gmail.com (Pete Kelly)
Date: Wed, 6 Oct 2010 10:18:11 +0100
Subject: [Freeswitch-users] Problem bridging 2 calls with bypass_media=true
Message-ID: <AANLkTinBT8MJAqHX3yq8u1TO9XFNn1+7zAgGuAQ2+JqZ@mail.gmail.com>

Hi

I am having some problems bridging 2 calls and bypassing the media.

I am using a lua script executed from the command line to phone out to the
first device (leg A), then on answer bridge a call to a second device (leg
B).

If I set the bypass_media=true, then freeswitch immediately sends out BYEs
after the reINVITEs have happened.

If I leave bypass_media unset, everything works fine except the media is
bridged.

The lua script I am using is a very simple one and looks like this:

obSession = freeswitch.Session("sofia/lpmedia/91979197 at 10.15.20.122:5060")
obSession:setVariable('bypass_media', 'true');

if obSession:ready() then
    -- Do something good here

    obSession:execute("bridge", "sofia/lpmedia/2000 at 10.15.20.122:5060")
end

I have tried variations on this including, putting [bypass_media=true]  on
leg A, and using the [bypass_media_after_bridge=true] on leg B

The logs say this:
 ...
2010-10-06 10:10:05.847278 [NOTICE] sofia.c:5085 Channel [sofia/lpmedia/
2000 at 10.15.20.122:5060] has been answered
2010-10-06 10:10:06.650155 [NOTICE] switch_cpp.cpp:976 Hangup sofia/lpmedia/
91979197 at 10.15.20.122:5060 [CS_HIBERNATE] [NORMAL_CLEARING]
2010-10-06 10:10:06.650155 [NOTICE] switch_ivr_bridge.c:936 Hangup
sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_HIBERNATE] [NORMAL_CLEARING]
2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1228 Session 27
(sofia/lpmedia/91979197 at 10.15.20.122:5060) Ended
2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close Channel
sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_DESTROY]
freeswitch at pete-desktop> 2010-10-06 10:10:06.651420 [NOTICE]
switch_core_session.c:1228 Session 28 (sofia/lpmedia/2000 at 10.15.20.122:5060)
Ended
2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close Channel
sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_DESTROY]


Does anyone have any ideas as to why freeswitch may be ending the calls?

I am using the latest git checkout - the behaviour was slightly different
with the 1.06 package. The call would setup but the reINVITEs did not happen
as they should.

Pete
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From david.ponzone at ipeva.fr  Wed Oct  6 02:26:11 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 6 Oct 2010 11:26:11 +0200
Subject: [Freeswitch-users] Problem bridging 2 calls with
	bypass_media=true
In-Reply-To: <AANLkTinBT8MJAqHX3yq8u1TO9XFNn1+7zAgGuAQ2+JqZ@mail.gmail.com>
References: <AANLkTinBT8MJAqHX3yq8u1TO9XFNn1+7zAgGuAQ2+JqZ@mail.gmail.com>
Message-ID: <74D51A69-A897-43FC-9581-FB31E39E0527@ipeva.fr>

Pete,

perhaps leg A and B can't agree on the codec ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 06/10/2010 ? 11:18, Pete Kelly a ?crit :

> Hi
>
> I am having some problems bridging 2 calls and bypassing the media.
>
> I am using a lua script executed from the command line to phone out  
> to the first device (leg A), then on answer bridge a call to a  
> second device (leg B).
>
> If I set the bypass_media=true, then freeswitch immediately sends  
> out BYEs after the reINVITEs have happened.
>
> If I leave bypass_media unset, everything works fine except the  
> media is bridged.
>
> The lua script I am using is a very simple one and looks like this:
>
> obSession = freeswitch.Session("sofia/lpmedia/ 
> 91979197 at 10.15.20.122:5060")
> obSession:setVariable('bypass_media', 'true');
>
> if obSession:ready() then
>     -- Do something good here
>
>     obSession:execute("bridge", "sofia/lpmedia/ 
> 2000 at 10.15.20.122:5060")
> end
>
> I have tried variations on this including, putting  
> [bypass_media=true]  on leg A, and using the  
> [bypass_media_after_bridge=true] on leg B
>
> The logs say this:
>  ...
> 2010-10-06 10:10:05.847278 [NOTICE] sofia.c:5085 Channel [sofia/lpmedia/2000 at 10.15.20.122 
> :5060] has been answered
> 2010-10-06 10:10:06.650155 [NOTICE] switch_cpp.cpp:976 Hangup sofia/lpmedia/91979197 at 10.15.20.122 
> :5060 [CS_HIBERNATE] [NORMAL_CLEARING]
> 2010-10-06 10:10:06.650155 [NOTICE] switch_ivr_bridge.c:936 Hangup sofia/lpmedia/2000 at 10.15.20.122 
> :5060 [CS_HIBERNATE] [NORMAL_CLEARING]
> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1228  
> Session 27 (sofia/lpmedia/91979197 at 10.15.20.122:5060) Ended
> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close  
> Channel sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_DESTROY]
> freeswitch at pete-desktop> 2010-10-06 10:10:06.651420 [NOTICE]  
> switch_core_session.c:1228 Session 28 (sofia/lpmedia/ 
> 2000 at 10.15.20.122:5060) Ended
> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close  
> Channel sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_DESTROY]
>
>
> Does anyone have any ideas as to why freeswitch may be ending the  
> calls?
>
> I am using the latest git checkout - the behaviour was slightly  
> different with the 1.06 package. The call would setup but the  
> reINVITEs did not happen as they should.
>
> Pete
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From pkelly at gmail.com  Wed Oct  6 02:33:40 2010
From: pkelly at gmail.com (Pete Kelly)
Date: Wed, 6 Oct 2010 10:33:40 +0100
Subject: [Freeswitch-users] Problem bridging 2 calls with
	bypass_media=true
In-Reply-To: <74D51A69-A897-43FC-9581-FB31E39E0527@ipeva.fr>
References: <AANLkTinBT8MJAqHX3yq8u1TO9XFNn1+7zAgGuAQ2+JqZ@mail.gmail.com>
	<74D51A69-A897-43FC-9581-FB31E39E0527@ipeva.fr>
Message-ID: <AANLkTimzOMMPw6KfsJ_imqQQw39zUvtFH0iX_vhYXbjp@mail.gmail.com>

I've checked and it's PCMU for all legs, including Freeswitch... anyway it's
Freeswitch which is ending the calls. I would expect leg A or leg B to end
the call if it was a codec issue.

On 6 October 2010 10:26, David Ponzone <david.ponzone at ipeva.fr> wrote:

> Pete,
>
> perhaps leg A and B can't agree on the codec ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 06/10/2010 ? 11:18, Pete Kelly a ?crit :
>
> Hi
>
> I am having some problems bridging 2 calls and bypassing the media.
>
> I am using a lua script executed from the command line to phone out to the
> first device (leg A), then on answer bridge a call to a second device (leg
> B).
>
> If I set the bypass_media=true, then freeswitch immediately sends out BYEs
> after the reINVITEs have happened.
>
> If I leave bypass_media unset, everything works fine except the media is
> bridged.
>
> The lua script I am using is a very simple one and looks like this:
>
> obSession = freeswitch.Session("sofia/lpmedia/91979197 at 10.15.20.122:5060")
> obSession:setVariable('bypass_media', 'true');
>
> if obSession:ready() then
>     -- Do something good here
>
>     obSession:execute("bridge", "sofia/lpmedia/2000 at 10.15.20.122:5060")
> end
>
> I have tried variations on this including, putting [bypass_media=true]  on
> leg A, and using the [bypass_media_after_bridge=true] on leg B
>
> The logs say this:
>  ...
> 2010-10-06 10:10:05.847278 [NOTICE] sofia.c:5085 Channel [sofia/lpmedia/
> 2000 at 10.15.20.122:5060] has been answered
> 2010-10-06 10:10:06.650155 [NOTICE] switch_cpp.cpp:976 Hangup
> sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_HIBERNATE] [NORMAL_CLEARING]
> 2010-10-06 10:10:06.650155 [NOTICE] switch_ivr_bridge.c:936 Hangup
> sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_HIBERNATE] [NORMAL_CLEARING]
> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1228 Session 27
> (sofia/lpmedia/91979197 at 10.15.20.122:5060) Ended
> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close
> Channel sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_DESTROY]
> freeswitch at pete-desktop> 2010-10-06 10:10:06.651420 [NOTICE]
> switch_core_session.c:1228 Session 28 (sofia/lpmedia/
> 2000 at 10.15.20.122:5060) Ended
> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close
> Channel sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_DESTROY]
>
>
> Does anyone have any ideas as to why freeswitch may be ending the calls?
>
> I am using the latest git checkout - the behaviour was slightly different
> with the 1.06 package. The call would setup but the reINVITEs did not happen
> as they should.
>
> Pete
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Wed Oct  6 02:42:11 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 6 Oct 2010 11:42:11 +0200
Subject: [Freeswitch-users] Problem bridging 2 calls with
	bypass_media=true
In-Reply-To: <AANLkTimzOMMPw6KfsJ_imqQQw39zUvtFH0iX_vhYXbjp@mail.gmail.com>
References: <AANLkTinBT8MJAqHX3yq8u1TO9XFNn1+7zAgGuAQ2+JqZ@mail.gmail.com>
	<74D51A69-A897-43FC-9581-FB31E39E0527@ipeva.fr>
	<AANLkTimzOMMPw6KfsJ_imqQQw39zUvtFH0iX_vhYXbjp@mail.gmail.com>
Message-ID: <7DB4D4EE-77F9-4353-A561-8F6EB8D99846@ipeva.fr>

my question is going to sound stupid but:
how long before the call is ended ?
are you sure A can talk to B directly ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 06/10/2010 ? 11:33, Pete Kelly a ?crit :

> I've checked and it's PCMU for all legs, including Freeswitch...  
> anyway it's Freeswitch which is ending the calls. I would expect leg  
> A or leg B to end the call if it was a codec issue.
>
> On 6 October 2010 10:26, David Ponzone <david.ponzone at ipeva.fr> wrote:
> Pete,
>
> perhaps leg A and B can't agree on the codec ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le  
> d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
> Le 06/10/2010 ? 11:18, Pete Kelly a ?crit :
>
>> Hi
>>
>> I am having some problems bridging 2 calls and bypassing the media.
>>
>> I am using a lua script executed from the command line to phone out  
>> to the first device (leg A), then on answer bridge a call to a  
>> second device (leg B).
>>
>> If I set the bypass_media=true, then freeswitch immediately sends  
>> out BYEs after the reINVITEs have happened.
>>
>> If I leave bypass_media unset, everything works fine except the  
>> media is bridged.
>>
>> The lua script I am using is a very simple one and looks like this:
>>
>> obSession = freeswitch.Session("sofia/lpmedia/ 
>> 91979197 at 10.15.20.122:5060")
>> obSession:setVariable('bypass_media', 'true');
>>
>> if obSession:ready() then
>>     -- Do something good here
>>
>>     obSession:execute("bridge", "sofia/lpmedia/ 
>> 2000 at 10.15.20.122:5060")
>> end
>>
>> I have tried variations on this including, putting  
>> [bypass_media=true]  on leg A, and using the  
>> [bypass_media_after_bridge=true] on leg B
>>
>> The logs say this:
>>  ...
>> 2010-10-06 10:10:05.847278 [NOTICE] sofia.c:5085 Channel [sofia/lpmedia/2000 at 10.15.20.122 
>> :5060] has been answered
>> 2010-10-06 10:10:06.650155 [NOTICE] switch_cpp.cpp:976 Hangup sofia/lpmedia/91979197 at 10.15.20.122 
>> :5060 [CS_HIBERNATE] [NORMAL_CLEARING]
>> 2010-10-06 10:10:06.650155 [NOTICE] switch_ivr_bridge.c:936 Hangup sofia/lpmedia/2000 at 10.15.20.122 
>> :5060 [CS_HIBERNATE] [NORMAL_CLEARING]
>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1228  
>> Session 27 (sofia/lpmedia/91979197 at 10.15.20.122:5060) Ended
>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230  
>> Close Channel sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_DESTROY]
>> freeswitch at pete-desktop> 2010-10-06 10:10:06.651420 [NOTICE]  
>> switch_core_session.c:1228 Session 28 (sofia/lpmedia/2000 at 10.15.20.122 
>> :5060) Ended
>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230  
>> Close Channel sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_DESTROY]
>>
>>
>> Does anyone have any ideas as to why freeswitch may be ending the  
>> calls?
>>
>> I am using the latest git checkout - the behaviour was slightly  
>> different with the 1.06 package. The call would setup but the  
>> reINVITEs did not happen as they should.
>>
>> Pete
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From pkelly at gmail.com  Wed Oct  6 02:52:58 2010
From: pkelly at gmail.com (Pete Kelly)
Date: Wed, 6 Oct 2010 10:52:58 +0100
Subject: [Freeswitch-users] Problem bridging 2 calls with
	bypass_media=true
In-Reply-To: <7DB4D4EE-77F9-4353-A561-8F6EB8D99846@ipeva.fr>
References: <AANLkTinBT8MJAqHX3yq8u1TO9XFNn1+7zAgGuAQ2+JqZ@mail.gmail.com>
	<74D51A69-A897-43FC-9581-FB31E39E0527@ipeva.fr>
	<AANLkTimzOMMPw6KfsJ_imqQQw39zUvtFH0iX_vhYXbjp@mail.gmail.com>
	<7DB4D4EE-77F9-4353-A561-8F6EB8D99846@ipeva.fr>
Message-ID: <AANLkTikhspWFyZAJh6Zi1v++bJdqBwPBkr-FQozWkuOi@mail.gmail.com>

After the call is set up, Freeswitch sends a reINVITE to legA, with legB's
media IP/port, then vice versa.

Then it sends a BYE to both legs immediately .

However I've put a dummy while loop after the bridge:

while(1==1) do

end

and the call remains up!

However the lua script now never ends.

On 6 October 2010 10:42, David Ponzone <david.ponzone at ipeva.fr> wrote:

> my question is going to sound stupid but:
> how long before the call is ended ?
> are you sure A can talk to B directly ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 06/10/2010 ? 11:33, Pete Kelly a ?crit :
>
> I've checked and it's PCMU for all legs, including Freeswitch... anyway
> it's Freeswitch which is ending the calls. I would expect leg A or leg B to
> end the call if it was a codec issue.
>
> On 6 October 2010 10:26, David Ponzone <david.ponzone at ipeva.fr> wrote:
>
>> Pete,
>>
>> perhaps leg A and B can't agree on the codec ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 06/10/2010 ? 11:18, Pete Kelly a ?crit :
>>
>> Hi
>>
>> I am having some problems bridging 2 calls and bypassing the media.
>>
>> I am using a lua script executed from the command line to phone out to the
>> first device (leg A), then on answer bridge a call to a second device (leg
>> B).
>>
>> If I set the bypass_media=true, then freeswitch immediately sends out BYEs
>> after the reINVITEs have happened.
>>
>> If I leave bypass_media unset, everything works fine except the media is
>> bridged.
>>
>> The lua script I am using is a very simple one and looks like this:
>>
>> obSession = freeswitch.Session("sofia/lpmedia/91979197 at 10.15.20.122:5060
>> ")
>> obSession:setVariable('bypass_media', 'true');
>>
>> if obSession:ready() then
>>     -- Do something good here
>>
>>     obSession:execute("bridge", "sofia/lpmedia/2000 at 10.15.20.122:5060")
>> end
>>
>> I have tried variations on this including, putting [bypass_media=true]  on
>> leg A, and using the [bypass_media_after_bridge=true] on leg B
>>
>> The logs say this:
>>  ...
>> 2010-10-06 10:10:05.847278 [NOTICE] sofia.c:5085 Channel [sofia/lpmedia/
>> 2000 at 10.15.20.122:5060] has been answered
>> 2010-10-06 10:10:06.650155 [NOTICE] switch_cpp.cpp:976 Hangup
>> sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_HIBERNATE] [NORMAL_CLEARING]
>> 2010-10-06 10:10:06.650155 [NOTICE] switch_ivr_bridge.c:936 Hangup
>> sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_HIBERNATE] [NORMAL_CLEARING]
>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1228 Session 27
>> (sofia/lpmedia/91979197 at 10.15.20.122:5060) Ended
>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close
>> Channel sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_DESTROY]
>> freeswitch at pete-desktop> 2010-10-06 10:10:06.651420 [NOTICE]
>> switch_core_session.c:1228 Session 28 (sofia/lpmedia/
>> 2000 at 10.15.20.122:5060) Ended
>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close
>> Channel sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_DESTROY]
>>
>>
>> Does anyone have any ideas as to why freeswitch may be ending the calls?
>>
>> I am using the latest git checkout - the behaviour was slightly different
>> with the 1.06 package. The call would setup but the reINVITEs did not happen
>> as they should.
>>
>> Pete
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Wed Oct  6 03:08:43 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Wed, 6 Oct 2010 12:08:43 +0200
Subject: [Freeswitch-users] Problem bridging 2 calls with
	bypass_media=true
In-Reply-To: <AANLkTikhspWFyZAJh6Zi1v++bJdqBwPBkr-FQozWkuOi@mail.gmail.com>
References: <AANLkTinBT8MJAqHX3yq8u1TO9XFNn1+7zAgGuAQ2+JqZ@mail.gmail.com>
	<74D51A69-A897-43FC-9581-FB31E39E0527@ipeva.fr>
	<AANLkTimzOMMPw6KfsJ_imqQQw39zUvtFH0iX_vhYXbjp@mail.gmail.com>
	<7DB4D4EE-77F9-4353-A561-8F6EB8D99846@ipeva.fr>
	<AANLkTikhspWFyZAJh6Zi1v++bJdqBwPBkr-FQozWkuOi@mail.gmail.com>
Message-ID: <85D2911B-02CF-42BC-9297-99B45AE95E92@ipeva.fr>

Can you just do the same bridge from dialplan, just to know if it's  
related to LUA ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 06/10/2010 ? 11:52, Pete Kelly a ?crit :

> After the call is set up, Freeswitch sends a reINVITE to legA, with  
> legB's media IP/port, then vice versa.
>
> Then it sends a BYE to both legs immediately .
>
> However I've put a dummy while loop after the bridge:
>
> while(1==1) do
>
> end
>
> and the call remains up!
>
> However the lua script now never ends.
>
> On 6 October 2010 10:42, David Ponzone <david.ponzone at ipeva.fr> wrote:
> my question is going to sound stupid but:
> how long before the call is ended ?
> are you sure A can talk to B directly ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le  
> d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
> Le 06/10/2010 ? 11:33, Pete Kelly a ?crit :
>
>> I've checked and it's PCMU for all legs, including Freeswitch...  
>> anyway it's Freeswitch which is ending the calls. I would expect  
>> leg A or leg B to end the call if it was a codec issue.
>>
>> On 6 October 2010 10:26, David Ponzone <david.ponzone at ipeva.fr>  
>> wrote:
>> Pete,
>>
>> perhaps leg A and B can't agree on the codec ?
>>
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ? l'intention exclusive de ses destinataires. Toute  
>> utilisation ou diffusion non autoris?e est interdite. Tout message  
>> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
>> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de  
>> le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>>
>> Le 06/10/2010 ? 11:18, Pete Kelly a ?crit :
>>
>>> Hi
>>>
>>> I am having some problems bridging 2 calls and bypassing the media.
>>>
>>> I am using a lua script executed from the command line to phone  
>>> out to the first device (leg A), then on answer bridge a call to a  
>>> second device (leg B).
>>>
>>> If I set the bypass_media=true, then freeswitch immediately sends  
>>> out BYEs after the reINVITEs have happened.
>>>
>>> If I leave bypass_media unset, everything works fine except the  
>>> media is bridged.
>>>
>>> The lua script I am using is a very simple one and looks like this:
>>>
>>> obSession = freeswitch.Session("sofia/lpmedia/ 
>>> 91979197 at 10.15.20.122:5060")
>>> obSession:setVariable('bypass_media', 'true');
>>>
>>> if obSession:ready() then
>>>     -- Do something good here
>>>
>>>     obSession:execute("bridge", "sofia/lpmedia/ 
>>> 2000 at 10.15.20.122:5060")
>>> end
>>>
>>> I have tried variations on this including, putting  
>>> [bypass_media=true]  on leg A, and using the  
>>> [bypass_media_after_bridge=true] on leg B
>>>
>>> The logs say this:
>>>  ...
>>> 2010-10-06 10:10:05.847278 [NOTICE] sofia.c:5085 Channel [sofia/lpmedia/2000 at 10.15.20.122 
>>> :5060] has been answered
>>> 2010-10-06 10:10:06.650155 [NOTICE] switch_cpp.cpp:976 Hangup sofia/lpmedia/91979197 at 10.15.20.122 
>>> :5060 [CS_HIBERNATE] [NORMAL_CLEARING]
>>> 2010-10-06 10:10:06.650155 [NOTICE] switch_ivr_bridge.c:936 Hangup sofia/lpmedia/2000 at 10.15.20.122 
>>> :5060 [CS_HIBERNATE] [NORMAL_CLEARING]
>>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1228  
>>> Session 27 (sofia/lpmedia/91979197 at 10.15.20.122:5060) Ended
>>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230  
>>> Close Channel sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_DESTROY]
>>> freeswitch at pete-desktop> 2010-10-06 10:10:06.651420 [NOTICE]  
>>> switch_core_session.c:1228 Session 28 (sofia/lpmedia/2000 at 10.15.20.122 
>>> :5060) Ended
>>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230  
>>> Close Channel sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_DESTROY]
>>>
>>>
>>> Does anyone have any ideas as to why freeswitch may be ending the  
>>> calls?
>>>
>>> I am using the latest git checkout - the behaviour was slightly  
>>> different with the 1.06 package. The call would setup but the  
>>> reINVITEs did not happen as they should.
>>>
>>> Pete
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mnhassan at usa.net  Wed Oct  6 03:21:14 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Wed, 6 Oct 2010 16:21:14 +0600
Subject: [Freeswitch-users] Problem bridging 2 calls with
	bypass_media=true
In-Reply-To: <AANLkTikhspWFyZAJh6Zi1v++bJdqBwPBkr-FQozWkuOi@mail.gmail.com>
References: <AANLkTinBT8MJAqHX3yq8u1TO9XFNn1+7zAgGuAQ2+JqZ@mail.gmail.com>
	<74D51A69-A897-43FC-9581-FB31E39E0527@ipeva.fr>
	<AANLkTimzOMMPw6KfsJ_imqQQw39zUvtFH0iX_vhYXbjp@mail.gmail.com>
	<7DB4D4EE-77F9-4353-A561-8F6EB8D99846@ipeva.fr>
	<AANLkTikhspWFyZAJh6Zi1v++bJdqBwPBkr-FQozWkuOi@mail.gmail.com>
Message-ID: <AANLkTimZbh6xHidmLQ5DAtRsXezY69XCXbnCyxB6tOYQ@mail.gmail.com>

That does not look like the full log. Can you put the complete log,
right from the start to the end, in the PasteBin?

Also, it appears that both the legs are to the same IP. Is that by design?

Regards
HASSAN


On 2010-10-06, Pete Kelly <pkelly at gmail.com> wrote:
> After the call is set up, Freeswitch sends a reINVITE to legA, with legB's
> media IP/port, then vice versa.
>
> Then it sends a BYE to both legs immediately .
>
> However I've put a dummy while loop after the bridge:
>
> while(1==1) do
>
> end
>
> and the call remains up!
>
> However the lua script now never ends.
>
> On 6 October 2010 10:42, David Ponzone <david.ponzone at ipeva.fr> wrote:
>
>> my question is going to sound stupid but:
>> how long before the call is ended ?
>> are you sure A can talk to B directly ?
>>
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et
>> d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 06/10/2010 ? 11:33, Pete Kelly a ?crit :
>>
>> I've checked and it's PCMU for all legs, including Freeswitch... anyway
>> it's Freeswitch which is ending the calls. I would expect leg A or leg B
>> to
>> end the call if it was a codec issue.
>>
>> On 6 October 2010 10:26, David Ponzone <david.ponzone at ipeva.fr> wrote:
>>
>>> Pete,
>>>
>>> perhaps leg A and B can't agree on the codec ?
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou
>>> diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et
>>> d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 06/10/2010 ? 11:18, Pete Kelly a ?crit :
>>>
>>> Hi
>>>
>>> I am having some problems bridging 2 calls and bypassing the media.
>>>
>>> I am using a lua script executed from the command line to phone out to
>>> the
>>> first device (leg A), then on answer bridge a call to a second device
>>> (leg
>>> B).
>>>
>>> If I set the bypass_media=true, then freeswitch immediately sends out
>>> BYEs
>>> after the reINVITEs have happened.
>>>
>>> If I leave bypass_media unset, everything works fine except the media is
>>> bridged.
>>>
>>> The lua script I am using is a very simple one and looks like this:
>>>
>>> obSession = freeswitch.Session("sofia/lpmedia/91979197 at 10.15.20.122:5060
>>> ")
>>> obSession:setVariable('bypass_media', 'true');
>>>
>>> if obSession:ready() then
>>>     -- Do something good here
>>>
>>>     obSession:execute("bridge", "sofia/lpmedia/2000 at 10.15.20.122:5060")
>>> end
>>>
>>> I have tried variations on this including, putting [bypass_media=true]
>>> on
>>> leg A, and using the [bypass_media_after_bridge=true] on leg B
>>>
>>> The logs say this:
>>>  ...
>>> 2010-10-06 10:10:05.847278 [NOTICE] sofia.c:5085 Channel [sofia/lpmedia/
>>> 2000 at 10.15.20.122:5060] has been answered
>>> 2010-10-06 10:10:06.650155 [NOTICE] switch_cpp.cpp:976 Hangup
>>> sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_HIBERNATE] [NORMAL_CLEARING]
>>> 2010-10-06 10:10:06.650155 [NOTICE] switch_ivr_bridge.c:936 Hangup
>>> sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_HIBERNATE] [NORMAL_CLEARING]
>>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1228 Session 27
>>> (sofia/lpmedia/91979197 at 10.15.20.122:5060) Ended
>>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close
>>> Channel sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_DESTROY]
>>> freeswitch at pete-desktop> 2010-10-06 10:10:06.651420 [NOTICE]
>>> switch_core_session.c:1228 Session 28 (sofia/lpmedia/
>>> 2000 at 10.15.20.122:5060) Ended
>>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close
>>> Channel sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_DESTROY]
>>>
>>>
>>> Does anyone have any ideas as to why freeswitch may be ending the calls?
>>>
>>> I am using the latest git checkout - the behaviour was slightly different
>>> with the 1.06 package. The call would setup but the reINVITEs did not
>>> happen
>>> as they should.
>>>
>>> Pete
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>

-- 
Sent from my mobile device


From t.mahe at telemaque.fr  Wed Oct  6 03:25:56 2010
From: t.mahe at telemaque.fr (=?ISO-8859-1?Q?Tristan_Mah=E9?=)
Date: Wed, 06 Oct 2010 12:25:56 +0200
Subject: [Freeswitch-users] Is anyone familiar with Telemaque?
In-Reply-To: <4CABF9D5.7000501@bksys.co.in>
References: <AANLkTimib+3T7L9sP28HN6BREJF5ca2j2nou-84YU=Ln@mail.gmail.com>	<4CAAE5DE.6080407@telemaque.fr>
	<4CABF9D5.7000501@bksys.co.in>
Message-ID: <4CAC4EB4.6080901@telemaque.fr>

  Hi,

The telco cluster is powered by IBM servers ( from dual Xeon/8gb ram to 
dual quad core/16gb ram depending on the node ) with FC13 and latest FS 
git ( well almost always, as of today I'm a week late, but will upgrade 
soon again to benefit of the amazing work the dev do ).

If you have more precise questions, don't hesitate, I'm sorry if I can't 
reveal all the details on how we're doing things,but I'm sure you can 
understand that...

Regards,

Tristan.
Le 10/06/2010 06:23 AM, abubacker a ?crit :
> On Tuesday 05 October 2010 02:16 PM, Tristan Mah? wrote:
>> There's someone working for them on list :)
>>
>> Feel free to ask me questions Michael.
>>
>> Regards,
>>
>> Gled.
>>
>> Le 10/04/2010 08:39 PM, Michael Collins a ?crit :
>>> Dear FreeSWITCH Community,
>>>
>>> Some of you may have stumbled upon this news item:
>>>
>>> http://blogs.oracle.com/mysql/2010/10/innovating_with_open_source_call_center_services.html
>>>
>>> Evidently there is a company in France (Telemaque) who is using 
>>> MySQL + Kamailio + FreeSWITCH for some heavy duty call processing. 
>>> If you are at all familiar with this company please let me know. I'd 
>>> like to learn more about what they are doing. (Je ne parle pas 
>>> boucoup de Francais :( )
>>>
>>> Thanks for your help,
>>> MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>      
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>    
>
> "Feel free to ask me questions Michael"
>
> I think I must use this opportunity ,
> please specify the software ( OS ) and the hardwares required to 
> handle the heavy duty call
> processing and also specify the FreeSWITCH version.
> I guess you could answer this very precisely.
>
> Thanks in Advance !
> -- 
> Best regards,
> N.Abubacker ,
> Associate system engineer ,
> bk systems pvt ltd ,
> Ph : 9144-43902701
>
> Disclaimer:http://www.bksystems.co.in/email-policy
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From pkelly at gmail.com  Wed Oct  6 03:28:55 2010
From: pkelly at gmail.com (Pete Kelly)
Date: Wed, 6 Oct 2010 11:28:55 +0100
Subject: [Freeswitch-users] Problem bridging 2 calls with
	bypass_media=true
In-Reply-To: <AANLkTimZbh6xHidmLQ5DAtRsXezY69XCXbnCyxB6tOYQ@mail.gmail.com>
References: <AANLkTinBT8MJAqHX3yq8u1TO9XFNn1+7zAgGuAQ2+JqZ@mail.gmail.com>
	<74D51A69-A897-43FC-9581-FB31E39E0527@ipeva.fr>
	<AANLkTimzOMMPw6KfsJ_imqQQw39zUvtFH0iX_vhYXbjp@mail.gmail.com>
	<7DB4D4EE-77F9-4353-A561-8F6EB8D99846@ipeva.fr>
	<AANLkTikhspWFyZAJh6Zi1v++bJdqBwPBkr-FQozWkuOi@mail.gmail.com>
	<AANLkTimZbh6xHidmLQ5DAtRsXezY69XCXbnCyxB6tOYQ@mail.gmail.com>
Message-ID: <AANLkTi=2uJpnFh-kzwdp3kdT4rYseENeEovP=-B34g_p@mail.gmail.com>

Full log output is below.

Yes it's to the same IP by design... both 91979197 and 2000 are registered
devices at 10.15.20.122.

Pete

freeswitch at pete-desktop> lua click2call_call.lua
2010-10-06 11:27:31.361309 [NOTICE] switch_channel.c:779 New Channel
sofia/lpmedia/91979197 at 10.15.20.122:5060[f27fbfb0-b47f-4f57-9022-dbce1f50122c]
2010-10-06 11:27:32.014779 [INFO] sofia.c:709 sofia/lpmedia/
91979197 at 10.15.20.122:5060 Update Callee ID to "91979197" <91979197>
2010-10-06 11:27:32.014779 [NOTICE] sofia.c:4584 Ring-Ready sofia/lpmedia/
91979197 at 10.15.20.122:5060!
2010-10-06 11:27:34.198968 [NOTICE] sofia.c:5085 Channel [sofia/lpmedia/
91979197 at 10.15.20.122:5060] has been answered
2010-10-06 11:27:34.200100 [NOTICE] switch_channel.c:779 New Channel
sofia/lpmedia/2000 at 10.15.20.122:5060 [cade82f5-f4b7-490b-9ad4-8c0e392c6183]
2010-10-06 11:27:34.243046 [INFO] sofia.c:709 sofia/lpmedia/
2000 at 10.15.20.122:5060 Update Callee ID to "2000" <2000>
2010-10-06 11:27:34.243046 [NOTICE] sofia.c:4584 Ring-Ready sofia/lpmedia/
2000 at 10.15.20.122:5060!
2010-10-06 11:27:36.156477 [NOTICE] sofia.c:5085 Channel [sofia/lpmedia/
2000 at 10.15.20.122:5060] has been answered
2010-10-06 11:27:36.957598 [INFO] switch_cpp.cpp:1181 uuid
isf27fbfb0-b47f-4f57-9022-dbce1f50122c
2010-10-06 11:27:36.957598 [INFO] switch_cpp.cpp:1181 uuid
isf27fbfb0-b47f-4f57-9022-dbce1f50122c
2010-10-06 11:27:36.957598 [NOTICE] switch_cpp.cpp:976 Hangup sofia/lpmedia/
91979197 at 10.15.20.122:5060 [CS_HIBERNATE] [NORMAL_CLEARING]
2010-10-06 11:27:36.958738 [NOTICE] switch_ivr_bridge.c:936 Hangup
sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_HIBERNATE] [NORMAL_CLEARING]
freeswitch at pete-desktop>
2010-10-06 11:27:36.959748 [NOTICE] switch_core_session.c:1228 Session 6
(sofia/lpmedia/91979197 at 10.15.20.122:5060) Ended
2010-10-06 11:27:36.959748 [NOTICE] switch_core_session.c:1230 Close Channel
sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_DESTROY]
2010-10-06 11:27:36.959748 [NOTICE] switch_core_session.c:1228 Session 7
(sofia/lpmedia/2000 at 10.15.20.122:5060) Ended
f2010-10-06 11:27:36.959748 [NOTICE] switch_core_session.c:1230 Close
Channel sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_DESTROY]


On 6 October 2010 11:21, Nyamul Hassan <mnhassan at usa.net> wrote:

> That does not look like the full log. Can you put the complete log,
> right from the start to the end, in the PasteBin?
>
> Also, it appears that both the legs are to the same IP. Is that by design?
>
> Regards
> HASSAN
>
>
> On 2010-10-06, Pete Kelly <pkelly at gmail.com> wrote:
> > After the call is set up, Freeswitch sends a reINVITE to legA, with
> legB's
> > media IP/port, then vice versa.
> >
> > Then it sends a BYE to both legs immediately .
> >
> > However I've put a dummy while loop after the bridge:
> >
> > while(1==1) do
> >
> > end
> >
> > and the call remains up!
> >
> > However the lua script now never ends.
> >
> > On 6 October 2010 10:42, David Ponzone <david.ponzone at ipeva.fr> wrote:
> >
> >> my question is going to sound stupid but:
> >> how long before the call is ended ?
> >> are you sure A can talk to B directly ?
> >>
> >> David Ponzone  Direction Technique
> >> email: david.ponzone at ipeva.fr
> >> tel:      01 74 03 18 97
> >> gsm:   06 66 98 76 34
> >>
> >> Service Client IPeva
> >> tel:      0811 46 26 26
> >> www.ipeva.fr  -   www.ipeva-studio.com
> >>
> >> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> >> l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion
> >> non autoris?e est interdite. Tout message ?lectronique est susceptible
> >> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> >> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> >> destinataire de ce message, merci de le d?truire imm?diatement et
> >> d'avertir
> >> l'exp?diteur.*
> >> *
> >> *
> >>
> >>
> >>
> >> Le 06/10/2010 ? 11:33, Pete Kelly a ?crit :
> >>
> >> I've checked and it's PCMU for all legs, including Freeswitch... anyway
> >> it's Freeswitch which is ending the calls. I would expect leg A or leg B
> >> to
> >> end the call if it was a codec issue.
> >>
> >> On 6 October 2010 10:26, David Ponzone <david.ponzone at ipeva.fr> wrote:
> >>
> >>> Pete,
> >>>
> >>> perhaps leg A and B can't agree on the codec ?
> >>>
> >>>  David Ponzone  Direction Technique
> >>> email: david.ponzone at ipeva.fr
> >>> tel:      01 74 03 18 97
> >>> gsm:   06 66 98 76 34
> >>>
> >>> Service Client IPeva
> >>> tel:      0811 46 26 26
> >>> www.ipeva.fr  -   www.ipeva-studio.com
> >>>
> >>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
> ?
> >>> l'intention exclusive de ses destinataires. Toute utilisation ou
> >>> diffusion
> >>> non autoris?e est interdite. Tout message ?lectronique est susceptible
> >>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> >>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> >>> destinataire de ce message, merci de le d?truire imm?diatement et
> >>> d'avertir
> >>> l'exp?diteur.*
> >>> *
> >>> *
> >>>
> >>>
> >>>
> >>> Le 06/10/2010 ? 11:18, Pete Kelly a ?crit :
> >>>
> >>> Hi
> >>>
> >>> I am having some problems bridging 2 calls and bypassing the media.
> >>>
> >>> I am using a lua script executed from the command line to phone out to
> >>> the
> >>> first device (leg A), then on answer bridge a call to a second device
> >>> (leg
> >>> B).
> >>>
> >>> If I set the bypass_media=true, then freeswitch immediately sends out
> >>> BYEs
> >>> after the reINVITEs have happened.
> >>>
> >>> If I leave bypass_media unset, everything works fine except the media
> is
> >>> bridged.
> >>>
> >>> The lua script I am using is a very simple one and looks like this:
> >>>
> >>> obSession = freeswitch.Session("sofia/lpmedia/
> 91979197 at 10.15.20.122:5060
> >>> ")
> >>> obSession:setVariable('bypass_media', 'true');
> >>>
> >>> if obSession:ready() then
> >>>     -- Do something good here
> >>>
> >>>     obSession:execute("bridge", "sofia/lpmedia/2000 at 10.15.20.122:5060
> ")
> >>> end
> >>>
> >>> I have tried variations on this including, putting [bypass_media=true]
> >>> on
> >>> leg A, and using the [bypass_media_after_bridge=true] on leg B
> >>>
> >>> The logs say this:
> >>>  ...
> >>> 2010-10-06 10:10:05.847278 [NOTICE] sofia.c:5085 Channel
> [sofia/lpmedia/
> >>> 2000 at 10.15.20.122:5060] has been answered
> >>> 2010-10-06 10:10:06.650155 [NOTICE] switch_cpp.cpp:976 Hangup
> >>> sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_HIBERNATE]
> [NORMAL_CLEARING]
> >>> 2010-10-06 10:10:06.650155 [NOTICE] switch_ivr_bridge.c:936 Hangup
> >>> sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_HIBERNATE] [NORMAL_CLEARING]
> >>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1228 Session
> 27
> >>> (sofia/lpmedia/91979197 at 10.15.20.122:5060) Ended
> >>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close
> >>> Channel sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_DESTROY]
> >>> freeswitch at pete-desktop> 2010-10-06 10:10:06.651420 [NOTICE]
> >>> switch_core_session.c:1228 Session 28 (sofia/lpmedia/
> >>> 2000 at 10.15.20.122:5060) Ended
> >>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close
> >>> Channel sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_DESTROY]
> >>>
> >>>
> >>> Does anyone have any ideas as to why freeswitch may be ending the
> calls?
> >>>
> >>> I am using the latest git checkout - the behaviour was slightly
> different
> >>> with the 1.06 package. The call would setup but the reINVITEs did not
> >>> happen
> >>> as they should.
> >>>
> >>> Pete
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
>
> --
> Sent from my mobile device
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From wasim at convergence.pk  Wed Oct  6 03:28:52 2010
From: wasim at convergence.pk (Wasim Baig)
Date: Wed, 6 Oct 2010 15:28:52 +0500
Subject: [Freeswitch-users] Problem bridging 2 calls with
	bypass_media=true
In-Reply-To: <AANLkTimZbh6xHidmLQ5DAtRsXezY69XCXbnCyxB6tOYQ@mail.gmail.com>
References: <AANLkTinBT8MJAqHX3yq8u1TO9XFNn1+7zAgGuAQ2+JqZ@mail.gmail.com>
	<74D51A69-A897-43FC-9581-FB31E39E0527@ipeva.fr>
	<AANLkTimzOMMPw6KfsJ_imqQQw39zUvtFH0iX_vhYXbjp@mail.gmail.com>
	<7DB4D4EE-77F9-4353-A561-8F6EB8D99846@ipeva.fr>
	<AANLkTikhspWFyZAJh6Zi1v++bJdqBwPBkr-FQozWkuOi@mail.gmail.com>
	<AANLkTimZbh6xHidmLQ5DAtRsXezY69XCXbnCyxB6tOYQ@mail.gmail.com>
Message-ID: <AANLkTi=SY+Hs-SnO490Tcgvui+J2+f-quUWjDfBOjTcy@mail.gmail.com>

Perhaps you've got it setup async, in which case nothing blocks FS from
going through the rest of the script after the bridge.
You need to tell Lua to watch out for a hangup on either leg after the
bridge before continuing with the script.
Or do it sync.

-wasim

On Wed, Oct 6, 2010 at 15:21, Nyamul Hassan <mnhassan at usa.net> wrote:

> That does not look like the full log. Can you put the complete log,
> right from the start to the end, in the PasteBin?
>
> Also, it appears that both the legs are to the same IP. Is that by design?
>
> Regards
> HASSAN
>
>
> On 2010-10-06, Pete Kelly <pkelly at gmail.com> wrote:
> > After the call is set up, Freeswitch sends a reINVITE to legA, with
> legB's
> > media IP/port, then vice versa.
> >
> > Then it sends a BYE to both legs immediately .
> >
> > However I've put a dummy while loop after the bridge:
> >
> > while(1==1) do
> >
> > end
> >
> > and the call remains up!
> >
> > However the lua script now never ends.
> >
> > On 6 October 2010 10:42, David Ponzone <david.ponzone at ipeva.fr> wrote:
> >
> >> my question is going to sound stupid but:
> >> how long before the call is ended ?
> >> are you sure A can talk to B directly ?
> >>
> >> David Ponzone  Direction Technique
> >> email: david.ponzone at ipeva.fr
> >> tel:      01 74 03 18 97
> >> gsm:   06 66 98 76 34
> >>
> >> Service Client IPeva
> >> tel:      0811 46 26 26
> >> www.ipeva.fr  -   www.ipeva-studio.com
> >>
> >> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> >> l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion
> >> non autoris?e est interdite. Tout message ?lectronique est susceptible
> >> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> >> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> >> destinataire de ce message, merci de le d?truire imm?diatement et
> >> d'avertir
> >> l'exp?diteur.*
> >> *
> >> *
> >>
> >>
> >>
> >> Le 06/10/2010 ? 11:33, Pete Kelly a ?crit :
> >>
> >> I've checked and it's PCMU for all legs, including Freeswitch... anyway
> >> it's Freeswitch which is ending the calls. I would expect leg A or leg B
> >> to
> >> end the call if it was a codec issue.
> >>
> >> On 6 October 2010 10:26, David Ponzone <david.ponzone at ipeva.fr> wrote:
> >>
> >>> Pete,
> >>>
> >>> perhaps leg A and B can't agree on the codec ?
> >>>
> >>>  David Ponzone  Direction Technique
> >>> email: david.ponzone at ipeva.fr
> >>> tel:      01 74 03 18 97
> >>> gsm:   06 66 98 76 34
> >>>
> >>> Service Client IPeva
> >>> tel:      0811 46 26 26
> >>> www.ipeva.fr  -   www.ipeva-studio.com
> >>>
> >>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
> ?
> >>> l'intention exclusive de ses destinataires. Toute utilisation ou
> >>> diffusion
> >>> non autoris?e est interdite. Tout message ?lectronique est susceptible
> >>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> >>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> >>> destinataire de ce message, merci de le d?truire imm?diatement et
> >>> d'avertir
> >>> l'exp?diteur.*
> >>> *
> >>> *
> >>>
> >>>
> >>>
> >>> Le 06/10/2010 ? 11:18, Pete Kelly a ?crit :
> >>>
> >>> Hi
> >>>
> >>> I am having some problems bridging 2 calls and bypassing the media.
> >>>
> >>> I am using a lua script executed from the command line to phone out to
> >>> the
> >>> first device (leg A), then on answer bridge a call to a second device
> >>> (leg
> >>> B).
> >>>
> >>> If I set the bypass_media=true, then freeswitch immediately sends out
> >>> BYEs
> >>> after the reINVITEs have happened.
> >>>
> >>> If I leave bypass_media unset, everything works fine except the media
> is
> >>> bridged.
> >>>
> >>> The lua script I am using is a very simple one and looks like this:
> >>>
> >>> obSession = freeswitch.Session("sofia/lpmedia/
> 91979197 at 10.15.20.122:5060
> >>> ")
> >>> obSession:setVariable('bypass_media', 'true');
> >>>
> >>> if obSession:ready() then
> >>>     -- Do something good here
> >>>
> >>>     obSession:execute("bridge", "sofia/lpmedia/2000 at 10.15.20.122:5060
> ")
> >>> end
> >>>
> >>> I have tried variations on this including, putting [bypass_media=true]
> >>> on
> >>> leg A, and using the [bypass_media_after_bridge=true] on leg B
> >>>
> >>> The logs say this:
> >>>  ...
> >>> 2010-10-06 10:10:05.847278 [NOTICE] sofia.c:5085 Channel
> [sofia/lpmedia/
> >>> 2000 at 10.15.20.122:5060] has been answered
> >>> 2010-10-06 10:10:06.650155 [NOTICE] switch_cpp.cpp:976 Hangup
> >>> sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_HIBERNATE]
> [NORMAL_CLEARING]
> >>> 2010-10-06 10:10:06.650155 [NOTICE] switch_ivr_bridge.c:936 Hangup
> >>> sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_HIBERNATE] [NORMAL_CLEARING]
> >>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1228 Session
> 27
> >>> (sofia/lpmedia/91979197 at 10.15.20.122:5060) Ended
> >>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close
> >>> Channel sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_DESTROY]
> >>> freeswitch at pete-desktop> 2010-10-06 10:10:06.651420 [NOTICE]
> >>> switch_core_session.c:1228 Session 28 (sofia/lpmedia/
> >>> 2000 at 10.15.20.122:5060) Ended
> >>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close
> >>> Channel sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_DESTROY]
> >>>
> >>>
> >>> Does anyone have any ideas as to why freeswitch may be ending the
> calls?
> >>>
> >>> I am using the latest git checkout - the behaviour was slightly
> different
> >>> with the 1.06 package. The call would setup but the reINVITEs did not
> >>> happen
> >>> as they should.
> >>>
> >>> Pete
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
>
> --
> Sent from my mobile device
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
peace be upon you ...
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From pkelly at gmail.com  Wed Oct  6 03:31:16 2010
From: pkelly at gmail.com (Pete Kelly)
Date: Wed, 6 Oct 2010 11:31:16 +0100
Subject: [Freeswitch-users] Problem bridging 2 calls with
	bypass_media=true
In-Reply-To: <85D2911B-02CF-42BC-9297-99B45AE95E92@ipeva.fr>
References: <AANLkTinBT8MJAqHX3yq8u1TO9XFNn1+7zAgGuAQ2+JqZ@mail.gmail.com>
	<74D51A69-A897-43FC-9581-FB31E39E0527@ipeva.fr>
	<AANLkTimzOMMPw6KfsJ_imqQQw39zUvtFH0iX_vhYXbjp@mail.gmail.com>
	<7DB4D4EE-77F9-4353-A561-8F6EB8D99846@ipeva.fr>
	<AANLkTikhspWFyZAJh6Zi1v++bJdqBwPBkr-FQozWkuOi@mail.gmail.com>
	<85D2911B-02CF-42BC-9297-99B45AE95E92@ipeva.fr>
Message-ID: <AANLkTi=_z5ewAh0YYEzRaLV8y8ZQ7rJzN6JYTVry24ZX@mail.gmail.com>

David, I'm not sure it is possible to originate 2 calls from dialplan and
then bridge them. I could be wrong though!

On 6 October 2010 11:08, David Ponzone <david.ponzone at ipeva.fr> wrote:

> Can you just do the same bridge from dialplan, just to know if it's related
> to LUA ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 06/10/2010 ? 11:52, Pete Kelly a ?crit :
>
> After the call is set up, Freeswitch sends a reINVITE to legA, with legB's
> media IP/port, then vice versa.
>
> Then it sends a BYE to both legs immediately .
>
> However I've put a dummy while loop after the bridge:
>
> while(1==1) do
>
> end
>
> and the call remains up!
>
> However the lua script now never ends.
>
> On 6 October 2010 10:42, David Ponzone <david.ponzone at ipeva.fr> wrote:
>
>> my question is going to sound stupid but:
>> how long before the call is ended ?
>> are you sure A can talk to B directly ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 06/10/2010 ? 11:33, Pete Kelly a ?crit :
>>
>> I've checked and it's PCMU for all legs, including Freeswitch... anyway
>> it's Freeswitch which is ending the calls. I would expect leg A or leg B to
>> end the call if it was a codec issue.
>>
>> On 6 October 2010 10:26, David Ponzone <david.ponzone at ipeva.fr> wrote:
>>
>>> Pete,
>>>
>>> perhaps leg A and B can't agree on the codec ?
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 06/10/2010 ? 11:18, Pete Kelly a ?crit :
>>>
>>> Hi
>>>
>>> I am having some problems bridging 2 calls and bypassing the media.
>>>
>>> I am using a lua script executed from the command line to phone out to
>>> the first device (leg A), then on answer bridge a call to a second device
>>> (leg B).
>>>
>>> If I set the bypass_media=true, then freeswitch immediately sends out
>>> BYEs after the reINVITEs have happened.
>>>
>>> If I leave bypass_media unset, everything works fine except the media is
>>> bridged.
>>>
>>> The lua script I am using is a very simple one and looks like this:
>>>
>>> obSession = freeswitch.Session("sofia/lpmedia/91979197 at 10.15.20.122:5060
>>> ")
>>> obSession:setVariable('bypass_media', 'true');
>>>
>>> if obSession:ready() then
>>>     -- Do something good here
>>>
>>>     obSession:execute("bridge", "sofia/lpmedia/2000 at 10.15.20.122:5060")
>>> end
>>>
>>> I have tried variations on this including, putting [bypass_media=true]
>>>  on leg A, and using the [bypass_media_after_bridge=true] on leg B
>>>
>>> The logs say this:
>>>  ...
>>> 2010-10-06 10:10:05.847278 [NOTICE] sofia.c:5085 Channel [sofia/lpmedia/
>>> 2000 at 10.15.20.122:5060] has been answered
>>> 2010-10-06 10:10:06.650155 [NOTICE] switch_cpp.cpp:976 Hangup
>>> sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_HIBERNATE]
>>> [NORMAL_CLEARING]
>>> 2010-10-06 10:10:06.650155 [NOTICE] switch_ivr_bridge.c:936 Hangup
>>> sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_HIBERNATE] [NORMAL_CLEARING]
>>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1228 Session 27
>>> (sofia/lpmedia/91979197 at 10.15.20.122:5060) Ended
>>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close
>>> Channel sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_DESTROY]
>>> freeswitch at pete-desktop> 2010-10-06 10:10:06.651420 [NOTICE]
>>> switch_core_session.c:1228 Session 28 (sofia/lpmedia/
>>> 2000 at 10.15.20.122:5060) Ended
>>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close
>>> Channel sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_DESTROY]
>>>
>>>
>>> Does anyone have any ideas as to why freeswitch may be ending the calls?
>>>
>>> I am using the latest git checkout - the behaviour was slightly different
>>> with the 1.06 package. The call would setup but the reINVITEs did not happen
>>> as they should.
>>>
>>> Pete
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pkelly at gmail.com  Wed Oct  6 03:38:27 2010
From: pkelly at gmail.com (Pete Kelly)
Date: Wed, 6 Oct 2010 11:38:27 +0100
Subject: [Freeswitch-users] Problem bridging 2 calls with
	bypass_media=true
In-Reply-To: <AANLkTi=SY+Hs-SnO490Tcgvui+J2+f-quUWjDfBOjTcy@mail.gmail.com>
References: <AANLkTinBT8MJAqHX3yq8u1TO9XFNn1+7zAgGuAQ2+JqZ@mail.gmail.com>
	<74D51A69-A897-43FC-9581-FB31E39E0527@ipeva.fr>
	<AANLkTimzOMMPw6KfsJ_imqQQw39zUvtFH0iX_vhYXbjp@mail.gmail.com>
	<7DB4D4EE-77F9-4353-A561-8F6EB8D99846@ipeva.fr>
	<AANLkTikhspWFyZAJh6Zi1v++bJdqBwPBkr-FQozWkuOi@mail.gmail.com>
	<AANLkTimZbh6xHidmLQ5DAtRsXezY69XCXbnCyxB6tOYQ@mail.gmail.com>
	<AANLkTi=SY+Hs-SnO490Tcgvui+J2+f-quUWjDfBOjTcy@mail.gmail.com>
Message-ID: <AANLkTinu66vRV+fHi0rckcjQc=5Uk9c84t1ZMJN26AuD@mail.gmail.com>

On 6 October 2010 11:28, Wasim Baig <wasim at convergence.pk> wrote:

> Perhaps you've got it setup async, in which case nothing blocks FS from
> going through the rest of the script after the bridge.
> You need to tell Lua to watch out for a hangup on either leg after the
> bridge before continuing with the script.
> Or do it sync.
>

That sounds like exactly what is happening, as the while loop prevents the
hangup.

How would you suggest I watch for the hangup? If I check the output
of obSession:hangupCause() and obSession:hangupState() within the while loop
they are always both set to SUCCESS (during the call, and after hangup).

Hangup hook maybe?


>
> -wasim
>
>
> On Wed, Oct 6, 2010 at 15:21, Nyamul Hassan <mnhassan at usa.net> wrote:
>
>> That does not look like the full log. Can you put the complete log,
>> right from the start to the end, in the PasteBin?
>>
>> Also, it appears that both the legs are to the same IP. Is that by design?
>>
>> Regards
>> HASSAN
>>
>>
>> On 2010-10-06, Pete Kelly <pkelly at gmail.com> wrote:
>> > After the call is set up, Freeswitch sends a reINVITE to legA, with
>> legB's
>> > media IP/port, then vice versa.
>> >
>> > Then it sends a BYE to both legs immediately .
>> >
>> > However I've put a dummy while loop after the bridge:
>> >
>> > while(1==1) do
>> >
>> > end
>> >
>> > and the call remains up!
>> >
>> > However the lua script now never ends.
>> >
>> > On 6 October 2010 10:42, David Ponzone <david.ponzone at ipeva.fr> wrote:
>> >
>> >> my question is going to sound stupid but:
>> >> how long before the call is ended ?
>> >> are you sure A can talk to B directly ?
>> >>
>> >> David Ponzone  Direction Technique
>> >> email: david.ponzone at ipeva.fr
>> >> tel:      01 74 03 18 97
>> >> gsm:   06 66 98 76 34
>> >>
>> >> Service Client IPeva
>> >> tel:      0811 46 26 26
>> >> www.ipeva.fr  -   www.ipeva-studio.com
>> >>
>> >> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>> ?
>> >> l'intention exclusive de ses destinataires. Toute utilisation ou
>> diffusion
>> >> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> >> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> >> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> >> destinataire de ce message, merci de le d?truire imm?diatement et
>> >> d'avertir
>> >> l'exp?diteur.*
>> >> *
>> >> *
>> >>
>> >>
>> >>
>> >> Le 06/10/2010 ? 11:33, Pete Kelly a ?crit :
>> >>
>> >> I've checked and it's PCMU for all legs, including Freeswitch... anyway
>> >> it's Freeswitch which is ending the calls. I would expect leg A or leg
>> B
>> >> to
>> >> end the call if it was a codec issue.
>> >>
>> >> On 6 October 2010 10:26, David Ponzone <david.ponzone at ipeva.fr> wrote:
>> >>
>> >>> Pete,
>> >>>
>> >>> perhaps leg A and B can't agree on the codec ?
>> >>>
>> >>>  David Ponzone  Direction Technique
>> >>> email: david.ponzone at ipeva.fr
>> >>> tel:      01 74 03 18 97
>> >>> gsm:   06 66 98 76 34
>> >>>
>> >>> Service Client IPeva
>> >>> tel:      0811 46 26 26
>> >>> www.ipeva.fr  -   www.ipeva-studio.com
>> >>>
>> >>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>> ?
>> >>> l'intention exclusive de ses destinataires. Toute utilisation ou
>> >>> diffusion
>> >>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> >>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> >>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> >>> destinataire de ce message, merci de le d?truire imm?diatement et
>> >>> d'avertir
>> >>> l'exp?diteur.*
>> >>> *
>> >>> *
>> >>>
>> >>>
>> >>>
>> >>> Le 06/10/2010 ? 11:18, Pete Kelly a ?crit :
>> >>>
>> >>> Hi
>> >>>
>> >>> I am having some problems bridging 2 calls and bypassing the media.
>> >>>
>> >>> I am using a lua script executed from the command line to phone out to
>> >>> the
>> >>> first device (leg A), then on answer bridge a call to a second device
>> >>> (leg
>> >>> B).
>> >>>
>> >>> If I set the bypass_media=true, then freeswitch immediately sends out
>> >>> BYEs
>> >>> after the reINVITEs have happened.
>> >>>
>> >>> If I leave bypass_media unset, everything works fine except the media
>> is
>> >>> bridged.
>> >>>
>> >>> The lua script I am using is a very simple one and looks like this:
>> >>>
>> >>> obSession = freeswitch.Session("sofia/lpmedia/
>> 91979197 at 10.15.20.122:5060
>> >>> ")
>> >>> obSession:setVariable('bypass_media', 'true');
>> >>>
>> >>> if obSession:ready() then
>> >>>     -- Do something good here
>> >>>
>> >>>     obSession:execute("bridge", "sofia/lpmedia/2000 at 10.15.20.122:5060
>> ")
>> >>> end
>> >>>
>> >>> I have tried variations on this including, putting [bypass_media=true]
>> >>> on
>> >>> leg A, and using the [bypass_media_after_bridge=true] on leg B
>> >>>
>> >>> The logs say this:
>> >>>  ...
>> >>> 2010-10-06 10:10:05.847278 [NOTICE] sofia.c:5085 Channel
>> [sofia/lpmedia/
>> >>> 2000 at 10.15.20.122:5060] has been answered
>> >>> 2010-10-06 10:10:06.650155 [NOTICE] switch_cpp.cpp:976 Hangup
>> >>> sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_HIBERNATE]
>> [NORMAL_CLEARING]
>> >>> 2010-10-06 10:10:06.650155 [NOTICE] switch_ivr_bridge.c:936 Hangup
>> >>> sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_HIBERNATE] [NORMAL_CLEARING]
>> >>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1228 Session
>> 27
>> >>> (sofia/lpmedia/91979197 at 10.15.20.122:5060) Ended
>> >>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close
>> >>> Channel sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_DESTROY]
>> >>> freeswitch at pete-desktop> 2010-10-06 10:10:06.651420 [NOTICE]
>> >>> switch_core_session.c:1228 Session 28 (sofia/lpmedia/
>> >>> 2000 at 10.15.20.122:5060) Ended
>> >>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close
>> >>> Channel sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_DESTROY]
>> >>>
>> >>>
>> >>> Does anyone have any ideas as to why freeswitch may be ending the
>> calls?
>> >>>
>> >>> I am using the latest git checkout - the behaviour was slightly
>> different
>> >>> with the 1.06 package. The call would setup but the reINVITEs did not
>> >>> happen
>> >>> as they should.
>> >>>
>> >>> Pete
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>>
>> --
>> Sent from my mobile device
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
> peace be upon you ...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pkelly at gmail.com  Wed Oct  6 03:55:22 2010
From: pkelly at gmail.com (Pete Kelly)
Date: Wed, 6 Oct 2010 11:55:22 +0100
Subject: [Freeswitch-users] Problem bridging 2 calls with
	bypass_media=true
In-Reply-To: <AANLkTinu66vRV+fHi0rckcjQc=5Uk9c84t1ZMJN26AuD@mail.gmail.com>
References: <AANLkTinBT8MJAqHX3yq8u1TO9XFNn1+7zAgGuAQ2+JqZ@mail.gmail.com>
	<74D51A69-A897-43FC-9581-FB31E39E0527@ipeva.fr>
	<AANLkTimzOMMPw6KfsJ_imqQQw39zUvtFH0iX_vhYXbjp@mail.gmail.com>
	<7DB4D4EE-77F9-4353-A561-8F6EB8D99846@ipeva.fr>
	<AANLkTikhspWFyZAJh6Zi1v++bJdqBwPBkr-FQozWkuOi@mail.gmail.com>
	<AANLkTimZbh6xHidmLQ5DAtRsXezY69XCXbnCyxB6tOYQ@mail.gmail.com>
	<AANLkTi=SY+Hs-SnO490Tcgvui+J2+f-quUWjDfBOjTcy@mail.gmail.com>
	<AANLkTinu66vRV+fHi0rckcjQc=5Uk9c84t1ZMJN26AuD@mail.gmail.com>
Message-ID: <AANLkTimy4KYZYuyH=9EmzMUwOXPDZ5qQXtM29LdPSWgf@mail.gmail.com>

This seems to do the trick, unless anyone can suggest anything more elegant
than the while loop?

obSession = freeswitch.Session("sofia/lpmedia/91979197 at 10.15.20.122:5060")
obSession:setVariable('bypass_media', 'true');

function myHangupHook()
freeswitch.consoleLog("INFO", "hungup\n");
end

if obSession:ready() then

    obSession:setHangupHook("myHangupHook")

    obSession:execute("bridge", "sofia/lpmedia/2000 at 10.15.20.122:5060")

    while (obSession:ready() == true) do
     obSession:sleep("1000");
    end
end



On 6 October 2010 11:38, Pete Kelly <pkelly at gmail.com> wrote:

>
>
> On 6 October 2010 11:28, Wasim Baig <wasim at convergence.pk> wrote:
>
>> Perhaps you've got it setup async, in which case nothing blocks FS from
>> going through the rest of the script after the bridge.
>> You need to tell Lua to watch out for a hangup on either leg after the
>> bridge before continuing with the script.
>> Or do it sync.
>>
>
> That sounds like exactly what is happening, as the while loop prevents the
> hangup.
>
> How would you suggest I watch for the hangup? If I check the output
> of obSession:hangupCause() and obSession:hangupState() within the while loop
> they are always both set to SUCCESS (during the call, and after hangup).
>
> Hangup hook maybe?
>
>
>>
>> -wasim
>>
>>
>> On Wed, Oct 6, 2010 at 15:21, Nyamul Hassan <mnhassan at usa.net> wrote:
>>
>>> That does not look like the full log. Can you put the complete log,
>>> right from the start to the end, in the PasteBin?
>>>
>>> Also, it appears that both the legs are to the same IP. Is that by
>>> design?
>>>
>>> Regards
>>> HASSAN
>>>
>>>
>>> On 2010-10-06, Pete Kelly <pkelly at gmail.com> wrote:
>>> > After the call is set up, Freeswitch sends a reINVITE to legA, with
>>> legB's
>>> > media IP/port, then vice versa.
>>> >
>>> > Then it sends a BYE to both legs immediately .
>>> >
>>> > However I've put a dummy while loop after the bridge:
>>> >
>>> > while(1==1) do
>>> >
>>> > end
>>> >
>>> > and the call remains up!
>>> >
>>> > However the lua script now never ends.
>>> >
>>> > On 6 October 2010 10:42, David Ponzone <david.ponzone at ipeva.fr> wrote:
>>> >
>>> >> my question is going to sound stupid but:
>>> >> how long before the call is ended ?
>>> >> are you sure A can talk to B directly ?
>>> >>
>>> >> David Ponzone  Direction Technique
>>> >> email: david.ponzone at ipeva.fr
>>> >> tel:      01 74 03 18 97
>>> >> gsm:   06 66 98 76 34
>>> >>
>>> >> Service Client IPeva
>>> >> tel:      0811 46 26 26
>>> >> www.ipeva.fr  -   www.ipeva-studio.com
>>> >>
>>> >> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>> ?
>>> >> l'intention exclusive de ses destinataires. Toute utilisation ou
>>> diffusion
>>> >> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> >> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> >> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> >> destinataire de ce message, merci de le d?truire imm?diatement et
>>> >> d'avertir
>>> >> l'exp?diteur.*
>>> >> *
>>> >> *
>>> >>
>>> >>
>>> >>
>>> >> Le 06/10/2010 ? 11:33, Pete Kelly a ?crit :
>>> >>
>>> >> I've checked and it's PCMU for all legs, including Freeswitch...
>>> anyway
>>> >> it's Freeswitch which is ending the calls. I would expect leg A or leg
>>> B
>>> >> to
>>> >> end the call if it was a codec issue.
>>> >>
>>> >> On 6 October 2010 10:26, David Ponzone <david.ponzone at ipeva.fr>
>>> wrote:
>>> >>
>>> >>> Pete,
>>> >>>
>>> >>> perhaps leg A and B can't agree on the codec ?
>>> >>>
>>> >>>  David Ponzone  Direction Technique
>>> >>> email: david.ponzone at ipeva.fr
>>> >>> tel:      01 74 03 18 97
>>> >>> gsm:   06 66 98 76 34
>>> >>>
>>> >>> Service Client IPeva
>>> >>> tel:      0811 46 26 26
>>> >>> www.ipeva.fr  -   www.ipeva-studio.com
>>> >>>
>>> >>> *Ce message et toutes les pi?ces jointes sont confidentiels et
>>> ?tablis ?
>>> >>> l'intention exclusive de ses destinataires. Toute utilisation ou
>>> >>> diffusion
>>> >>> non autoris?e est interdite. Tout message ?lectronique est
>>> susceptible
>>> >>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> >>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> >>> destinataire de ce message, merci de le d?truire imm?diatement et
>>> >>> d'avertir
>>> >>> l'exp?diteur.*
>>> >>> *
>>> >>> *
>>> >>>
>>> >>>
>>> >>>
>>> >>> Le 06/10/2010 ? 11:18, Pete Kelly a ?crit :
>>> >>>
>>> >>> Hi
>>> >>>
>>> >>> I am having some problems bridging 2 calls and bypassing the media.
>>> >>>
>>> >>> I am using a lua script executed from the command line to phone out
>>> to
>>> >>> the
>>> >>> first device (leg A), then on answer bridge a call to a second device
>>> >>> (leg
>>> >>> B).
>>> >>>
>>> >>> If I set the bypass_media=true, then freeswitch immediately sends out
>>> >>> BYEs
>>> >>> after the reINVITEs have happened.
>>> >>>
>>> >>> If I leave bypass_media unset, everything works fine except the media
>>> is
>>> >>> bridged.
>>> >>>
>>> >>> The lua script I am using is a very simple one and looks like this:
>>> >>>
>>> >>> obSession = freeswitch.Session("sofia/lpmedia/
>>> 91979197 at 10.15.20.122:5060
>>> >>> ")
>>> >>> obSession:setVariable('bypass_media', 'true');
>>> >>>
>>> >>> if obSession:ready() then
>>> >>>     -- Do something good here
>>> >>>
>>> >>>     obSession:execute("bridge", "sofia/lpmedia/
>>> 2000 at 10.15.20.122:5060")
>>> >>> end
>>> >>>
>>> >>> I have tried variations on this including, putting
>>> [bypass_media=true]
>>> >>> on
>>> >>> leg A, and using the [bypass_media_after_bridge=true] on leg B
>>> >>>
>>> >>> The logs say this:
>>> >>>  ...
>>> >>> 2010-10-06 10:10:05.847278 [NOTICE] sofia.c:5085 Channel
>>> [sofia/lpmedia/
>>> >>> 2000 at 10.15.20.122:5060] has been answered
>>> >>> 2010-10-06 10:10:06.650155 [NOTICE] switch_cpp.cpp:976 Hangup
>>> >>> sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_HIBERNATE]
>>> [NORMAL_CLEARING]
>>> >>> 2010-10-06 10:10:06.650155 [NOTICE] switch_ivr_bridge.c:936 Hangup
>>> >>> sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_HIBERNATE]
>>> [NORMAL_CLEARING]
>>> >>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1228
>>> Session 27
>>> >>> (sofia/lpmedia/91979197 at 10.15.20.122:5060) Ended
>>> >>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close
>>> >>> Channel sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_DESTROY]
>>> >>> freeswitch at pete-desktop> 2010-10-06 10:10:06.651420 [NOTICE]
>>> >>> switch_core_session.c:1228 Session 28 (sofia/lpmedia/
>>> >>> 2000 at 10.15.20.122:5060) Ended
>>> >>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close
>>> >>> Channel sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_DESTROY]
>>> >>>
>>> >>>
>>> >>> Does anyone have any ideas as to why freeswitch may be ending the
>>> calls?
>>> >>>
>>> >>> I am using the latest git checkout - the behaviour was slightly
>>> different
>>> >>> with the 1.06 package. The call would setup but the reINVITEs did not
>>> >>> happen
>>> >>> as they should.
>>> >>>
>>> >>> Pete
>>> >>>
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>>
>>> >>>
>>> >>>
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>>
>>> >>>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >>
>>> >
>>>
>>> --
>>> Sent from my mobile device
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
>> peace be upon you ...
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From ale975 at gmail.com  Wed Oct  6 02:10:40 2010
From: ale975 at gmail.com (Ale)
Date: Wed, 6 Oct 2010 11:10:40 +0200
Subject: [Freeswitch-users] Fax hint request
Message-ID: <AANLkTim8NhxTcypcA9kuT4vnmRNObZ2JbZzgVNdGpsgW@mail.gmail.com>

Hi all,

I've compiled freeswitch from git, configured an external trunk and
added enabled-t38=true in fax.conf.
Both servers have a public ip. Tring to send a fax from cli via
origiante all t.38 fax fail with

2010-10-05 19:48:14.228734 [NOTICE] mod_sofia.c:880 Hangup
sofia/external/xxxxxx at ip [CS_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE]
...
2010-10-05 19:48:14.228734 [DEBUG] mod_spandsp_fax.c:335 Fax
processing not successful - result (49) The call dropped prematurely.

Reading documentation i've found that RECOVERY_ON_TIMER_EXPIRE can be
caused by nat configuration problem, but i think configuration could
be reasonably  correct.

sip flow: http://i56.tinypic.com/30j1ezn.png
remote trying after the initial invite of fs: http://i52.tinypic.com/255hlps.png
remote 183 session in progress: http://i54.tinypic.com/293jew2.png

After 183 fs wait some time, probably a timeout, and send a cancel request.

On fs cli i see also the data below, but fs never send to remote trunk.
2010-10-05 19:48:04.235005 [DEBUG] sofia_glue.c:137
sofia/external/xxxx at ip image media sdp:
v=0
o=FreeSWITCH 1286300921 1286300923 IN IP4 213.187.9.30
s=FreeSWITCH
c=IN IP4 xxx.xxx.9.30
t=0 0
m=image 21562 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:2000
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPRedundancy

Can anyone give me a hint, if i'm wrong something?

Many thx,
Ale


From pkelly at gmail.com  Wed Oct  6 06:00:47 2010
From: pkelly at gmail.com (Pete Kelly)
Date: Wed, 6 Oct 2010 14:00:47 +0100
Subject: [Freeswitch-users] Problem bridging 2 calls with
	bypass_media=true
In-Reply-To: <AANLkTimy4KYZYuyH=9EmzMUwOXPDZ5qQXtM29LdPSWgf@mail.gmail.com>
References: <AANLkTinBT8MJAqHX3yq8u1TO9XFNn1+7zAgGuAQ2+JqZ@mail.gmail.com>
	<74D51A69-A897-43FC-9581-FB31E39E0527@ipeva.fr>
	<AANLkTimzOMMPw6KfsJ_imqQQw39zUvtFH0iX_vhYXbjp@mail.gmail.com>
	<7DB4D4EE-77F9-4353-A561-8F6EB8D99846@ipeva.fr>
	<AANLkTikhspWFyZAJh6Zi1v++bJdqBwPBkr-FQozWkuOi@mail.gmail.com>
	<AANLkTimZbh6xHidmLQ5DAtRsXezY69XCXbnCyxB6tOYQ@mail.gmail.com>
	<AANLkTi=SY+Hs-SnO490Tcgvui+J2+f-quUWjDfBOjTcy@mail.gmail.com>
	<AANLkTinu66vRV+fHi0rckcjQc=5Uk9c84t1ZMJN26AuD@mail.gmail.com>
	<AANLkTimy4KYZYuyH=9EmzMUwOXPDZ5qQXtM29LdPSWgf@mail.gmail.com>
Message-ID: <AANLkTinopvA02xdbGh4zafu44m6zNHTQ5beh9EhSy4Em@mail.gmail.com>

Sorry to be a pain - does anyone have a more elegant suggestion (or could
this be a bug?)

In the while loop I have below, as soon as the call is hung up the obSession
variable seems to be destroyed completely so the lua script crashes when it
tries to call obSession:ready() with the following error:

010-10-06 13:57:14.082645 [ERR] mod_lua.cpp:182 attempt to call a nil value
stack traceback:
        [C]: in function 'ready'
        /usr/local/freeswitch/scripts/click2call_call.lua:58: in main chunk


On 6 October 2010 11:55, Pete Kelly <pkelly at gmail.com> wrote:

> This seems to do the trick, unless anyone can suggest anything more elegant
> than the while loop?
>
> obSession = freeswitch.Session("sofia/lpmedia/91979197 at 10.15.20.122:5060")
> obSession:setVariable('bypass_media', 'true');
>
> function myHangupHook()
> freeswitch.consoleLog("INFO", "hungup\n");
> end
>
> if obSession:ready() then
>
>     obSession:setHangupHook("myHangupHook")
>
>     obSession:execute("bridge", "sofia/lpmedia/2000 at 10.15.20.122:5060")
>
>     while (obSession:ready() == true) do
>      obSession:sleep("1000");
>     end
> end
>
>
>
> On 6 October 2010 11:38, Pete Kelly <pkelly at gmail.com> wrote:
>
>>
>>
>> On 6 October 2010 11:28, Wasim Baig <wasim at convergence.pk> wrote:
>>
>>> Perhaps you've got it setup async, in which case nothing blocks FS from
>>> going through the rest of the script after the bridge.
>>> You need to tell Lua to watch out for a hangup on either leg after the
>>> bridge before continuing with the script.
>>> Or do it sync.
>>>
>>
>> That sounds like exactly what is happening, as the while loop prevents the
>> hangup.
>>
>> How would you suggest I watch for the hangup? If I check the output
>> of obSession:hangupCause() and obSession:hangupState() within the while loop
>> they are always both set to SUCCESS (during the call, and after hangup).
>>
>> Hangup hook maybe?
>>
>>
>>>
>>> -wasim
>>>
>>>
>>> On Wed, Oct 6, 2010 at 15:21, Nyamul Hassan <mnhassan at usa.net> wrote:
>>>
>>>> That does not look like the full log. Can you put the complete log,
>>>> right from the start to the end, in the PasteBin?
>>>>
>>>> Also, it appears that both the legs are to the same IP. Is that by
>>>> design?
>>>>
>>>> Regards
>>>> HASSAN
>>>>
>>>>
>>>> On 2010-10-06, Pete Kelly <pkelly at gmail.com> wrote:
>>>> > After the call is set up, Freeswitch sends a reINVITE to legA, with
>>>> legB's
>>>> > media IP/port, then vice versa.
>>>> >
>>>> > Then it sends a BYE to both legs immediately .
>>>> >
>>>> > However I've put a dummy while loop after the bridge:
>>>> >
>>>> > while(1==1) do
>>>> >
>>>> > end
>>>> >
>>>> > and the call remains up!
>>>> >
>>>> > However the lua script now never ends.
>>>> >
>>>> > On 6 October 2010 10:42, David Ponzone <david.ponzone at ipeva.fr>
>>>> wrote:
>>>> >
>>>> >> my question is going to sound stupid but:
>>>> >> how long before the call is ended ?
>>>> >> are you sure A can talk to B directly ?
>>>> >>
>>>> >> David Ponzone  Direction Technique
>>>> >> email: david.ponzone at ipeva.fr
>>>> >> tel:      01 74 03 18 97
>>>> >> gsm:   06 66 98 76 34
>>>> >>
>>>> >> Service Client IPeva
>>>> >> tel:      0811 46 26 26
>>>> >> www.ipeva.fr  -   www.ipeva-studio.com
>>>> >>
>>>> >> *Ce message et toutes les pi?ces jointes sont confidentiels et
>>>> ?tablis ?
>>>> >> l'intention exclusive de ses destinataires. Toute utilisation ou
>>>> diffusion
>>>> >> non autoris?e est interdite. Tout message ?lectronique est
>>>> susceptible
>>>> >> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>>> >> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>>> >> destinataire de ce message, merci de le d?truire imm?diatement et
>>>> >> d'avertir
>>>> >> l'exp?diteur.*
>>>> >> *
>>>> >> *
>>>> >>
>>>> >>
>>>> >>
>>>> >> Le 06/10/2010 ? 11:33, Pete Kelly a ?crit :
>>>> >>
>>>> >> I've checked and it's PCMU for all legs, including Freeswitch...
>>>> anyway
>>>> >> it's Freeswitch which is ending the calls. I would expect leg A or
>>>> leg B
>>>> >> to
>>>> >> end the call if it was a codec issue.
>>>> >>
>>>> >> On 6 October 2010 10:26, David Ponzone <david.ponzone at ipeva.fr>
>>>> wrote:
>>>> >>
>>>> >>> Pete,
>>>> >>>
>>>> >>> perhaps leg A and B can't agree on the codec ?
>>>> >>>
>>>> >>>  David Ponzone  Direction Technique
>>>> >>> email: david.ponzone at ipeva.fr
>>>> >>> tel:      01 74 03 18 97
>>>> >>> gsm:   06 66 98 76 34
>>>> >>>
>>>> >>> Service Client IPeva
>>>> >>> tel:      0811 46 26 26
>>>> >>> www.ipeva.fr  -   www.ipeva-studio.com
>>>> >>>
>>>> >>> *Ce message et toutes les pi?ces jointes sont confidentiels et
>>>> ?tablis ?
>>>> >>> l'intention exclusive de ses destinataires. Toute utilisation ou
>>>> >>> diffusion
>>>> >>> non autoris?e est interdite. Tout message ?lectronique est
>>>> susceptible
>>>> >>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>>> >>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>>> >>> destinataire de ce message, merci de le d?truire imm?diatement et
>>>> >>> d'avertir
>>>> >>> l'exp?diteur.*
>>>> >>> *
>>>> >>> *
>>>> >>>
>>>> >>>
>>>> >>>
>>>> >>> Le 06/10/2010 ? 11:18, Pete Kelly a ?crit :
>>>> >>>
>>>> >>> Hi
>>>> >>>
>>>> >>> I am having some problems bridging 2 calls and bypassing the media.
>>>> >>>
>>>> >>> I am using a lua script executed from the command line to phone out
>>>> to
>>>> >>> the
>>>> >>> first device (leg A), then on answer bridge a call to a second
>>>> device
>>>> >>> (leg
>>>> >>> B).
>>>> >>>
>>>> >>> If I set the bypass_media=true, then freeswitch immediately sends
>>>> out
>>>> >>> BYEs
>>>> >>> after the reINVITEs have happened.
>>>> >>>
>>>> >>> If I leave bypass_media unset, everything works fine except the
>>>> media is
>>>> >>> bridged.
>>>> >>>
>>>> >>> The lua script I am using is a very simple one and looks like this:
>>>> >>>
>>>> >>> obSession = freeswitch.Session("sofia/lpmedia/
>>>> 91979197 at 10.15.20.122:5060
>>>> >>> ")
>>>> >>> obSession:setVariable('bypass_media', 'true');
>>>> >>>
>>>> >>> if obSession:ready() then
>>>> >>>     -- Do something good here
>>>> >>>
>>>> >>>     obSession:execute("bridge", "sofia/lpmedia/
>>>> 2000 at 10.15.20.122:5060")
>>>> >>> end
>>>> >>>
>>>> >>> I have tried variations on this including, putting
>>>> [bypass_media=true]
>>>> >>> on
>>>> >>> leg A, and using the [bypass_media_after_bridge=true] on leg B
>>>> >>>
>>>> >>> The logs say this:
>>>> >>>  ...
>>>> >>> 2010-10-06 10:10:05.847278 [NOTICE] sofia.c:5085 Channel
>>>> [sofia/lpmedia/
>>>> >>> 2000 at 10.15.20.122:5060] has been answered
>>>> >>> 2010-10-06 10:10:06.650155 [NOTICE] switch_cpp.cpp:976 Hangup
>>>> >>> sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_HIBERNATE]
>>>> [NORMAL_CLEARING]
>>>> >>> 2010-10-06 10:10:06.650155 [NOTICE] switch_ivr_bridge.c:936 Hangup
>>>> >>> sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_HIBERNATE]
>>>> [NORMAL_CLEARING]
>>>> >>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1228
>>>> Session 27
>>>> >>> (sofia/lpmedia/91979197 at 10.15.20.122:5060) Ended
>>>> >>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close
>>>> >>> Channel sofia/lpmedia/91979197 at 10.15.20.122:5060 [CS_DESTROY]
>>>> >>> freeswitch at pete-desktop> 2010-10-06 10:10:06.651420 [NOTICE]
>>>> >>> switch_core_session.c:1228 Session 28 (sofia/lpmedia/
>>>> >>> 2000 at 10.15.20.122:5060) Ended
>>>> >>> 2010-10-06 10:10:06.651420 [NOTICE] switch_core_session.c:1230 Close
>>>> >>> Channel sofia/lpmedia/2000 at 10.15.20.122:5060 [CS_DESTROY]
>>>> >>>
>>>> >>>
>>>> >>> Does anyone have any ideas as to why freeswitch may be ending the
>>>> calls?
>>>> >>>
>>>> >>> I am using the latest git checkout - the behaviour was slightly
>>>> different
>>>> >>> with the 1.06 package. The call would setup but the reINVITEs did
>>>> not
>>>> >>> happen
>>>> >>> as they should.
>>>> >>>
>>>> >>> Pete
>>>> >>>
>>>> >>> _______________________________________________
>>>> >>> FreeSWITCH-users mailing list
>>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>> http://www.freeswitch.org
>>>> >>>
>>>> >>>
>>>> >>>
>>>> >>> _______________________________________________
>>>> >>> FreeSWITCH-users mailing list
>>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>> http://www.freeswitch.org
>>>> >>>
>>>> >>>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >>
>>>> >>
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >>
>>>> >>
>>>> >
>>>>
>>>> --
>>>> Sent from my mobile device
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070
>>> | peace be upon you ...
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
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From Joshua.Foshee at LogixCom.com  Wed Oct  6 06:12:00 2010
From: Joshua.Foshee at LogixCom.com (Joshua Foshee)
Date: Wed, 6 Oct 2010 08:12:00 -0500
Subject: [Freeswitch-users] FW: Play_fsv application Issue
Message-ID: <06502C073AD9394AADB3CA7FD94931BC0519DF2E@okc1x1.Logixcom.com>

I have something strange happen when I try to play back a fsv file. It
will play super-fast. You can hear the audio go fast too then you get a
long busy. Any ideas what would cause the speed to increase on playback?

 

Thanks,

Josh

 

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From testa at voicetechnology.com.br  Wed Oct  6 06:23:39 2010
From: testa at voicetechnology.com.br (Fernando Gregianin Testa)
Date: Wed, 06 Oct 2010 10:23:39 -0300
Subject: [Freeswitch-users] SIP and FreeSWITCH presented at FATEC
Message-ID: <4CAC785B.5070101@voicetechnology.com.br>

Hi folks,

You may be pleased to know that FreeSWITCH was presented at IV Semana de
Tecnologia at FATEC Carapicu?ba (http://www.fateccarapicuiba.com.br/),
near S?o Paulo, Brazil. FATEC is a state funding faculty that offers a
sort of undergraduate courses in IT, logistics, games, etc. Around 150
students attended the lecture with clear interest.
Presentation files are available (in pt-BR) at
 http://dl.dropbox.com/u/410277/fatec/index.html

Fernando G. Testa

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From gmaruzz at gmail.com  Wed Oct  6 07:04:15 2010
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 6 Oct 2010 16:04:15 +0200
Subject: [Freeswitch-users] SIP and FreeSWITCH presented at FATEC
In-Reply-To: <4CAC785B.5070101@voicetechnology.com.br>
References: <4CAC785B.5070101@voicetechnology.com.br>
Message-ID: <AANLkTinCeEFMN=mXhtmJH6JSNcsT+ym8RSbW+4BtmV2t@mail.gmail.com>

Yay!

On Wed, Oct 6, 2010 at 3:23 PM, Fernando Gregianin Testa
<testa at voicetechnology.com.br> wrote:
> Hi folks,
>
> You may be pleased to know that FreeSWITCH was presented at IV Semana de
> Tecnologia at FATEC Carapicu?ba (http://www.fateccarapicuiba.com.br/),
> near S?o Paulo, Brazil. FATEC is a state funding faculty that offers a
> sort of undergraduate courses in IT, logistics, games, etc. Around 150
> students attended the lecture with clear interest.
> Presentation files are available (in pt-BR) at
> ?http://dl.dropbox.com/u/410277/fatec/index.html
>
> Fernando G. Testa
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From jeff at jefflenk.com  Wed Oct  6 07:57:46 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 6 Oct 2010 07:57:46 -0700 (PDT)
Subject: [Freeswitch-users] mod_managed on linux centos
In-Reply-To: <AANLkTinF2yvyA6ZerxjFoOi+ti8DAGHAyEYfBFrRGiR1@mail.gmail.com>
References: <AANLkTi=xt9CZBPZLnBX56e2Rjdc_FRf_=er6Yy7w7ZuN@mail.gmail.com>
	<AANLkTi=S=tpzczDM8WSm6N8AeTkN3XNTPda6048sRoJ9@mail.gmail.com>
	<4C801B6E.4050103@puzzled.xs4all.nl>
	<AANLkTimzjbLHbObgsQKqX3t-hfaZWr95UGMi17YxZFHS@mail.gmail.com>
	<4C838C3B.3070202@puzzled.xs4all.nl>
	<AANLkTinP+UfvkM0dmBycy5GqivAXNHGF9RnTBNvV0gV4@mail.gmail.com>
	<AANLkTik1CY5-6DUvsLg512bGRr66xLF5H8Gi7WYRenHx@mail.gmail.com>
	<1286205151753-5599563.post@n2.nabble.com>
	<AANLkTinKemVBa5HNSrJk6EOfWYM5wd7CyL4L-JX-BawQ@mail.gmail.com>
	<AANLkTinF2yvyA6ZerxjFoOi+ti8DAGHAyEYfBFrRGiR1@mail.gmail.com>
Message-ID: <1286377066483-5607396.post@n2.nabble.com>


Not sure how this makes sense yet. Yitzchok would you try this with only the
mono_thread_detach calls removed leaving the mono_thread_attach in place and
see what happens.

Thanks
Jeff
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-managed-on-linux-centos-tp5485480p5607396.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Wed Oct  6 08:19:18 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 Oct 2010 08:19:18 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <AANLkTimrmU-05t9kNPdUMs932GCVnW+BL+MtzeAjH0MO@mail.gmail.com>

Hello all!

Our agenda for today is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_10_06

We have a few things to discuss, such as some additions to the default
configs and some Jira tasks that we could use some help with. Also, we are
hoping that Darren Schreiber will be calling in to give us an update from
ITExpo down in L.A.

Talk to you soon,
Michael
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From Joshua.Foshee at LogixCom.com  Wed Oct  6 08:28:37 2010
From: Joshua.Foshee at LogixCom.com (Joshua Foshee)
Date: Wed, 6 Oct 2010 10:28:37 -0500
Subject: [Freeswitch-users] Application Play_fsv issues
Message-ID: <06502C073AD9394AADB3CA7FD94931BC0519DF30@okc1x1.Logixcom.com>

I have a problem that if I play back a FSV file that it plays
super-fast. It records fine though when it plays back I see the video
and auto go by fast and then it plays a long busy tone. Any ideas?

Thanks in advance,

Josh


From adminjew at gmail.com  Wed Oct  6 08:40:06 2010
From: adminjew at gmail.com (Yitzchok)
Date: Wed, 6 Oct 2010 11:40:06 -0400
Subject: [Freeswitch-users] mod_managed on linux centos
In-Reply-To: <1286377066483-5607396.post@n2.nabble.com>
References: <AANLkTi=xt9CZBPZLnBX56e2Rjdc_FRf_=er6Yy7w7ZuN@mail.gmail.com>
	<AANLkTi=S=tpzczDM8WSm6N8AeTkN3XNTPda6048sRoJ9@mail.gmail.com>
	<4C801B6E.4050103@puzzled.xs4all.nl>
	<AANLkTimzjbLHbObgsQKqX3t-hfaZWr95UGMi17YxZFHS@mail.gmail.com>
	<4C838C3B.3070202@puzzled.xs4all.nl>
	<AANLkTinP+UfvkM0dmBycy5GqivAXNHGF9RnTBNvV0gV4@mail.gmail.com>
	<AANLkTik1CY5-6DUvsLg512bGRr66xLF5H8Gi7WYRenHx@mail.gmail.com>
	<1286205151753-5599563.post@n2.nabble.com>
	<AANLkTinKemVBa5HNSrJk6EOfWYM5wd7CyL4L-JX-BawQ@mail.gmail.com>
	<AANLkTinF2yvyA6ZerxjFoOi+ti8DAGHAyEYfBFrRGiR1@mail.gmail.com>
	<1286377066483-5607396.post@n2.nabble.com>
Message-ID: <AANLkTin0HvwpYfOCoH3upSGvA+UFiirjRoBic72yovqg@mail.gmail.com>

I tried that and it seems to work.


---diff from git head---


@@ -385,9 +385,7 @@ SWITCH_STANDARD_API(managedrun_api_function)
  } else {
  stream->write_function(stream, "-ERR ExecuteBackground returned false
(unknown module or exception?).\n");
  }
-#ifndef _MANAGED
- mono_thread_detach(mono_thread_current());
-#endif
+
  return SWITCH_STATUS_SUCCESS;
 }

@@ -403,9 +401,7 @@ SWITCH_STANDARD_API(managed_api_function)
  if (!(executeDelegate(cmd, stream, stream->param_event))) {
  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Execute failed
for %s (unknown module or exception).\n", cmd);
  }
-#ifndef _MANAGED
- mono_thread_detach(mono_thread_current());
-#endif
+
  return SWITCH_STATUS_SUCCESS;
 }

@@ -421,9 +417,7 @@ SWITCH_STANDARD_APP(managed_app_function)
  if (!(runDelegate(data, session))) {
  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Application run
failed for %s (unknown module or exception).\n", data);
  }
-#ifndef _MANAGED
- mono_thread_detach(mono_thread_current());
-#endif
+
 }

 SWITCH_STANDARD_API(managedreload_api_function)
@@ -438,9 +432,7 @@ SWITCH_STANDARD_API(managedreload_api_function)
  if (!(reloadDelegate(cmd))) {
  switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Execute failed
for %s (unknown module or exception).\n", cmd);
  }
-#ifndef _MANAGED
- mono_thread_detach(mono_thread_current());
-#endif
+
  return SWITCH_STATUS_SUCCESS;
 }



-----

Yitzchok


On Wed, Oct 6, 2010 at 10:57 AM, Jeff Lenk <jeff at jefflenk.com> wrote:

>
> Not sure how this makes sense yet. Yitzchok would you try this with only
> the
> mono_thread_detach calls removed leaving the mono_thread_attach in place
> and
> see what happens.
>
> Thanks
> Jeff
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/mod-managed-on-linux-centos-tp5485480p5607396.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Wed Oct  6 09:06:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Oct 2010 11:06:56 -0500
Subject: [Freeswitch-users] ClueCon video
Message-ID: <AANLkTi=1KS+h7FAB8TrYA6BMtpOuEw8eBxP2U4e1YVYC@mail.gmail.com>

http://www.freeswitch.org/node/287

-- 
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


From mthakershi at gmail.com  Wed Oct  6 09:17:09 2010
From: mthakershi at gmail.com (Malay Thakershi)
Date: Wed, 6 Oct 2010 11:17:09 -0500
Subject: [Freeswitch-users] Config issue after router/Firewall change
Message-ID: <AANLkTi=8p5S40+JV3cAitrudYdXmbP3jcM5zXG2BeNEK@mail.gmail.com>

We changed firewall our FS server is behind.

All off sudden, no voice heard on answered calls.

We made sure all rules are ditto as older router/firewall. We even
tried opening up everything but result is same. Called Vitelity: they
say according to them calls are OK.

Please help with you suggestions.


From mike at van.lammeren.net  Wed Oct  6 09:18:16 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Wed, 6 Oct 2010 12:18:16 -0400
Subject: [Freeswitch-users] Question about Lua freeswitch.bridge
Message-ID: <AANLkTimc9ChvXn=_Sz5ZSkJ21hng=YWbmAhs_f9VwDsZ@mail.gmail.com>

Hello!

I'm a huge FreeSWITCH fan and am always amazed at how much I can do with it.
I have a Lua script that has been working great, but now have been tasked
with enhancing it, and have run into a problem.

I want to be able to play media, detect DTMF and hang up a call after
bridging two sessions, but my Lua script blocks on freeswitch.bridge(), and
doesn't resume until one side or the other hangs up.

Here is my problem reduced to a Lua script in its simplest form:

-- first session
session1 = session
-- call out
session2 = freeswitch.Session("{ignore_early_media=true}sofia/gateway/
etc.com/" .. phoneNumber)
-- bridge the calls
freeswitch.bridge(session1, session2)
-- this next line doesn't execute until after either session 1 or 2 hangs up
freeswitch.consoleLog("info", "This is a test.\n")


How can I do an asynchronous bridge, and still be able to play media and
detect dtmf afterwards?

Mike van Lammeren
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From gmaruzz at gmail.com  Wed Oct  6 09:27:34 2010
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 6 Oct 2010 18:27:34 +0200
Subject: [Freeswitch-users] ClueCon video
In-Reply-To: <AANLkTi=1KS+h7FAB8TrYA6BMtpOuEw8eBxP2U4e1YVYC@mail.gmail.com>
References: <AANLkTi=1KS+h7FAB8TrYA6BMtpOuEw8eBxP2U4e1YVYC@mail.gmail.com>
Message-ID: <AANLkTimJ2cv_vE7SvF999ZYB_4AjCEQ_B60RZ0xYnZQ4@mail.gmail.com>

Nice tie!

:)
-giovanni

On Wed, Oct 6, 2010 at 6:06 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> http://www.freeswitch.org/node/287
>
> --
> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From steveayre at gmail.com  Wed Oct  6 09:30:15 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 6 Oct 2010 17:30:15 +0100
Subject: [Freeswitch-users] Config issue after router/Firewall change
In-Reply-To: <AANLkTi=8p5S40+JV3cAitrudYdXmbP3jcM5zXG2BeNEK@mail.gmail.com>
References: <AANLkTi=8p5S40+JV3cAitrudYdXmbP3jcM5zXG2BeNEK@mail.gmail.com>
Message-ID: <AANLkTimW57oqURt1jif8ujMMYFW9x6iQJt6O55O-W=0Z@mail.gmail.com>

Well, something must be different...
- Has FS IP changed at all?
- Any NAT?
- Any SIP ALG?
- Have you verified the media IPs in the SDP are still correct?

-Steve


On 6 October 2010 17:17, Malay Thakershi <mthakershi at gmail.com> wrote:
> We changed firewall our FS server is behind.
>
> All off sudden, no voice heard on answered calls.
>
> We made sure all rules are ditto as older router/firewall. We even
> tried opening up everything but result is same. Called Vitelity: they
> say according to them calls are OK.
>
> Please help with you suggestions.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From ale975 at gmail.com  Wed Oct  6 09:43:09 2010
From: ale975 at gmail.com (Ale)
Date: Wed, 6 Oct 2010 18:43:09 +0200
Subject: [Freeswitch-users] Fax hint request
In-Reply-To: <AANLkTim8NhxTcypcA9kuT4vnmRNObZ2JbZzgVNdGpsgW@mail.gmail.com>
References: <AANLkTim8NhxTcypcA9kuT4vnmRNObZ2JbZzgVNdGpsgW@mail.gmail.com>
Message-ID: <AANLkTimmjxQjfzyb=+ZxvJzFDR5EduyM8pLXBXbX+62p@mail.gmail.com>

On Wed, Oct 6, 2010 at 11:10 AM, Ale <ale975 at gmail.com> wrote:
>
> Can anyone give me a hint
>

ok i suppose my question could be a bit dumb, can anyone explain what
i miss o point me to some documentation.

Many thx,
Ale


From rupa at rupa.com  Wed Oct  6 09:47:28 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 6 Oct 2010 11:47:28 -0500
Subject: [Freeswitch-users] compare bill rate of different increment
In-Reply-To: <AANLkTi=vb8mne+rQiNzGKTpRRq5LCXSVXYX-1FP0wXk6@mail.gmail.com>
References: <AANLkTi=vb8mne+rQiNzGKTpRRq5LCXSVXYX-1FP0wXk6@mail.gmail.com>
Message-ID: <AANLkTi=yBFgnYmE9P=BNysTdfwxXQC5tAxqD+C9ddYhB@mail.gmail.com>

You have to normalize your data.  Just like you normalize your prefixes to
e164 format, you should normalize your rates to a common increment.  You
*could* load both the real rate and real increment and do math in your sql
using custom_sql but it would be more efficient and I think easier to
normalize on load.

On Tue, Oct 5, 2010 at 11:20 PM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hi,
>
> I would like to use mod_lcr but I am wondering how I should due with
> rate decks from different providers that have different bill
> increment?
>
> If one deck's increment is 6 s and another one is 1 s, is it fair to
> compare the two deck just purely based on the rate or is there a
> better way of handling that?
>
> Thank you for your input.
>
> Thanks,
>
> Woody
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From rupa at rupa.com  Wed Oct  6 09:49:15 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 6 Oct 2010 11:49:15 -0500
Subject: [Freeswitch-users] FreeSWITCH CLI description?
In-Reply-To: <02db01cb651d$576a1ca0$063e55e0$@khirman.com>
References: <02db01cb651d$576a1ca0$063e55e0$@khirman.com>
Message-ID: <AANLkTim_5yqX78K9T-u-KNu5XzfZYAndaK79e80D=S_s@mail.gmail.com>

Try "help" from the cli.  And of course the wiki link given by Nagalenoj.
 Remember that modules can add commands, so there are generally more
commands available than what is provided on that single wiki page.  "help"
is equivalent to "show api" which would include modules other than just
mod_commands.

On Wed, Oct 6, 2010 at 1:11 AM, Stas Khirman <stas at khirman.com> wrote:

> Hi,
>
>
>
> Sorry for probably dumb  questions, but I can?t find answer in the
> docs/wiki:
>
>
>
> I?m looking for a good list of major CLI commands and their descriptions ?
> any pointers are deeply appreciated.
>
>
>
> Regards
>
> Stas
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From mario_fs at mgtech.com  Wed Oct  6 09:55:14 2010
From: mario_fs at mgtech.com (Mario)
Date: Wed, 06 Oct 2010 09:55:14 -0700
Subject: [Freeswitch-users] Inbound caller ID lost after return from bridge
Message-ID: <4CACA9F2.7070801@mgtech.com>

THe extension below works fine except that the inbound caller ID is lost
between the first and second bridge, it is replaced by 0000000000 on the
local extensions. Does the first bridge function cause the inbound
caller ID to be lost? Is there a way to keep it so it shows up during
the second bridge? Thanks. Mario


    <extension name="u1phone">
      <condition field="destination_number" expression="^2920$">
        <!--  Make sure they hear rings after playing hi  -->
	<action application="set" data="ringback=${us-ring}"/>
         <!--action application="playback" data="phrase:mvpeople:100"/-->
        <action application="execute_extension" data="2921 XML default"/>
        <!-- Next lines allow control after bridge timeout and hangup
after bridge -->
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
        <!-- ring home extensions only for ## seconds -->
        <action application="set" data="call_timeout=15"/>
        <action application="bridge" data="group/MXT@${domain_name}"/>
        <!-- Say we are keep trying.... -->
        <action application="execute_extension" data="2922 XML default"/>
        <action application="playback"
data="/usr/local/freeswitch/conf/mvsounds/we_trying_please_hold.wav"/>
        <!-- Setup so they hear music while ringing a second time -->
	<action application="set" data="ringback=$${hold_music}"/>
        <!-- this is needed to allow call_timeout to work after bridging
to a gateway -->
        <action application="set" data="ignore_early_media=true"/>
        <!-- ring home extensions +cell phone for ## seconds  NOTE THE
NEXT 2 LINES DID NOT WORK TO SET TIMEOUT -->
        <!--action application="set" data="bridge_answer_timeout=10"/-->
        <!--action application="set" data="call_timeout=15"/-->
to ring simoultaneously -->
        <action application="bridge"
data="<originate_timeout=15>group/MXT@${domain_name}:_:sofia/gateway/uid1/19161234567:_:sofia/gateway/uid1/19162223333"/>
        <!-- No answer go to voicemail -->
        <action application="answer"/>
        <action application="voicemail" data="default ${domain_name} 100"/>
      </condition>
   </extension>


From peder at networkoblivion.com  Wed Oct  6 10:11:41 2010
From: peder at networkoblivion.com (Peder)
Date: Wed, 6 Oct 2010 12:11:41 -0500
Subject: [Freeswitch-users] Config issue after router/Firewall change
In-Reply-To: <AANLkTi=8p5S40+JV3cAitrudYdXmbP3jcM5zXG2BeNEK@mail.gmail.com>
References: <AANLkTi=8p5S40+JV3cAitrudYdXmbP3jcM5zXG2BeNEK@mail.gmail.com>
Message-ID: <0ea901cb6579$8b77bf00$a2673d00$@com>

What kind of firewall is it now and was it before?  Turn off SIP ALG if
there is one (unless it is a PIX/ASA or Fortinet as those are the only two
that seem to handle it ok).

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Malay
Thakershi
Sent: Wednesday, October 06, 2010 11:17 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Config issue after router/Firewall change

We changed firewall our FS server is behind.

All off sudden, no voice heard on answered calls.

We made sure all rules are ditto as older router/firewall. We even
tried opening up everything but result is same. Called Vitelity: they
say according to them calls are OK.

Please help with you suggestions.

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



From peder at networkoblivion.com  Wed Oct  6 10:13:59 2010
From: peder at networkoblivion.com (Peder)
Date: Wed, 6 Oct 2010 12:13:59 -0500
Subject: [Freeswitch-users] Fax hint request
In-Reply-To: <AANLkTimmjxQjfzyb=+ZxvJzFDR5EduyM8pLXBXbX+62p@mail.gmail.com>
References: <AANLkTim8NhxTcypcA9kuT4vnmRNObZ2JbZzgVNdGpsgW@mail.gmail.com>
	<AANLkTimmjxQjfzyb=+ZxvJzFDR5EduyM8pLXBXbX+62p@mail.gmail.com>
Message-ID: <0eb201cb6579$de02ba40$9a082ec0$@com>

What are you trying to do?  And what is the problem?  This works for me to
receive a fax:


<extension name="fax_receive">
     <condition field="destination_number" expression="^(1010)$">
        <action application="answer"/>
        <action application="playback" data="silence_stream://2000"/>
        <action application="rxfax" data="/tmp/${uuid}.rxfax.tiff"/>
        <action application="hangup"/>
     </condition>
   </extension>

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ale
Sent: Wednesday, October 06, 2010 11:43 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Fax hint request

On Wed, Oct 6, 2010 at 11:10 AM, Ale <ale975 at gmail.com> wrote:
>
> Can anyone give me a hint
>

ok i suppose my question could be a bit dumb, can anyone explain what
i miss o point me to some documentation.

Many thx,
Ale

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



From d at d-man.org  Wed Oct  6 10:32:20 2010
From: d at d-man.org (Darren Schreiber)
Date: Wed, 6 Oct 2010 10:32:20 -0700
Subject: [Freeswitch-users] ONE WEEK LEFT - Official FreeSWITCH Training in
	New York City
Message-ID: <790757FF-DF56-421F-99C6-BF47EE29B083@d-man.org>

Hi folks,
The Official FreeSWITCH Training course in New York is just one week away and is almost full. If you are interested in learning a ton of FreeSWITCH, taught by one of the authors of the FreeSWITCH Book, you should consider signing up.

Code FREEVOIP is still valid for $400 off the training. You can register at http://www.voipkb.com/ . There is also a huge discount for registering two or more people together.

The course is an in-depth dive into FreeSWITCH with completely different material then is in the book. We will cover:

 *   Understanding configuration files and the default configuration
 *   Call authentication and routing basics
 *   Integration modules (mod_skypiax, mod_dingaling for Skype/GTalk/XMPP integration)
 *   Understanding presence
 *   Load balancing and high availability
 *   FreeSWITCH Internals
 *   How to debug and troubleshoot FreeSWITCH
 *   Building Custom C Modules
 *   Advanced Modules

Please let me know if you have any questions. We're excited to continue offering this course and thank the FreeSWITCH team for their continued outstanding work. See you in New York!

Thanks,

Darren Schreiber

Join us in October for FreeSWITCH Training!
Visit www.voipkb.com<http://www.voipkb.com/> for more information
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From mthakershi at gmail.com  Wed Oct  6 10:35:11 2010
From: mthakershi at gmail.com (Malay Thakershi)
Date: Wed, 6 Oct 2010 12:35:11 -0500
Subject: [Freeswitch-users] Config issue after router/Firewall change
In-Reply-To: <0ea901cb6579$8b77bf00$a2673d00$@com>
References: <AANLkTi=8p5S40+JV3cAitrudYdXmbP3jcM5zXG2BeNEK@mail.gmail.com>
	<0ea901cb6579$8b77bf00$a2673d00$@com>
Message-ID: <AANLkTi=VTDYarQ6Jf+46b44cUc_vXOn-3EfNr3Rz8Jjs@mail.gmail.com>

Router is pfsense

On Wed, Oct 6, 2010 at 12:11 PM, Peder <peder at networkoblivion.com> wrote:
> What kind of firewall is it now and was it before? ?Turn off SIP ALG if
> there is one (unless it is a PIX/ASA or Fortinet as those are the only two
> that seem to handle it ok).
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Malay
> Thakershi
> Sent: Wednesday, October 06, 2010 11:17 AM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] Config issue after router/Firewall change
>
> We changed firewall our FS server is behind.
>
> All off sudden, no voice heard on answered calls.
>
> We made sure all rules are ditto as older router/firewall. We even
> tried opening up everything but result is same. Called Vitelity: they
> say according to them calls are OK.
>
> Please help with you suggestions.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mario_fs at mgtech.com  Wed Oct  6 10:57:42 2010
From: mario_fs at mgtech.com (Mario)
Date: Wed, 06 Oct 2010 10:57:42 -0700
Subject: [Freeswitch-users] What are <> in bridge as opposed to {} []
Message-ID: <4CACB896.5040207@mgtech.com>

The wiki explains what {} and [] are used for in the line below, but I
was provided <> from this list that fixed a problem and I can't find how
<> is different from the others. Anyone?

<action application="bridge"
data="<originate_timeout=15>group/MGT@${domain_name}:_:sofia/gate...


From fs-list at communicatefreely.net  Wed Oct  6 11:15:16 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Wed, 06 Oct 2010 14:15:16 -0400
Subject: [Freeswitch-users] What are <> in bridge as opposed to {} []
In-Reply-To: <4CACB896.5040207@mgtech.com>
References: <4CACB896.5040207@mgtech.com>
Message-ID: <4CACBCB4.5060804@communicatefreely.net>

The syntax you have below is the "enterprise bridge" method, using :_: instead of ,.  It spawns a
separate thread for each leg.

In this case, the <> are global options, {} are options for each leg.



Mario wrote:
> The wiki explains what {} and [] are used for in the line below, but I
> was provided <> from this list that fixed a problem and I can't find how
> <> is different from the others. Anyone?
> 
> <action application="bridge"
> data="<originate_timeout=15>group/MGT@${domain_name}:_:sofia/gate...
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mario_fs at mgtech.com  Wed Oct  6 11:24:30 2010
From: mario_fs at mgtech.com (Mario)
Date: Wed, 06 Oct 2010 11:24:30 -0700
Subject: [Freeswitch-users] Question about effective caller ID & origination
	caller ID
Message-ID: <4CACBEDE.8060805@mgtech.com>

I have the inbound caller ID working for a bridge but can't reconcile
why effective_caller_id.. does not work. I spent a lot of time the last
few days reading the wiki and other web info. Since I can't find the
answer anywhere can anyone shed some light on this?

The wiki states:
effective_caller_id_number
Sets the effective callerid number. This is automatically exported to
the B-leg; however, it is not valid in an origination string. In other
words, set this before calling bridge, otherwise use
origination_caller_id_number

Does NOT work before the bridge:
        <!--action application="set"
data="effective_caller_id_number=${caller_id_number}"/>
        <action application="set"
data="effective_caller_id_name=${caller_id_name}"/-->

Does NOT work before the bridge:
        <!--action application="set"
data="origination_caller_id_number=${caller_id_number}"/>
        <action application="set"
data="origination_caller_id_name=${caller_id_name}"/-->
        <!--action application="set"

ONLY way IT WORKS (must be added before dialstring):
        <action application="bridge"
data="<originate_timeout=15,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/MGT@${domain_name}:_:sofia/gateway/uuid1/19162223333:_:sofia/gateway/uuid1/19164445555"/>


The caller ID is lost in an extension, although I still don't know why
at least I got it back.



From mario_fs at mgtech.com  Wed Oct  6 11:26:14 2010
From: mario_fs at mgtech.com (Mario)
Date: Wed, 06 Oct 2010 11:26:14 -0700
Subject: [Freeswitch-users] What are <> in bridge as opposed to {} []
In-Reply-To: <4CACBCB4.5060804@communicatefreely.net>
References: <4CACB896.5040207@mgtech.com>
	<4CACBCB4.5060804@communicatefreely.net>
Message-ID: <4CACBF46.3050702@mgtech.com>

Ahhhh. thank you! I need to use :_: because there are multiple
registrations to the same user.

On 10/06/10 11:15, Tim St. Pierre wrote:
> The syntax you have below is the "enterprise bridge" method, using :_: instead of ,.  It spawns a
> separate thread for each leg.
> 
> In this case, the <> are global options, {} are options for each leg.
> 
> 
> 
> Mario wrote:
>> The wiki explains what {} and [] are used for in the line below, but I
>> was provided <> from this list that fixed a problem and I can't find how
>> <> is different from the others. Anyone?
>>
>> <action application="bridge"
>> data="<originate_timeout=15>group/MGT@${domain_name}:_:sofia/gate...
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From mario_fs at mgtech.com  Wed Oct  6 11:34:14 2010
From: mario_fs at mgtech.com (Mario)
Date: Wed, 06 Oct 2010 11:34:14 -0700
Subject: [Freeswitch-users] What are <> in bridge as opposed to {} []
In-Reply-To: <4CACBF46.3050702@mgtech.com>
References: <4CACB896.5040207@mgtech.com>	<4CACBCB4.5060804@communicatefreely.net>
	<4CACBF46.3050702@mgtech.com>
Message-ID: <4CACC126.8010006@mgtech.com>

FYI, I looked up enterprise in the wiki but it makes no mention of <>
which is why it's impossible to find: Thanks again!

  Enterprise originate

The so called 'enterprise originate' syntax is a way to simultaneously
originate multiple calls in several threads (regular simultaneous
originate is single threaded). To use it simply separate valid dial
strings by :_: . Each dial string can have its own {} and [] blocks and
can themselves be simultaneous or sequential (containing ',' or '|') as
above.

This can be helpful, for example, when dealing with call forwarding.
Without it, FreeSWITCH? would drop the simultaneous dial and transfer to
the forwarded extension.


On 10/06/10 11:26, Mario wrote:
> Ahhhh. thank you! I need to use :_: because there are multiple
> registrations to the same user.
> 
> On 10/06/10 11:15, Tim St. Pierre wrote:
>> The syntax you have below is the "enterprise bridge" method, using :_: instead of ,.  It spawns a
>> separate thread for each leg.
>>
>> In this case, the <> are global options, {} are options for each leg.
>>
>>
>>
>> Mario wrote:
>>> The wiki explains what {} and [] are used for in the line below, but I
>>> was provided <> from this list that fixed a problem and I can't find how
>>> <> is different from the others. Anyone?
>>>
>>> <action application="bridge"
>>> data="<originate_timeout=15>group/MGT@${domain_name}:_:sofia/gate...
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From jonyoung111 at gmail.com  Wed Oct  6 11:16:21 2010
From: jonyoung111 at gmail.com (Jon Young)
Date: Wed, 6 Oct 2010 11:16:21 -0700
Subject: [Freeswitch-users] Sending "alert-info:...." to devices from
	Conference
In-Reply-To: <4CA4AB60.8050505@communicatefreely.net>
References: <AANLkTi=V4HSLG2TcjaXWQtM1TdSFvGJ2dVoeX7yR=ze_@mail.gmail.com>
	<C4463379-4708-4585-A820-D527896B2809@carmickle.com>
	<4CA4AB60.8050505@communicatefreely.net>
Message-ID: <AANLkTinkj7FOC95-KSR==ggu-vE2FRWRCBBGCs_GDMd5@mail.gmail.com>

I have tried using the suggested without luck.

<action application="export" data="alert_info=${domain};info=alert-internal"/>

However, I entered it in the page/intercom dialplan app that I
previously mentioned.  Below is an section of it:

- <extension name="paging02">
- <condition field="destination_number" expression="^2999$">
  <action application="set"
data="conference_auto_outcall_caller_id_name=Page2" />
  <action application="set"
data="conference_auto_outcall_caller_id_number=2999" />
  <action application="set" data="conference_auto_outcall_timeout=60" />
  <action application="set" data="conference_auto_outcall_flags=mute" />
  <action application="set"
data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app
2 a s1 transfer::intercept:${uuid} inline'}" />
  <action application="export"
data="alert_info=${domain};info=alert-autoanswer"/>
  <action application="set" data="sip_exclude_contact=${network_addr}" />
  <action application="conference_set_auto_outcall"
data="${group_call(pagezone02)}" />
  <action application="conference"
data="paging2 at default+flags{endconf|deaf}" />
  </condition>
  </extension>

This results in no change, but is this the correct place or format?
The existing app uses "conference_auto_outcall_prefix=..." that works
fine for the Aastra phones, but the "Alert-info..." that my LG phones
need is not included and I think I need to submit something different
into the "conference_auto_outcall" API or does it need to go somewhere
else.

Thanks,

Jon

On Thu, Sep 30, 2010 at 8:23 AM, Tim St. Pierre
<fs-list at communicatefreely.net> wrote:
> So far, 9133i, 9143i, 6731i, 9753i, 9157i, 9139i
>
> I'm assuming all of them since they use the same firmware.
>
> You may need to put this little snippet in your aastra config file though.
>
>
> sip allow auto-answer: 1
> sip intercom allow barge in: 0
>
> priority alerting enabled: 1
> alert group: 4
> alert external: 0
> alert internal: 1
> alert emergency: 2
> alert priority: 3
>
>
> Frank Carmickle wrote:
>> What phones?
>>
>> --FC
>>
>> On Sep 29, 2010, at 3:22 PM, Jon Young wrote:
>>
>>> Hi,
>>>
>>> I'm using a derivative of the "mad_boss_intercom" to set up paging. ?I
>>> have some devices that need an "alert-info:<...url...>;
>>> info=alert-autoanswer" in order to auto answer. ?Can I set or export
>>> that from within the paging/intercom XML or can it be included in the
>>> "conference_auto_outcall_prefix". ?Here is what one of the zones looks
>>> like:
>>>
>>> <extension name="paging01">
>>> ?<condition field="destination_number" expression="^1999$">
>>> ? ?<action application="set"
>>> data="conference_auto_outcall_caller_id_name=Page1"/>
>>> ? ?<action application="set"
>>> data="conference_auto_outcall_caller_id_number=1999"/>
>>> ? ?<action application="set" data="conference_auto_outcall_timeout=60"/>
>>> ? ?<action application="set" data="conference_auto_outcall_flags=mute"/>
>>> ? ?<action application="set"
>>> data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app
>>> 2 a s1 transfer::intercept:${uuid} inline'}"/>
>>> ? ?<action application="set" data="sip_exclude_contact=${network_addr}"/>
>>> ? ?<action application="conference_set_auto_outcall"
>>> data="${group_call(pagezone01)}"/>
>>> ? ?<action application="conference"
>>> data="paging1 at default+flags{endconf|deaf}"/>
>>> ?</condition>
>>> </extension>
>>>
>>> This currently works fine with Aastra phones, but not my other phones
>>> which need the "alert-info..."
>>>
>>> Thanks,
>>>
>>> Jon
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From edpimentl at gmail.com  Wed Oct  6 12:21:29 2010
From: edpimentl at gmail.com (EdPimentl)
Date: Wed, 6 Oct 2010 15:21:29 -0400
Subject: [Freeswitch-users] ONE WEEK LEFT - Official FreeSWITCH Training
 in New York City
In-Reply-To: <790757FF-DF56-421F-99C6-BF47EE29B083@d-man.org>
References: <790757FF-DF56-421F-99C6-BF47EE29B083@d-man.org>
Message-ID: <AANLkTincKR7t7MnV+36HFqwcn2HxgeendxixDaG-F3kO@mail.gmail.com>

For those who will not be there, suggest a HD VIDEO/Mpeg4 purchasing option.

-E
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From tculjaga at gmail.com  Wed Oct  6 12:34:12 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 6 Oct 2010 21:34:12 +0200
Subject: [Freeswitch-users] Radius AAA
In-Reply-To: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>
References: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>
Message-ID: <AANLkTi=Nt+3CcE=QEnggv9gJqdSw8jBX1gSh=GbhSgMi@mail.gmail.com>

On Wed, Oct 6, 2010 at 7:44 AM, Abid Saleem <abid_freeswitch at live.com>wrote:

>  Hi All,
>
> Can someone provide a clear guideline on how to setup Radius AAA to work
> with FS. Please send complete installation & configuration instructions.
> Thanks for help in advance.
>
>

well, yahh... this is still pending on my task list ...

sorry guys... the module works quite nice but didn't document it.
promise to do it this week-end.

T.
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From fs-list at communicatefreely.net  Wed Oct  6 12:43:52 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Wed, 06 Oct 2010 15:43:52 -0400
Subject: [Freeswitch-users] What are <> in bridge as opposed to {} []
In-Reply-To: <4CACC126.8010006@mgtech.com>
References: <4CACB896.5040207@mgtech.com>	<4CACBCB4.5060804@communicatefreely.net>	<4CACBF46.3050702@mgtech.com>
	<4CACC126.8010006@mgtech.com>
Message-ID: <4CACD178.9020609@communicatefreely.net>

Yep, I just went through the same thing!

One thing I found about the enterprise bridge syntax, is that not all variables get passed as you
would expect with the normal bridge syntax.

You also don't get an originate_disposition variable set

-Tim


Mario wrote:
> FYI, I looked up enterprise in the wiki but it makes no mention of <>
> which is why it's impossible to find: Thanks again!
> 
>   Enterprise originate
> 
> The so called 'enterprise originate' syntax is a way to simultaneously
> originate multiple calls in several threads (regular simultaneous
> originate is single threaded). To use it simply separate valid dial
> strings by :_: . Each dial string can have its own {} and [] blocks and
> can themselves be simultaneous or sequential (containing ',' or '|') as
> above.
> 
> This can be helpful, for example, when dealing with call forwarding.
> Without it, FreeSWITCH? would drop the simultaneous dial and transfer to
> the forwarded extension.
> 
> 
> On 10/06/10 11:26, Mario wrote:
>> Ahhhh. thank you! I need to use :_: because there are multiple
>> registrations to the same user.
>>
>> On 10/06/10 11:15, Tim St. Pierre wrote:
>>> The syntax you have below is the "enterprise bridge" method, using :_: instead of ,.  It spawns a
>>> separate thread for each leg.
>>>
>>> In this case, the <> are global options, {} are options for each leg.
>>>
>>>
>>>
>>> Mario wrote:
>>>> The wiki explains what {} and [] are used for in the line below, but I
>>>> was provided <> from this list that fixed a problem and I can't find how
>>>> <> is different from the others. Anyone?
>>>>
>>>> <action application="bridge"
>>>> data="<originate_timeout=15>group/MGT@${domain_name}:_:sofia/gate...
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From stas at khirman.com  Wed Oct  6 12:55:06 2010
From: stas at khirman.com (Stas Khirman)
Date: Wed, 6 Oct 2010 12:55:06 -0700
Subject: [Freeswitch-users] ClueCon video
In-Reply-To: <AANLkTi=1KS+h7FAB8TrYA6BMtpOuEw8eBxP2U4e1YVYC@mail.gmail.com>
References: <AANLkTi=1KS+h7FAB8TrYA6BMtpOuEw8eBxP2U4e1YVYC@mail.gmail.com>
Message-ID: <049301cb6590$60f02d00$22d08700$@khirman.com>

Anthony,

Did you published ClueCon presentations? Will appreciate a pointer.

Regards
Stas

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-
> bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
> Sent: Wednesday, October 06, 2010 9:07 AM
> To: Freeswitch-users
> Subject: [Freeswitch-users] ClueCon video
> 
> http://www.freeswitch.org/node/287
> 
> --
> 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
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From stephen at stephenjc.com  Wed Oct  6 13:15:40 2010
From: stephen at stephenjc.com (stephen at stephenjc)
Date: Wed, 6 Oct 2010 16:15:40 -0400
Subject: [Freeswitch-users] ClueCon video
In-Reply-To: <049301cb6590$60f02d00$22d08700$@khirman.com>
References: <AANLkTi=1KS+h7FAB8TrYA6BMtpOuEw8eBxP2U4e1YVYC@mail.gmail.com>
	<049301cb6590$60f02d00$22d08700$@khirman.com>
Message-ID: <AANLkTinSUL8m+x9r+tu_NjRhjF1m-RrhXFfkaUzeWmgC@mail.gmail.com>

haha.

Thanks,
Stephen C
-All of my email addresses go to the same place
-Save Paper, think before you print



On Wed, Oct 6, 2010 at 3:55 PM, Stas Khirman <stas at khirman.com> wrote:

> Anthony,
>
> Did you published ClueCon presentations? Will appreciate a pointer.
>
> Regards
> Stas
>
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-
> > bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
> > Sent: Wednesday, October 06, 2010 9:07 AM
> > To: Freeswitch-users
> > Subject: [Freeswitch-users] ClueCon video
> >
> > http://www.freeswitch.org/node/287
> >
> > --
> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Peter.Hinman at ParcelPool.com  Wed Oct  6 14:19:01 2010
From: Peter.Hinman at ParcelPool.com (Peter Hinman)
Date: Wed, 06 Oct 2010 15:19:01 -0600
Subject: [Freeswitch-users] ODBC and MSSQL
In-Reply-To: <1286331931285-5605569.post@n2.nabble.com>
References: <4CAB9F5A.2090503@ParcelPool.com>
	<1286331931285-5605569.post@n2.nabble.com>
Message-ID: <4CACE7C5.6060208@ParcelPool.com>

  Thanks Jeff.

I guess I'll stop trying to tweak the config files.

Peter


On 10/5/2010 8:25 PM, Jeff Lenk wrote:
> This is a known issue, and is waiting for submittal.




From msc at freeswitch.org  Wed Oct  6 14:57:55 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 Oct 2010 14:57:55 -0700
Subject: [Freeswitch-users] ClueCon video
In-Reply-To: <049301cb6590$60f02d00$22d08700$@khirman.com>
References: <AANLkTi=1KS+h7FAB8TrYA6BMtpOuEw8eBxP2U4e1YVYC@mail.gmail.com>
	<049301cb6590$60f02d00$22d08700$@khirman.com>
Message-ID: <AANLkTimD4CQd4pgDy5A=+TGXdA4Dw_h33S335eTXsvaQ@mail.gmail.com>

The ClueCon MMX videos have not been published yet. Our video editor has
been away for some time. He's due back next week. We will keep everyone
posted.
-MC

On Wed, Oct 6, 2010 at 12:55 PM, Stas Khirman <stas at khirman.com> wrote:

> Anthony,
>
> Did you published ClueCon presentations? Will appreciate a pointer.
>
> Regards
> Stas
>
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-
> > bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
> > Sent: Wednesday, October 06, 2010 9:07 AM
> > To: Freeswitch-users
> > Subject: [Freeswitch-users] ClueCon video
> >
> > http://www.freeswitch.org/node/287
> >
> > --
> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From oseslija at gmail.com  Wed Oct  6 15:18:36 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Thu, 7 Oct 2010 00:18:36 +0200
Subject: [Freeswitch-users] ClueCon video
In-Reply-To: <AANLkTi=1KS+h7FAB8TrYA6BMtpOuEw8eBxP2U4e1YVYC@mail.gmail.com>
References: <AANLkTi=1KS+h7FAB8TrYA6BMtpOuEw8eBxP2U4e1YVYC@mail.gmail.com>
Message-ID: <AANLkTi=EFO9qoKgATO0P6gwKeAB1WVC7CC4Op=vBd+gu@mail.gmail.com>

Who was the role model for the clown?

On Wed, Oct 6, 2010 at 6:06 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> http://www.freeswitch.org/node/287
>
> --
> 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Oct  6 16:15:37 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 Oct 2010 16:15:37 -0700
Subject: [Freeswitch-users] Radius AAA
In-Reply-To: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>
References: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>
Message-ID: <AANLkTi=GiQaO-48BP=T8c-A+09sJMKzQH7BX=ewBdB8R@mail.gmail.com>

I'm afraid no such comprehensive documentation exists. Submissions to our
wiki are most welcomed. This is all we have at the moment:

http://wiki.freeswitch.org/wiki/Mod_rad_auth

Be sure to install freeradius2 (server) and freeradius-client before trying
to install mod_auth_rad.

-MC

On Tue, Oct 5, 2010 at 10:44 PM, Abid Saleem <abid_freeswitch at live.com>wrote:

>  Hi All,
>
> Can someone provide a clear guideline on how to setup Radius AAA to work
> with FS. Please send complete installation & configuration instructions.
> Thanks for help in advance.
>
> Regards
> --------------
> Abid Saleem
> Sr. Product Manager
> Terminus Technologies
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Oct  6 16:16:41 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 Oct 2010 16:16:41 -0700
Subject: [Freeswitch-users] SIP and FreeSWITCH presented at FATEC
In-Reply-To: <4CAC785B.5070101@voicetechnology.com.br>
References: <4CAC785B.5070101@voicetechnology.com.br>
Message-ID: <AANLkTimRak4JZ1fdShqz9by3q13M88uGKFtMXWT1MVVQ@mail.gmail.com>

Glad to hear it! Thanks!
-MC

On Wed, Oct 6, 2010 at 6:23 AM, Fernando Gregianin Testa <
testa at voicetechnology.com.br> wrote:

> Hi folks,
>
> You may be pleased to know that FreeSWITCH was presented at IV Semana de
> Tecnologia at FATEC Carapicu?ba (http://www.fateccarapicuiba.com.br/),
> near S?o Paulo, Brazil. FATEC is a state funding faculty that offers a
> sort of undergraduate courses in IT, logistics, games, etc. Around 150
> students attended the lecture with clear interest.
> Presentation files are available (in pt-BR) at
>  http://dl.dropbox.com/u/410277/fatec/index.html
>
> Fernando G. Testa
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rodrigoferreiralang at gmail.com  Wed Oct  6 16:33:40 2010
From: rodrigoferreiralang at gmail.com (Rodrigo Lang)
Date: Thu, 7 Oct 2010 00:33:40 +0100
Subject: [Freeswitch-users] SIP and FreeSWITCH presented at FATEC
In-Reply-To: <AANLkTimRak4JZ1fdShqz9by3q13M88uGKFtMXWT1MVVQ@mail.gmail.com>
References: <4CAC785B.5070101@voicetechnology.com.br>
	<AANLkTimRak4JZ1fdShqz9by3q13M88uGKFtMXWT1MVVQ@mail.gmail.com>
Message-ID: <AANLkTinZ3wTd_AJ4n=vVY1ddczw-fYigaAMOTVo=5Dom@mail.gmail.com>

Many VoIP professionals in Brazil is interessed in the FreeSwitch, I'm one.


Regards,
Rodrigo Lang.


2010/10/7 Michael Collins <msc at freeswitch.org>

> Glad to hear it! Thanks!
> -MC
>
> On Wed, Oct 6, 2010 at 6:23 AM, Fernando Gregianin Testa <
> testa at voicetechnology.com.br> wrote:
>
>> Hi folks,
>>
>> You may be pleased to know that FreeSWITCH was presented at IV Semana de
>> Tecnologia at FATEC Carapicu?ba (http://www.fateccarapicuiba.com.br/),
>> near S?o Paulo, Brazil. FATEC is a state funding faculty that offers a
>> sort of undergraduate courses in IT, logistics, games, etc. Around 150
>> students attended the lecture with clear interest.
>> Presentation files are available (in pt-BR) at
>>  http://dl.dropbox.com/u/410277/fatec/index.html
>>
>> Fernando G. Testa
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Rodrigo Lang
http://rodrigorecipes.blogspot.com/<http://rodrigorecipes.blogspot.com/2010/08/ssh-rapido-e-pratico.html>
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From msc at freeswitch.org  Wed Oct  6 16:44:32 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 Oct 2010 16:44:32 -0700
Subject: [Freeswitch-users] Question about Lua freeswitch.bridge
In-Reply-To: <AANLkTimc9ChvXn=_Sz5ZSkJ21hng=YWbmAhs_f9VwDsZ@mail.gmail.com>
References: <AANLkTimc9ChvXn=_Sz5ZSkJ21hng=YWbmAhs_f9VwDsZ@mail.gmail.com>
Message-ID: <AANLkTi=fx8jjpyj6MZYea-NyLBfcq1er7-myv_MNWKw_@mail.gmail.com>

Just curious - would a bind_meta_app call work for you? I ask because you
can't do asynchronous operations like you described when calling scripts
from the dialplan. I'm afraid you will need to graduate to ESL. :) Check out
ESL and the event socket on the wiki and chapter 9 of the FreeSWITCH book.

Some things you will want to check out:
uuid_broadcast (to play media to a channel)
the DTMF event (http://wiki.freeswitch.org/wiki/Event_list#DTMF)
fs-ivrd (example of a daemon that sits there and connects outbound event
socket connects to an ESL script)

Using the event socket allows total and complete 3rd party control of your
sessions. The drawback is the the learning curve. It's a bit steep but it's
worth it.

-MC


On Wed, Oct 6, 2010 at 9:18 AM, Mike van Lammeren <mike at van.lammeren.net>wrote:

> Hello!
>
> I'm a huge FreeSWITCH fan and am always amazed at how much I can do with
> it. I have a Lua script that has been working great, but now have been
> tasked with enhancing it, and have run into a problem.
>
> I want to be able to play media, detect DTMF and hang up a call after
> bridging two sessions, but my Lua script blocks on freeswitch.bridge(), and
> doesn't resume until one side or the other hangs up.
>
> Here is my problem reduced to a Lua script in its simplest form:
>
> -- first session
> session1 = session
> -- call out
> session2 = freeswitch.Session("{ignore_early_media=true}sofia/gateway/
> etc.com/" .. phoneNumber)
> -- bridge the calls
> freeswitch.bridge(session1, session2)
> -- this next line doesn't execute until after either session 1 or 2 hangs
> up
> freeswitch.consoleLog("info", "This is a test.\n")
>
>
> How can I do an asynchronous bridge, and still be able to play media and
> detect dtmf afterwards?
>
> Mike van Lammeren
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Wed Oct  6 16:51:43 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 6 Oct 2010 20:51:43 -0300
Subject: [Freeswitch-users] SIP and FreeSWITCH presented at FATEC
In-Reply-To: <AANLkTinZ3wTd_AJ4n=vVY1ddczw-fYigaAMOTVo=5Dom@mail.gmail.com>
References: <4CAC785B.5070101@voicetechnology.com.br>
	<AANLkTimRak4JZ1fdShqz9by3q13M88uGKFtMXWT1MVVQ@mail.gmail.com>
	<AANLkTinZ3wTd_AJ4n=vVY1ddczw-fYigaAMOTVo=5Dom@mail.gmail.com>
Message-ID: <AANLkTinP=1P4rdENEY95fOWBZKwi=tk2tGT=txfK-o6o@mail.gmail.com>

Brazilian and Latin American FreeSWITCH? users should be tuned in the up
coming news from Khomp. They are getting ready to release their officially
supported mod_khomp on git anytime now.

Regards,
Jo?o Mesquita


On Wed, Oct 6, 2010 at 8:33 PM, Rodrigo Lang
<rodrigoferreiralang at gmail.com>wrote:

> Many VoIP professionals in Brazil is interessed in the FreeSwitch, I'm one.
>
>
> Regards,
> Rodrigo Lang.
>
>
> 2010/10/7 Michael Collins <msc at freeswitch.org>
>
> Glad to hear it! Thanks!
>> -MC
>>
>> On Wed, Oct 6, 2010 at 6:23 AM, Fernando Gregianin Testa <
>> testa at voicetechnology.com.br> wrote:
>>
>>> Hi folks,
>>>
>>> You may be pleased to know that FreeSWITCH was presented at IV Semana de
>>> Tecnologia at FATEC Carapicu?ba (http://www.fateccarapicuiba.com.br/),
>>> near S?o Paulo, Brazil. FATEC is a state funding faculty that offers a
>>> sort of undergraduate courses in IT, logistics, games, etc. Around 150
>>> students attended the lecture with clear interest.
>>> Presentation files are available (in pt-BR) at
>>>  http://dl.dropbox.com/u/410277/fatec/index.html
>>>
>>> Fernando G. Testa
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Rodrigo Lang
> http://rodrigorecipes.blogspot.com/<http://rodrigorecipes.blogspot.com/2010/08/ssh-rapido-e-pratico.html>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peter.schrock at gmail.com  Wed Oct  6 17:23:26 2010
From: peter.schrock at gmail.com (Peter Schrock)
Date: Wed, 6 Oct 2010 17:23:26 -0700
Subject: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4
Message-ID: <AANLkTingeYfPT_b6dG3bKQOVjxYsmBTKhDuLw_3BKGac@mail.gmail.com>

Okay, so I managed to get FS working the other day and I even managed to
test a call and test my voicemail. All seemed to be working smoothly until,
because of the rain here, my power went out and I had to reboot my computer.
I logged in through the terminal, set up FS in background went to fs_cli and
I get this error message:
[ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
Error]
I managed to figure out that in the file "fs_cli.c" on line 1206 you find
the code for displaying this error message. The problem is that I don't know
why this error message is occurring. Does anyone have any helpful hints as
to what I should look at to resolve this problem?
I even tried going to the git tree and make current, but that gave me
problems that forced me to turn off mod_spandsp and mod_skyopen in the
modules.conf, which I had running earlier. Any thoughts?

PeterS

PS
I am not sure if this is of any help, but in addition to the error line
above, it also posted this info:
Usage: fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x
command] [profile]

  -?,-h --help                    Usage Information
  -H, --host=hostname             Host to connect
  -P, --port=port                 Port to connect (1 - 65535)
  -u, --user=user at domain          user at domain
  -p, --password=password         Password
  -x, --execute=command           Execute Command and Exit
  -l, --loglevel=command          Log Level
  -q, --quiet                     Disable logging
  -r, --retry                     Retry connection on failure
  -R, --reconnect                 Reconnect if disconnected
  -d, --debug=level               Debug Level (0 - 7)
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From mrene_lists at avgs.ca  Wed Oct  6 17:29:03 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 6 Oct 2010 20:29:03 -0400
Subject: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4
In-Reply-To: <AANLkTingeYfPT_b6dG3bKQOVjxYsmBTKhDuLw_3BKGac@mail.gmail.com>
References: <AANLkTingeYfPT_b6dG3bKQOVjxYsmBTKhDuLw_3BKGac@mail.gmail.com>
Message-ID: <A499E7B3-9727-4822-B2AB-8FDDCA3213FA@avgs.ca>

Hi,

Did you start FreeSWITCH before? fs_cli is only used to connect to the console of a running FreeSWITCH instance.

If you did, try starting it without the -nc switch, it should stay in the foreground and tell you whats wrong, if you have a console you can also check that mod_event_socket is properly loaded (load mod_event_socket)

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-10-06, at 8:23 PM, Peter Schrock wrote:

> Okay, so I managed to get FS working the other day and I even managed to test a call and test my voicemail. All seemed to be working smoothly until, because of the rain here, my power went out and I had to reboot my computer. I logged in through the terminal, set up FS in background went to fs_cli and I get this error message:
> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection Error]
> I managed to figure out that in the file "fs_cli.c" on line 1206 you find the code for displaying this error message. The problem is that I don't know why this error message is occurring. Does anyone have any helpful hints as to what I should look at to resolve this problem?
> I even tried going to the git tree and make current, but that gave me problems that forced me to turn off mod_spandsp and mod_skyopen in the modules.conf, which I had running earlier. Any thoughts?
> 
> PeterS
> 
> PS
> I am not sure if this is of any help, but in addition to the error line above, it also posted this info:
> Usage: fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x command] [profile]
> 
>   -?,-h --help                    Usage Information
>   -H, --host=hostname             Host to connect
>   -P, --port=port                 Port to connect (1 - 65535)
>   -u, --user=user at domain          user at domain
>   -p, --password=password         Password
>   -x, --execute=command           Execute Command and Exit
>   -l, --loglevel=command          Log Level
>   -q, --quiet                     Disable logging
>   -r, --retry                     Retry connection on failure
>   -R, --reconnect                 Reconnect if disconnected
>   -d, --debug=level               Debug Level (0 - 7)
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From adminjew at gmail.com  Wed Oct  6 18:00:24 2010
From: adminjew at gmail.com (Yitzchok)
Date: Wed, 6 Oct 2010 21:00:24 -0400
Subject: [Freeswitch-users] Mono 2.8 released and need to update mod_managed
Message-ID: <AANLkTin3rM+NaW1_O2d4HaDL9yuESGmZvd8v29Kxh6Y8@mail.gmail.com>

Mono 2.8 was released today with support for C# 4 and more
http://www.mono-project.com/Release_Notes_Mono_2.8

<http://www.mono-project.com/Release_Notes_Mono_2.8>It seems like there was
some changes in the way applications embed mono (which includes removing the
reference to glib and replacing it with eglib from what I understand) in
this link you can find information for the changes made with embedding mono
http://www.mono-project.com/Embedding_Mono

I am trying to see if I can make it work but since I am just a C# windows
developer (not a c or c++ dev) what I am doing is just hacking around and
don't know if I will even get it to work but I think a c++ linux developer
might get it to work with only a little work (the info needed I think is in
the link above).

So I am wondering if anyone is interested to get it to work.

<http://www.mono-project.com/Embedding_Mono>

Yitzchok


On Wed, Oct 6, 2010 at 11:40 AM, Yitzchok <adminjew at gmail.com> wrote:

> I tried that and it seems to work.
>
>
> ---diff from git head---
>
>
> @@ -385,9 +385,7 @@ SWITCH_STANDARD_API(managedrun_api_function)
>    } else {
>   stream->write_function(stream, "-ERR ExecuteBackground returned false
> (unknown module or exception?).\n");
>   }
> -#ifndef _MANAGED
> - mono_thread_detach(mono_thread_current());
> -#endif
> +
>   return SWITCH_STATUS_SUCCESS;
>  }
>
> @@ -403,9 +401,7 @@ SWITCH_STANDARD_API(managed_api_function)
>   if (!(executeDelegate(cmd, stream, stream->param_event))) {
>   switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Execute failed
> for %s (unknown module or exception).\n", cmd);
>   }
> -#ifndef _MANAGED
> - mono_thread_detach(mono_thread_current());
> -#endif
> +
>   return SWITCH_STATUS_SUCCESS;
>  }
>
> @@ -421,9 +417,7 @@ SWITCH_STANDARD_APP(managed_app_function)
>   if (!(runDelegate(data, session))) {
>   switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Application run
> failed for %s (unknown module or exception).\n", data);
>   }
> -#ifndef _MANAGED
> - mono_thread_detach(mono_thread_current());
> -#endif
> +
>  }
>
>  SWITCH_STANDARD_API(managedreload_api_function)
> @@ -438,9 +432,7 @@ SWITCH_STANDARD_API(managedreload_api_function)
>   if (!(reloadDelegate(cmd))) {
>   switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Execute failed
> for %s (unknown module or exception).\n", cmd);
>   }
> -#ifndef _MANAGED
> - mono_thread_detach(mono_thread_current());
> -#endif
> +
>   return SWITCH_STATUS_SUCCESS;
>  }
>
>
>
> -----
>
> Yitzchok
>
>
>
> On Wed, Oct 6, 2010 at 10:57 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
>
>>
>> Not sure how this makes sense yet. Yitzchok would you try this with only
>> the
>> mono_thread_detach calls removed leaving the mono_thread_attach in place
>> and
>> see what happens.
>>
>> Thanks
>> Jeff
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/mod-managed-on-linux-centos-tp5485480p5607396.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From adminjew at gmail.com  Wed Oct  6 18:48:52 2010
From: adminjew at gmail.com (Yitzchok)
Date: Wed, 6 Oct 2010 21:48:52 -0400
Subject: [Freeswitch-users] Mono 2.8 released and need to update
	mod_managed
In-Reply-To: <AANLkTin3rM+NaW1_O2d4HaDL9yuESGmZvd8v29Kxh6Y8@mail.gmail.com>
References: <AANLkTin3rM+NaW1_O2d4HaDL9yuESGmZvd8v29Kxh6Y8@mail.gmail.com>
Message-ID: <AANLkTimoS2=NaMhg2dVVEuyEjwEqagLie3KDVNHYv=8B@mail.gmail.com>

It seems like glib itself is easy to get around by removing the glig
includes and changing g_free(x) to mono_free(x)

But I am getting stuck by this one.


   - The MonoAssemblyName struct is no longer fully visible: to access its
   fields

you need to use the newly-provided accessors. Note also that it can't be
allocated on the stack anymore and you'll need to create and destroy it with
the following API:


MonoAssemlyName *aname = mono_assembly_name_new ("mscorlib");
mono_assembly_name_free (aname);



(from http://www.mono-project.com/Embedding_Mono)

Yitzchok


On Wed, Oct 6, 2010 at 9:00 PM, Yitzchok <adminjew at gmail.com> wrote:

> Mono 2.8 was released today with support for C# 4 and more
> http://www.mono-project.com/Release_Notes_Mono_2.8
>
> <http://www.mono-project.com/Release_Notes_Mono_2.8>It seems like there
> was some changes in the way applications embed mono (which includes removing
> the reference to glib and replacing it with eglib from what I understand) in
> this link you can find information for the changes made with embedding mono
> http://www.mono-project.com/Embedding_Mono
>
> I am trying to see if I can make it work but since I am just a C# windows
> developer (not a c or c++ dev) what I am doing is just hacking around and
> don't know if I will even get it to work but I think a c++ linux developer
> might get it to work with only a little work (the info needed I think is in
> the link above).
>
> So I am wondering if anyone is interested to get it to work.
>
> <http://www.mono-project.com/Embedding_Mono>
>
> Yitzchok
>
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From adminjew at gmail.com  Wed Oct  6 20:20:42 2010
From: adminjew at gmail.com (Yitzchok)
Date: Wed, 6 Oct 2010 23:20:42 -0400
Subject: [Freeswitch-users] Mono 2.8 released and need to update
	mod_managed
In-Reply-To: <AANLkTimoS2=NaMhg2dVVEuyEjwEqagLie3KDVNHYv=8B@mail.gmail.com>
References: <AANLkTin3rM+NaW1_O2d4HaDL9yuESGmZvd8v29Kxh6Y8@mail.gmail.com>
	<AANLkTimoS2=NaMhg2dVVEuyEjwEqagLie3KDVNHYv=8B@mail.gmail.com>
Message-ID: <AANLkTi=_aLLty1F_iFkf_vQRuHzWTYJrZVc-7Zkos=Dh@mail.gmail.com>

I got it to work and attached is the patch file with the changes I made to
get it to work. (Jeff this includes the changes from my other post)

This code changes has to be reviewed and I think you should checkout
*mono_assembly_name_free
(name); *I don't call that but maybe it should be called somewhere in the
code.

Thanks

Yitzchok


On Wed, Oct 6, 2010 at 9:48 PM, Yitzchok <adminjew at gmail.com> wrote:

> It seems like glib itself is easy to get around by removing the glig
> includes and changing g_free(x) to mono_free(x)
>
> But I am getting stuck by this one.
>
>
>    - The MonoAssemblyName struct is no longer fully visible: to access its
>    fields
>
> you need to use the newly-provided accessors. Note also that it can't be
> allocated on the stack anymore and you'll need to create and destroy it with
> the following API:
>
>
> MonoAssemlyName *aname = mono_assembly_name_new ("mscorlib");
> mono_assembly_name_free (aname);
>
>
>
> (from http://www.mono-project.com/Embedding_Mono)
>
> Yitzchok
>
>
>
> On Wed, Oct 6, 2010 at 9:00 PM, Yitzchok <adminjew at gmail.com> wrote:
>
>> Mono 2.8 was released today with support for C# 4 and more
>> http://www.mono-project.com/Release_Notes_Mono_2.8
>>
>> <http://www.mono-project.com/Release_Notes_Mono_2.8>It seems like there
>> was some changes in the way applications embed mono (which includes removing
>> the reference to glib and replacing it with eglib from what I understand) in
>> this link you can find information for the changes made with embedding mono
>> http://www.mono-project.com/Embedding_Mono
>>
>> I am trying to see if I can make it work but since I am just a C# windows
>> developer (not a c or c++ dev) what I am doing is just hacking around and
>> don't know if I will even get it to work but I think a c++ linux developer
>> might get it to work with only a little work (the info needed I think is in
>> the link above).
>>
>> So I am wondering if anyone is interested to get it to work.
>>
>> <http://www.mono-project.com/Embedding_Mono>
>>
>> Yitzchok
>>
>
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From bwibowo at gmail.com  Wed Oct  6 22:06:38 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Thu, 7 Oct 2010 12:06:38 +0700
Subject: [Freeswitch-users] sip client
In-Reply-To: <AANLkTik75hyXxBCFhAvob9_vjUkyGa4N_MRSqfYJ=dQg@mail.gmail.com>
References: <AANLkTimnafsORnCFib+3FCpp=gp0ekhoevGggDzKLfmp@mail.gmail.com>
	<AANLkTi=bsnUVd+L5BFLtQ65viAFSeKuAv_RhNyqTMNGp@mail.gmail.com>
	<AANLkTik75hyXxBCFhAvob9_vjUkyGa4N_MRSqfYJ=dQg@mail.gmail.com>
Message-ID: <AANLkTingm-baqQgx1VxBFcR5hWYe0qRhLZ5-pNxUJxsN@mail.gmail.com>

thx, that looks great but i'm looking the ready to use product that can run
on major mobile phone (blackbery iphone android winmo symbian)
voipswitch is good, but the server running on windows platform. i prefer the
server running on unix platform.
any info please share


regards

budi




On Fri, Oct 1, 2010 at 7:19 AM, jesse <chat2jesse at gmail.com> wrote:

> Try open source client sip communicator that is in java.
> On Sep 30, 2010 2:53 PM, "budi wibowo" <bwibowo at gmail.com> wrote:
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peter.schrock at gmail.com  Wed Oct  6 22:54:52 2010
From: peter.schrock at gmail.com (Peter Schrock)
Date: Wed, 6 Oct 2010 22:54:52 -0700
Subject: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4
In-Reply-To: <A499E7B3-9727-4822-B2AB-8FDDCA3213FA@avgs.ca>
References: <AANLkTingeYfPT_b6dG3bKQOVjxYsmBTKhDuLw_3BKGac@mail.gmail.com>
	<A499E7B3-9727-4822-B2AB-8FDDCA3213FA@avgs.ca>
Message-ID: <AANLkTinOUhDgzCB2rTL20_RzGq3=uS7zOKE064DOR0v6@mail.gmail.com>

In answer to your first question, yes. I guess my next question would be, if
I don't need fs_cli, then what is it for? I guess I am assuming to access FS
when it is running in background. What would be the difference between
running it in the foreground vs background?
Also, I ran FS without -nc and it showed that mod_event_socket loaded
successfully. So why won't it with fs_cli?

PeterS

On Wed, Oct 6, 2010 at 5:29 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
>
> Did you start FreeSWITCH before? fs_cli is only used to connect to the
> console of a running FreeSWITCH instance.
>
> If you did, try starting it without the -nc switch, it should stay in the
> foreground and tell you whats wrong, if you have a console you can also
> check that mod_event_socket is properly loaded (load mod_event_socket)
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2010-10-06, at 8:23 PM, Peter Schrock wrote:
>
> Okay, so I managed to get FS working the other day and I even managed to
> test a call and test my voicemail. All seemed to be working smoothly until,
> because of the rain here, my power went out and I had to reboot my computer.
> I logged in through the terminal, set up FS in background went to fs_cliand I get this error message:
> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
> Error]
> I managed to figure out that in the file "fs_cli.c" on line 1206 you find
> the code for displaying this error message. The problem is that I don't know
> why this error message is occurring. Does anyone have any helpful hints as
> to what I should look at to resolve this problem?
> I even tried going to the git tree and make current, but that gave me
> problems that forced me to turn off mod_spandsp and mod_skyopen in the
> modules.conf, which I had running earlier. Any thoughts?
>
> PeterS
>
> PS
> I am not sure if this is of any help, but in addition to the error line
> above, it also posted this info:
> Usage: fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x
> command] [profile]
>
>   -?,-h --help                    Usage Information
>   -H, --host=hostname             Host to connect
>   -P, --port=port                 Port to connect (1 - 65535)
>   -u, --user=user at domain          user at domain
>   -p, --password=password         Password
>   -x, --execute=command           Execute Command and Exit
>   -l, --loglevel=command          Log Level
>   -q, --quiet                     Disable logging
>   -r, --retry                     Retry connection on failure
>   -R, --reconnect                 Reconnect if disconnected
>   -d, --debug=level               Debug Level (0 - 7)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From covici at ccs.covici.com  Wed Oct  6 23:12:10 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 07 Oct 2010 02:12:10 -0400
Subject: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4
In-Reply-To: <AANLkTinOUhDgzCB2rTL20_RzGq3=uS7zOKE064DOR0v6@mail.gmail.com>
References: <AANLkTingeYfPT_b6dG3bKQOVjxYsmBTKhDuLw_3BKGac@mail.gmail.com>
	<A499E7B3-9727-4822-B2AB-8FDDCA3213FA@avgs.ca>
	<AANLkTinOUhDgzCB2rTL20_RzGq3=uS7zOKE064DOR0v6@mail.gmail.com>
Message-ID: <9026.1286431930@ccs.covici.com>

Did you copy fs_cli.conf to  /etc and make sure its parameters agreed
with those in the config for mod_esl?  

Peter Schrock <peter.schrock at gmail.com> wrote:

> In answer to your first question, yes. I guess my next question would be, if
> I don't need fs_cli, then what is it for? I guess I am assuming to access FS
> when it is running in background. What would be the difference between
> running it in the foreground vs background?
> Also, I ran FS without -nc and it showed that mod_event_socket loaded
> successfully. So why won't it with fs_cli?
> 
> PeterS
> 
> On Wed, Oct 6, 2010 at 5:29 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> 
> > Hi,
> >
> > Did you start FreeSWITCH before? fs_cli is only used to connect to the
> > console of a running FreeSWITCH instance.
> >
> > If you did, try starting it without the -nc switch, it should stay in the
> > foreground and tell you whats wrong, if you have a console you can also
> > check that mod_event_socket is properly loaded (load mod_event_socket)
> >
> > Mathieu Rene
> > Avant-Garde Solutions Inc
> > Office: + 1 (514) 664-1044 x100
> > Cell: +1 (514) 664-1044 x200
> > mrene at avgs.ca
> >
> >
> >
> >
> > On 2010-10-06, at 8:23 PM, Peter Schrock wrote:
> >
> > Okay, so I managed to get FS working the other day and I even managed to
> > test a call and test my voicemail. All seemed to be working smoothly until,
> > because of the rain here, my power went out and I had to reboot my computer.
> > I logged in through the terminal, set up FS in background went to fs_cliand I get this error message:
> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
> > Error]
> > I managed to figure out that in the file "fs_cli.c" on line 1206 you find
> > the code for displaying this error message. The problem is that I don't know
> > why this error message is occurring. Does anyone have any helpful hints as
> > to what I should look at to resolve this problem?
> > I even tried going to the git tree and make current, but that gave me
> > problems that forced me to turn off mod_spandsp and mod_skyopen in the
> > modules.conf, which I had running earlier. Any thoughts?
> >
> > PeterS
> >
> > PS
> > I am not sure if this is of any help, but in addition to the error line
> > above, it also posted this info:
> > Usage: fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x
> > command] [profile]
> >
> >   -?,-h --help                    Usage Information
> >   -H, --host=hostname             Host to connect
> >   -P, --port=port                 Port to connect (1 - 65535)
> >   -u, --user=user at domain          user at domain
> >   -p, --password=password         Password
> >   -x, --execute=command           Execute Command and Exit
> >   -l, --loglevel=command          Log Level
> >   -q, --quiet                     Disable logging
> >   -r, --retry                     Retry connection on failure
> >   -R, --reconnect                 Reconnect if disconnected
> >   -d, --debug=level               Debug Level (0 - 7)
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From peter.olsson at visionutveckling.se  Wed Oct  6 23:14:02 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 7 Oct 2010 08:14:02 +0200
Subject: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4
In-Reply-To: <AANLkTinOUhDgzCB2rTL20_RzGq3=uS7zOKE064DOR0v6@mail.gmail.com>
References: <AANLkTingeYfPT_b6dG3bKQOVjxYsmBTKhDuLw_3BKGac@mail.gmail.com>
	<A499E7B3-9727-4822-B2AB-8FDDCA3213FA@avgs.ca>
	<AANLkTinOUhDgzCB2rTL20_RzGq3=uS7zOKE064DOR0v6@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57E0E68E2E@cooper>

fs_cli is a tool to connect to a running instance of FS, on the same computer, or another computer. It communicates with FS using the mod_event_socket loaded within FS. The server socket in FS listens for requests on port 8021 by default, this can be changed in event_socket.conf.xml

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Peter Schrock
Skickat: den 7 oktober 2010 07:55
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4

In answer to your first question, yes. I guess my next question would be, if I don't need fs_cli, then what is it for? I guess I am assuming to access FS when it is running in background. What would be the difference between running it in the foreground vs background?
Also, I ran FS without -nc and it showed that mod_event_socket loaded successfully. So why won't it with fs_cli?

PeterS
On Wed, Oct 6, 2010 at 5:29 PM, Mathieu Rene <mrene_lists at avgs.ca<mailto:mrene_lists at avgs.ca>> wrote:
Hi,

Did you start FreeSWITCH before? fs_cli is only used to connect to the console of a running FreeSWITCH instance.

If you did, try starting it without the -nc switch, it should stay in the foreground and tell you whats wrong, if you have a console you can also check that mod_event_socket is properly loaded (load mod_event_socket)

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca<mailto:mrene at avgs.ca>




On 2010-10-06, at 8:23 PM, Peter Schrock wrote:

Okay, so I managed to get FS working the other day and I even managed to test a call and test my voicemail. All seemed to be working smoothly until, because of the rain here, my power went out and I had to reboot my computer. I logged in through the terminal, set up FS in background went to fs_cli and I get this error message:
[ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection Error]
I managed to figure out that in the file "fs_cli.c" on line 1206 you find the code for displaying this error message. The problem is that I don't know why this error message is occurring. Does anyone have any helpful hints as to what I should look at to resolve this problem?
I even tried going to the git tree and make current, but that gave me problems that forced me to turn off mod_spandsp and mod_skyopen in the modules.conf, which I had running earlier. Any thoughts?

PeterS

PS
I am not sure if this is of any help, but in addition to the error line above, it also posted this info:
Usage: fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x command] [profile]

  -?,-h --help                    Usage Information
  -H, --host=hostname             Host to connect
  -P, --port=port                 Port to connect (1 - 65535)
  -u, --user=user at domain          user at domain
  -p, --password=password         Password
  -x, --execute=command           Execute Command and Exit
  -l, --loglevel=command          Log Level
  -q, --quiet                     Disable logging
  -r, --retry                     Retry connection on failure
  -R, --reconnect                 Reconnect if disconnected
  -d, --debug=level               Debug Level (0 - 7)

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4cad626f32932643394250!
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From ale975 at gmail.com  Thu Oct  7 00:34:30 2010
From: ale975 at gmail.com (Ale)
Date: Thu, 7 Oct 2010 09:34:30 +0200
Subject: [Freeswitch-users] Fax hint request
In-Reply-To: <0eb201cb6579$de02ba40$9a082ec0$@com>
References: <AANLkTim8NhxTcypcA9kuT4vnmRNObZ2JbZzgVNdGpsgW@mail.gmail.com>
	<AANLkTimmjxQjfzyb=+ZxvJzFDR5EduyM8pLXBXbX+62p@mail.gmail.com>
	<0eb201cb6579$de02ba40$9a082ec0$@com>
Message-ID: <AANLkTimTbh_wCYd4UDE2LXe1NTxWj59g_ttC5cbAdy79@mail.gmail.com>

Hi Peder

On Wed, Oct 6, 2010 at 7:13 PM, Peder <peder at networkoblivion.com> wrote:
> What are you trying to do? ?And what is the problem? ?This works for me to
> receive a fax:
>

I'm trying to send fax, i've not tested rxfax. I've a little script
that send fax via originate command, i also tried to send manually via
cli with the same result.

If i set enable-t38 to false all seems to works, but when i enable it,
all originate fail with the error show in the first mail. It's
absolutely reproducible, so i believe to make something wrong, but i
does not understand what could be. I'm trying to read the
documentation but configuration seems to be correct.

If you want a complete console output, a dump or any other information
feel free to ask, i would be happy if someone point me on the right
direction.

Many thanx
Ale


From steveayre at gmail.com  Thu Oct  7 00:41:51 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 7 Oct 2010 08:41:51 +0100
Subject: [Freeswitch-users] Question about effective caller ID &
 origination caller ID
In-Reply-To: <4CACBEDE.8060805@mgtech.com>
References: <4CACBEDE.8060805@mgtech.com>
Message-ID: <AANLkTikVj0YXKTfWEvhcFn6RLAicK4rSKt-3eXwXNSs+@mail.gmail.com>

Have you tried using effective_* with export?
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_export

-Steve



On 6 October 2010 19:24, Mario <mario_fs at mgtech.com> wrote:
> I have the inbound caller ID working for a bridge but can't reconcile
> why effective_caller_id.. does not work. I spent a lot of time the last
> few days reading the wiki and other web info. Since I can't find the
> answer anywhere can anyone shed some light on this?
>
> The wiki states:
> effective_caller_id_number
> Sets the effective callerid number. This is automatically exported to
> the B-leg; however, it is not valid in an origination string. In other
> words, set this before calling bridge, otherwise use
> origination_caller_id_number
>
> Does NOT work before the bridge:
> ? ? ? ?<!--action application="set"
> data="effective_caller_id_number=${caller_id_number}"/>
> ? ? ? ?<action application="set"
> data="effective_caller_id_name=${caller_id_name}"/-->
>
> Does NOT work before the bridge:
> ? ? ? ?<!--action application="set"
> data="origination_caller_id_number=${caller_id_number}"/>
> ? ? ? ?<action application="set"
> data="origination_caller_id_name=${caller_id_name}"/-->
> ? ? ? ?<!--action application="set"
>
> ONLY way IT WORKS (must be added before dialstring):
> ? ? ? ?<action application="bridge"
> data="<originate_timeout=15,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/MGT@${domain_name}:_:sofia/gateway/uuid1/19162223333:_:sofia/gateway/uuid1/19164445555"/>
>
>
> The caller ID is lost in an extension, although I still don't know why
> at least I got it back.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From steveayre at gmail.com  Thu Oct  7 00:44:02 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 7 Oct 2010 08:44:02 +0100
Subject: [Freeswitch-users] Question about effective caller ID &
 origination caller ID
In-Reply-To: <AANLkTikVj0YXKTfWEvhcFn6RLAicK4rSKt-3eXwXNSs+@mail.gmail.com>
References: <4CACBEDE.8060805@mgtech.com>
	<AANLkTikVj0YXKTfWEvhcFn6RLAicK4rSKt-3eXwXNSs+@mail.gmail.com>
Message-ID: <AANLkTikYPAT2=V7euHq4+R+E1v4VnV=eJ7YnCf7kV0Ma@mail.gmail.com>

It's also possible that it's a difference between enterprise and
normal bridge syntax - I've never used enterprise so don't know too
much about it.

-Steve


On 7 October 2010 08:41, Steven Ayre <steveayre at gmail.com> wrote:
> Have you tried using effective_* with export?
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_export
>
> -Steve
>
>
>
> On 6 October 2010 19:24, Mario <mario_fs at mgtech.com> wrote:
>> I have the inbound caller ID working for a bridge but can't reconcile
>> why effective_caller_id.. does not work. I spent a lot of time the last
>> few days reading the wiki and other web info. Since I can't find the
>> answer anywhere can anyone shed some light on this?
>>
>> The wiki states:
>> effective_caller_id_number
>> Sets the effective callerid number. This is automatically exported to
>> the B-leg; however, it is not valid in an origination string. In other
>> words, set this before calling bridge, otherwise use
>> origination_caller_id_number
>>
>> Does NOT work before the bridge:
>> ? ? ? ?<!--action application="set"
>> data="effective_caller_id_number=${caller_id_number}"/>
>> ? ? ? ?<action application="set"
>> data="effective_caller_id_name=${caller_id_name}"/-->
>>
>> Does NOT work before the bridge:
>> ? ? ? ?<!--action application="set"
>> data="origination_caller_id_number=${caller_id_number}"/>
>> ? ? ? ?<action application="set"
>> data="origination_caller_id_name=${caller_id_name}"/-->
>> ? ? ? ?<!--action application="set"
>>
>> ONLY way IT WORKS (must be added before dialstring):
>> ? ? ? ?<action application="bridge"
>> data="<originate_timeout=15,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/MGT@${domain_name}:_:sofia/gateway/uuid1/19162223333:_:sofia/gateway/uuid1/19164445555"/>
>>
>>
>> The caller ID is lost in an extension, although I still don't know why
>> at least I got it back.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From tculjaga at gmail.com  Thu Oct  7 01:48:54 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 7 Oct 2010 10:48:54 +0200
Subject: [Freeswitch-users] Radius AAA
In-Reply-To: <AANLkTi=GiQaO-48BP=T8c-A+09sJMKzQH7BX=ewBdB8R@mail.gmail.com>
References: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>
	<AANLkTi=GiQaO-48BP=T8c-A+09sJMKzQH7BX=ewBdB8R@mail.gmail.com>
Message-ID: <AANLkTikw8LGDqQki16D30b8JL0eJid2UOTs-bEaZSc2k@mail.gmail.com>

On Thu, Oct 7, 2010 at 1:15 AM, Michael Collins <msc at freeswitch.org> wrote:

> I'm afraid no such comprehensive documentation exists. Submissions to our
> wiki are most welcomed. This is all we have at the moment:
>
> http://wiki.freeswitch.org/wiki/Mod_rad_auth
>
> Be sure to install freeradius2 (server) and freeradius-client before trying
> to install mod_auth_rad.
>
> -MC
>
>
>

correct,

Im the author of the module and im going to provide the documentation and
how-to.


in brief, this module does radius auth (not accounting). It is based on
freeradius-client library and as such this is the only dependency.

you can specify your own list of VSAs to be included in the packet along
with the standard ones that are being used.


         name:       just a description
         value:      direct input or variable
         pec:        vendor ID (0 for default, 9 for cisco...)
         expr:       1 for channel variable, 2 for direct input (string)
         direction:  in for radius-request, out for radius-response


Im not going to describe it here... its better i do it on the wiki itself...




T.
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From rajesh.npnr at yahoo.com  Thu Oct  7 04:21:00 2010
From: rajesh.npnr at yahoo.com (rex.alex)
Date: Thu, 7 Oct 2010 04:21:00 -0700 (PDT)
Subject: [Freeswitch-users] FreeSWITCH parameters for voice quality
Message-ID: <1286450460556-5610915.post@n2.nabble.com>


Hello,


I have FreeSWITCH on Cent OS 5.4 physical server and I am using X-Lite soft
phone on Windows client to make external calling through a registered
gateway. But, I am facing issues like voice breakage and static noise on
calls. I am monitoring the network connectivity but before that I want to
make sure that all FreeSWITCH parameters have been configured correctly.


Please let me know as far as FreeSWITCH is concern what are the parameters
that I have to check for these kind of issues?

Thanks,

Rex
-- 
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From fdelawarde at wirelessmundi.com  Thu Oct  7 05:35:43 2010
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Thu, 07 Oct 2010 14:35:43 +0200
Subject: [Freeswitch-users] FreeSwitch Out of threads!
In-Reply-To: <AANLkTino=xF7HAMXqaKmubswR26ytU4sNkGWB4Ch8ngz@mail.gmail.com>
References: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
	<AANLkTing67qXyj3msDQaxun2XOgocRDVYXvPh9hsZWFW@mail.gmail.com>
	<AANLkTimYA=XM4pPJRrUOA_er5WkhZfqhJQO0E4E6LKgj@mail.gmail.com>
	<1286271927.5790.33.camel@luna.tc.commsmundi.com>
	<AANLkTi=Gis9k=pHQ1si5cXeOCjk6EskjN0ri3Nak9YVE@mail.gmail.com>
	<AANLkTingwTJpYLtXChhGyzw6jPLq2ot_82bdptmRZR18@mail.gmail.com>
	<1286290551.5790.58.camel@luna.tc.commsmundi.com>
	<AANLkTinhiuJUQA26WevUPXiOtMHb=5bXKTR3N2dv3vrS@mail.gmail.com>
	<AANLkTimr9mqzT6UUULWHvim5bWGUwLakYs_1=J8tHPFX@mail.gmail.com>
	<1286293526.5790.71.camel@luna.tc.commsmundi.com>
	<AANLkTi=hTTkfXY_o4ZKNHt21x4EJvq-bBXCU2YzkeyFc@mail.gmail.com>
	<AANLkTino=xF7HAMXqaKmubswR26ytU4sNkGWB4Ch8ngz@mail.gmail.com>
Message-ID: <1286454943.5850.70.camel@luna.tc.commsmundi.com>

I can't reproduce so far after updating to HEAD, I'll post a Jira issue
if it happens again.

How do you guys do to always resolve all my problems in a few hours or
days at max? I won't believe it anymore if you say you used to be an
Asterisk bug marshal (I'm used to wait months for issues to be attended
there)... :-)

Thanks,
Fran?ois.


On Tue, 2010-10-05 at 11:17 -0500, Anthony Minessale wrote:
> Can you update first to this revision of HEAD or later?
> 
> commit 80883ebeb81c73a3dc5c4ee65050d57e2c6ccf00
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date:   Tue Oct 5 11:11:21 2010 -0500
> 
> I added a small patch that I think may be related to your problem.
> 
> 
> On Tue, Oct 5, 2010 at 10:59 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > if you are on centos or something like it.
> > you can run this from the FS build root.
> >
> > ./support-d/fscore_pb gcore <your nick>
> >
> > while FS is running in this state.
> >
> > That will give you a URL you can tell me, it will upload the trace I
> > need to see right to our pastebin.
> >
> > If you can get it that way, and add our ssh key from
> > ./support-d/shinzon.pub, install it and email or irc pm me the
> > hostname.  And I can examine it.
> >
> >
> > On Tue, Oct 5, 2010 at 10:45 AM, Fran?ois Delawarde
> > <fdelawarde at wirelessmundi.com> wrote:
> >> GIT from 1st of October.
> >>
> >> Fran?ois.
> >>
> >> On Tue, 2010-10-05 at 10:28 -0500, Anthony Minessale wrote:
> >>> BTW,
> >>>
> >>> If you are any less than this revision don't bother to report it until
> >>> you have tested latest GIT
> >>>
> >>> commit b02c69bba9f46cad37225a0986cae068f24dcd81
> >>> Author: Anthony Minessale <anthm at freeswitch.org>
> >>> Date:   Wed Sep 22 18:14:24 2010 -0500
> >>>
> >>>
> >>>
> >>> On Tue, Oct 5, 2010 at 10:26 AM, Anthony Minessale
> >>> <anthony.minessale at gmail.com> wrote:
> >>> > If there is a bug why are you not opening it on JIRA?
> >>> > http://jira.freeswitch.org
> >>> >
> >>> > We are doing work on improving the event system to be faster and use
> >>> > less memory.
> >>> >
> >>> > If it caused some problems, that is not uncommon, software development
> >>> > requires testing.  The right thing to do is to report it not to make
> >>> > underhanded comments about rethinking your strategy.
> >>> >
> >>> > Ideally I would like to visit the machine while it's in that state or
> >>> > otherwise I would like you to get a gcore while it's int that state,
> >>> > open it in gdb and produce a "thread apply all bt" trace in text
> >>> > format and attach it to the JIRA.
> >>> >
> >>> >
> >>> >
> >>> >
> >>> > On Tue, Oct 5, 2010 at 9:55 AM, Fran?ois Delawarde
> >>> > <fdelawarde at wirelessmundi.com> wrote:
> >>> >> I wouldn't jump to conclusions that fast, it's just a bug with current
> >>> >> git head, probably only happening in very specific cases.
> >>> >>
> >>> >> If I learned anything from hollywood movies, it would be that bugs or
> >>> >> bad people always end up being tracked down and destroyed.
> >>> >>
> >>> >> Fran?ois.
> >>> >>
> >>> >>
> >>> >> On Tue, 2010-10-05 at 20:35 +0600, Nyamul Hassan wrote:
> >>> >>> We learned from another thread about how Sangoma regularly tests 32 E1
> >>> >>> setups in the lab, and were preparing a concept paper to show a
> >>> >>> telecom that traditional vendors can be replaced using open source
> >>> >>> even for large scale implementations. Does this mean we should rethink
> >>> >>> our strategy?
> >>> >>>
> >>> >>> Regards
> >>> >>> HASSAN
> >>> >>>
> >>> >>>
> >>> >>> On 2010-10-05, Moises Silva <moises.silva at gmail.com> wrote:
> >>> >>> > On Tue, Oct 5, 2010 at 5:45 AM, Fran?ois Delawarde
> >>> >>> > <fdelawarde at wirelessmundi.com> wrote:
> >>> >>> >> I maintain "only" 100 sessions at all times, with a duration of 20-30s,
> >>> >>> >> and no more than 10cps, so it is NOT a stress test, come on!
> >>> >>> >>
> >>> >>> >> I mean this would not even load an Asterisk server, and loading the
> >>> >>> >> server was not an objective, just to maintain low-medium load over a
> >>> >>> >> period of time. It worked very well during 5h.
> >>> >>> >>
> >>> >>> >> Why would I need to ulimit the crap out of my server then? :-)
> >>> >>> >
> >>> >>> > Don't feel alone, you're not the only one.
> >>> >>> >
> >>> >>> > We're getting the same pair of messages. We know what it means, we
> >>> >>> > have just not found the cause of it. In our case happens quite fast
> >>> >>> > (about an hour IIRC) when doing stress testing of about 900 calls at a
> >>> >>> > time lasting 10 seconds each at a rate of about 40cps.
> >>> >>> >
> >>> >>> > Moises Silva
> >>> >>> > Senior Software Engineer
> >>> >>> > Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
> >>> >>> > L3R 9R6 Canada
> >>> >>> > t. 1 905 474 1990 x128 | e. moy at sangoma.com
> >>> >>> >
> >>> >>> > _______________________________________________
> >>> >>> > FreeSWITCH-users mailing list
> >>> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >>> > http://www.freeswitch.org
> >>> >>> >
> >>> >>>
> >>> >>
> >>> >>
> >>> >> _______________________________________________
> >>> >> FreeSWITCH-users mailing list
> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> 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
> >>> >
> >>>
> >>>
> >>>
> >>> --
> >>> 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
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> 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
> >
> 
> 
> 



From neilp at cs.stanford.edu  Thu Oct  7 05:52:22 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Thu, 7 Oct 2010 18:22:22 +0530
Subject: [Freeswitch-users] outbound calling error: multiple PRIs with
	Sangoma + FS
Message-ID: <AANLkTimn7P64Exf9JTwdLZB8Vam8wBLuvrLS5MY+xAwT@mail.gmail.com>

Hi All,

I am having a problem getting outbound calls to connect on one of my two PRI
lines connected to FS via a Sangoma A102 card. Here are my config files:

Bottom of /etc/wanpipe/smg_pri.conf:

*;AFT-A102 on port 1
signalling=pri_cpe
switchtype=euroisdn
group=1
spans=1

;AFT-A102 on port 2
signalling=pri_cpe
switchtype=euroisdn
group=2
spans=2

*
openzap.conf:

*[span wanpipe smg_prid]
name => smg_prid
trunk_type =>e1
b-channel => 1:1-15
b-channel => 1:17-31
trunk_type =>e1
b-channel => 2:1-15
b-channel => 2:17-31*

autoload_configs/openzap.conf.xml:

*<configuration name="openzap.conf" description="OpenZAP Configuration">*
*  <settings>*
*    <param name="debug" value="0"/>*
*   *
*  </settings>*
*  <boost_spans>*
*        <span id="smg_prid">*
*        <param name="dialplan" value="XML"/>*
*        <param name="context" value="default"/>*
*        </span>*
*  </boost_spans>*
*  *
*  <analog_spans>*
*        <span id="smg_prid">*
*        <param name="dialplan" value="XML"/>*
*        <param name="context" value="default"/>*
*      *
 *        </span>*
*  </analog_spans>*
*</configuration>*



...and here is the output from FS CLI. Note that the PRI line connected to
port 2 (g2) works great; it is g1 that is producing the NO_ROUTE_DESTINATION
error:

*freeswitch at otalo> originate openzap/smg_prid/a/958655XXXX at g1 &echo*
2010-10-07 18:08:25.273814 [WARNING] ozmod_sangoma_boost.c:344 TX EVENT:
CALL_START:(80) [w1g1] CSid=5 Seq=41 Cn=[N/A] Cd=[958655XXXX] Ci=[N/A]
Rdnis=[]
2010-10-07 18:08:25.306941 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT
(N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=5 Seq=34
2010-10-07 18:08:25.307953 [NOTICE] switch_channel.c:675 New Channel
OpenZAP/1:1/958655XXXX at g1 [c4305297-3a6e-4bd8-a25c-71e96da6b77d]
2010-10-07 18:08:25.411216 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT
(N): CALL_STOPPED:(85) [w1g1] Rc=3 CSid=5 Seq=35
2010-10-07 18:08:25.411216 [NOTICE] mod_openzap.c:1935 Hangup
OpenZAP/1:1/958655XXXX at g1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]

-ERR NO_ROUTE_DESTINATION

2010-10-07 18:08:25.412218 [NOTICE] switch_core_session.c:1188 Session 16
(OpenZAP/1:1/958655XXXX at g1) Ended
2010-10-07 18:08:25.412218 [NOTICE] switch_core_session.c:1190 Close Channel
OpenZAP/1:1/958655XXXX at g1 [CS_DESTROY]
2010-10-07 18:08:25.441373 [WARNING] sangoma_boost_client.c:221 TX EVENT
(N): CALL_STOPPED_ACK:(86) [w1g1] Rc=0 CSid=0 Seq=42
*freeswitch at otalo> originate openzap/smg_prid/a/958655XXXX at g2 &echo*
2010-10-07 18:08:30.199197 [WARNING] ozmod_sangoma_boost.c:344 TX EVENT:
CALL_START:(80) [w1g1] CSid=6 Seq=43 Cn=[N/A] Cd=[958655XXXX] Ci=[N/A]
Rdnis=[]
2010-10-07 18:08:30.263078 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT
(N): CALL_START_ACK:(81) [w2g1] Rc=0 CSid=6 Seq=36
2010-10-07 18:08:30.264081 [NOTICE] switch_channel.c:675 New Channel
OpenZAP/1:31/958655XXXX at g2 [fe1dad03-1086-4d56-bc55-62563e170a78]
2010-10-07 18:08:30.493273 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT:
CALL PROGRESS:(50) [w2g1] CSid=6 Seq=37 Cn=[N/A] Cd=[N/A] Ci=[N/A] Rdnis=[]
2010-10-07 18:08:30.493273 [NOTICE] mod_openzap.c:1962 Pre-Answer
OpenZAP/1:31/958655XXXX at g2!

+OK fe1dad03-1086-4d56-bc55-62563e170a78

I don't notice much going on in /var/log/sangoma_mgd.log or
/var/log/messages, but then again I'm not sure what to look for. Any
guidance would be greatly appreciated!
Neil
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From nazim.aghabayov at gmail.com  Thu Oct  7 05:53:56 2010
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Thu, 07 Oct 2010 17:53:56 +0500
Subject: [Freeswitch-users] Radius AAA
In-Reply-To: <AANLkTikw8LGDqQki16D30b8JL0eJid2UOTs-bEaZSc2k@mail.gmail.com>
References: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>	<AANLkTi=GiQaO-48BP=T8c-A+09sJMKzQH7BX=ewBdB8R@mail.gmail.com>
	<AANLkTikw8LGDqQki16D30b8JL0eJid2UOTs-bEaZSc2k@mail.gmail.com>
Message-ID: <4CADC2E4.90608@gmail.com>

  Hello Tihomir,

I've just updated the Mod_rad_auth wiki with installation instructions.
Thanks a lot for the mod. I use it in production and it works just great!

Regards,
Nazim

On 10/07/2010 01:48 PM, Tihomir Culjaga wrote:
> On Thu, Oct 7, 2010 at 1:15 AM, Michael Collins<msc at freeswitch.org>  wrote:
>
>> I'm afraid no such comprehensive documentation exists. Submissions to our
>> wiki are most welcomed. This is all we have at the moment:
>>
>> http://wiki.freeswitch.org/wiki/Mod_rad_auth
>>
>> Be sure to install freeradius2 (server) and freeradius-client before trying
>> to install mod_auth_rad.
>>
>> -MC
>>
>>
>>
> correct,
>
> Im the author of the module and im going to provide the documentation and
> how-to.
>
>
> in brief, this module does radius auth (not accounting). It is based on
> freeradius-client library and as such this is the only dependency.
>
> you can specify your own list of VSAs to be included in the packet along
> with the standard ones that are being used.
>
>
>           name:       just a description
>           value:      direct input or variable
>           pec:        vendor ID (0 for default, 9 for cisco...)
>           expr:       1 for channel variable, 2 for direct input (string)
>           direction:  in for radius-request, out for radius-response
>
>
> Im not going to describe it here... its better i do it on the wiki itself...
>
>
>
>
> T.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From rodrigoferreiralang at gmail.com  Wed Oct  6 17:24:54 2010
From: rodrigoferreiralang at gmail.com (Rodrigo Lang)
Date: Thu, 7 Oct 2010 01:24:54 +0100
Subject: [Freeswitch-users] SIP and FreeSWITCH presented at FATEC
In-Reply-To: <AANLkTinP=1P4rdENEY95fOWBZKwi=tk2tGT=txfK-o6o@mail.gmail.com>
References: <4CAC785B.5070101@voicetechnology.com.br>
	<AANLkTimRak4JZ1fdShqz9by3q13M88uGKFtMXWT1MVVQ@mail.gmail.com>
	<AANLkTinZ3wTd_AJ4n=vVY1ddczw-fYigaAMOTVo=5Dom@mail.gmail.com>
	<AANLkTinP=1P4rdENEY95fOWBZKwi=tk2tGT=txfK-o6o@mail.gmail.com>
Message-ID: <AANLkTimzFLfZhR+y4hsAQovKyGTgYEsaWs3BRpyRvoPz@mail.gmail.com>

Yes, Khomp is the best hardware i have used to Asterisk, is even better than
the Digium. I think is a success and is great to the FreeSwitch this support
from Khomp, mainly in the Brazil and Latin American.


Regards,
Rodrigo Lang.



Em 7 de outubro de 2010 00:51, Jo?o Mesquita <jmesquita at freeswitch.org>escreveu:

> Brazilian and Latin American FreeSWITCH? users should be tuned in the up
> coming news from Khomp. They are getting ready to release their officially
> supported mod_khomp on git anytime now.
>
> Regards,
> Jo?o Mesquita
>
>
>
> On Wed, Oct 6, 2010 at 8:33 PM, Rodrigo Lang <
> rodrigoferreiralang at gmail.com> wrote:
>
>> Many VoIP professionals in Brazil is interessed in the FreeSwitch, I'm
>> one.
>>
>>
>> Regards,
>> Rodrigo Lang.
>>
>>
>> 2010/10/7 Michael Collins <msc at freeswitch.org>
>>
>> Glad to hear it! Thanks!
>>> -MC
>>>
>>> On Wed, Oct 6, 2010 at 6:23 AM, Fernando Gregianin Testa <
>>> testa at voicetechnology.com.br> wrote:
>>>
>>>> Hi folks,
>>>>
>>>> You may be pleased to know that FreeSWITCH was presented at IV Semana de
>>>> Tecnologia at FATEC Carapicu?ba (http://www.fateccarapicuiba.com.br/),
>>>> near S?o Paulo, Brazil. FATEC is a state funding faculty that offers a
>>>> sort of undergraduate courses in IT, logistics, games, etc. Around 150
>>>> students attended the lecture with clear interest.
>>>> Presentation files are available (in pt-BR) at
>>>>  http://dl.dropbox.com/u/410277/fatec/index.html
>>>>
>>>> Fernando G. Testa
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Rodrigo Lang
>> http://rodrigorecipes.blogspot.com/<http://rodrigorecipes.blogspot.com/2010/08/ssh-rapido-e-pratico.html>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Rodrigo Lang
http://rodrigorecipes.blogspot.com/<http://rodrigorecipes.blogspot.com/2010/08/ssh-rapido-e-pratico.html>
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From jeff at jefflenk.com  Thu Oct  7 06:46:29 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 7 Oct 2010 06:46:29 -0700 (PDT)
Subject: [Freeswitch-users] ODBC and MSSQL
In-Reply-To: <4CACE7C5.6060208@ParcelPool.com>
References: <4CAB9F5A.2090503@ParcelPool.com>
	<1286331931285-5605569.post@n2.nabble.com>
	<4CACE7C5.6060208@ParcelPool.com>
Message-ID: <1286459189732-5611425.post@n2.nabble.com>


Hi Peter,

If your so inclined please test the patch here:
http://jira.freeswitch.org/browse/FS-2050

and please let me know how it works for you.

After patching and building
remove commented -> <!--  -->


-Jeff
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ODBC-and-MSSQL-tp5605076p5611425.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mike at van.lammeren.net  Thu Oct  7 06:50:54 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Thu, 7 Oct 2010 09:50:54 -0400
Subject: [Freeswitch-users] Question about Lua freeswitch.bridge
In-Reply-To: <AANLkTi=fx8jjpyj6MZYea-NyLBfcq1er7-myv_MNWKw_@mail.gmail.com>
References: <AANLkTimc9ChvXn=_Sz5ZSkJ21hng=YWbmAhs_f9VwDsZ@mail.gmail.com>
	<AANLkTi=fx8jjpyj6MZYea-NyLBfcq1er7-myv_MNWKw_@mail.gmail.com>
Message-ID: <AANLkTikKAvkKa2U5Z-53wMq=wQ8NBTgPg+PT9Mq4AQKV@mail.gmail.com>

Hello!

Thank you for your suggestions!

I took a look at bind_meta_app in the wiki, and although it might be
possible to do what I need, I think the best route will be to re-write my
app to use mod_event_socket. There are some other feature requests that will
be easier to support that way. I think I'm at the limit of what can be done
with a Lua script.

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app
http://wiki.freeswitch.org/wiki/Event_Socket

I'll also be placing an order for the FreeSWITCH book!

Thanks again!

Mike van Lammeren


On Wed, Oct 6, 2010 at 7:44 PM, Michael Collins <msc at freeswitch.org> wrote:

> Just curious - would a bind_meta_app call work for you? I ask because you
> can't do asynchronous operations like you described when calling scripts
> from the dialplan. I'm afraid you will need to graduate to ESL. :) Check out
> ESL and the event socket on the wiki and chapter 9 of the FreeSWITCH book.
>
> Some things you will want to check out:
> uuid_broadcast (to play media to a channel)
> the DTMF event (http://wiki.freeswitch.org/wiki/Event_list#DTMF)
> fs-ivrd (example of a daemon that sits there and connects outbound event
> socket connects to an ESL script)
>
> Using the event socket allows total and complete 3rd party control of your
> sessions. The drawback is the the learning curve. It's a bit steep but it's
> worth it.
>
> -MC
>
>
> On Wed, Oct 6, 2010 at 9:18 AM, Mike van Lammeren <mike at van.lammeren.net>wrote:
>
>> Hello!
>>
>> I'm a huge FreeSWITCH fan and am always amazed at how much I can do with
>> it. I have a Lua script that has been working great, but now have been
>> tasked with enhancing it, and have run into a problem.
>>
>> I want to be able to play media, detect DTMF and hang up a call after
>> bridging two sessions, but my Lua script blocks on freeswitch.bridge(), and
>> doesn't resume until one side or the other hangs up.
>>
>> Here is my problem reduced to a Lua script in its simplest form:
>>
>> -- first session
>> session1 = session
>> -- call out
>> session2 = freeswitch.Session("{ignore_early_media=true}sofia/gateway/
>> etc.com/" .. phoneNumber)
>> -- bridge the calls
>> freeswitch.bridge(session1, session2)
>> -- this next line doesn't execute until after either session 1 or 2 hangs
>> up
>> freeswitch.consoleLog("info", "This is a test.\n")
>>
>>
>> How can I do an asynchronous bridge, and still be able to play media and
>> detect dtmf afterwards?
>>
>> Mike van Lammeren
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Joshua.Foshee at LogixCom.com  Thu Oct  7 07:36:26 2010
From: Joshua.Foshee at LogixCom.com (Joshua Foshee)
Date: Thu, 7 Oct 2010 09:36:26 -0500
Subject: [Freeswitch-users] Application Play_fsv issues
Message-ID: <06502C073AD9394AADB3CA7FD94931BC0519DF35@okc1x1.Logixcom.com>

No one has any input on this?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Joshua Foshee
Sent: Wednesday, October 06, 2010 10:29 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Application Play_fsv issues

I have a problem that if I play back a FSV file that it plays
super-fast. It records fine though when it plays back I see the video
and auto go by fast and then it plays a long busy tone. Any ideas?

Thanks in advance,

Josh

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


From pkelly at gmail.com  Thu Oct  7 07:37:35 2010
From: pkelly at gmail.com (Pete Kelly)
Date: Thu, 7 Oct 2010 15:37:35 +0100
Subject: [Freeswitch-users] Possible bug with bypass_media=true functionality
Message-ID: <AANLkTimqMMVvKR2PojPMnnnp-0KtWsHj43inGy43PDAN@mail.gmail.com>

Hi, I posted a question to this list yesterday because I was having problems
with my calls ending as soon as they were bridged. I was originating 2
outbound calls from a lua script, then bridging them. I managed to get round
the problem of the calls shutting down immediately by initiating calls using
the originate command, thusly:

originate sofia/lpmedia/2000 at 10.15.20.122 '&lua(click2call_call.lua)'

This causes the lua script to initiate with a session already present and
then a bridge to a new call (legB) completes successfully.

Within the lua script I am bridging to leg B like this:

session:execute("bridge", "[bypass_media_after_bridge=true]sofia/lpmedia/
91979197 at 10.15.20.122")

This works, but the reINVITEs which get sent out after legB is established
contain the wrong sip request URI.

- The SIP URI in the reINVITE to Leg A is to the MEDIA IP of leg B (the SIP
packet is sent to the correct IP, just the request URI contains the media
IP)
- The SIP URI in the reINVITE to Leg B is to the MEDIA IP of leg A... again
the SIP packet is sent to the correct IP, it's just the request URI which
contains the media IP.

In some instances the request URI doesn't contain the username either, it is
just INVITE sip:<MEDIA_IP>

I am using the latest sources of freeswitch from git

Is this a bug I need to report? It is causing me to receive 404 and 402
errors on the reinvite as the request URI is invalid.

Pete
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From brian at freeswitch.org  Thu Oct  7 08:29:37 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 7 Oct 2010 10:29:37 -0500
Subject: [Freeswitch-users] Application Play_fsv issues
In-Reply-To: <06502C073AD9394AADB3CA7FD94931BC0519DF35@okc1x1.Logixcom.com>
References: <06502C073AD9394AADB3CA7FD94931BC0519DF35@okc1x1.Logixcom.com>
Message-ID: <B2E8B944-1604-47F0-BC6B-A685FABACE92@freeswitch.org>

please funnel thru jira any issue you feel is a bug... I would rather close the issue as not a bug then lose track of it.

/b

On Oct 7, 2010, at 9:36 AM, Joshua Foshee wrote:

> No one has any input on this?
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Joshua Foshee
> Sent: Wednesday, October 06, 2010 10:29 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Application Play_fsv issues
> 
> I have a problem that if I play back a FSV file that it plays
> super-fast. It records fine though when it plays back I see the video
> and auto go by fast and then it plays a long busy tone. Any ideas?
> 
> Thanks in advance,
> 
> Josh



From mario_fs at mgtech.com  Thu Oct  7 08:42:57 2010
From: mario_fs at mgtech.com (Mario)
Date: Thu, 07 Oct 2010 08:42:57 -0700
Subject: [Freeswitch-users] Question about effective caller ID &
 origination caller ID
In-Reply-To: <AANLkTikYPAT2=V7euHq4+R+E1v4VnV=eJ7YnCf7kV0Ma@mail.gmail.com>
References: <4CACBEDE.8060805@mgtech.com>	<AANLkTikVj0YXKTfWEvhcFn6RLAicK4rSKt-3eXwXNSs+@mail.gmail.com>
	<AANLkTikYPAT2=V7euHq4+R+E1v4VnV=eJ7YnCf7kV0Ma@mail.gmail.com>
Message-ID: <4CADEA81.5060007@mgtech.com>

Thanks for responding Steve, I am beginning to wonder if this is by
design or a bug. I am closer to the end of building the system for our
small office and I thought this would be an easy one. Not so, I already
checked enterprise (:_:) vs (,) and that is where the problem is. I
tried the exports you suggested and had high hopes but no dice. Seems
when using :_: everything is lost and has to be passed as part of the
dialstring.  BTW, I need :_: because the SPA962 phones I use now don't
use SUBSCRIBE only NOTIFY for the mailbox so I have to use multiple
registrations since we are not big and share mailboxes. Thanks for the
help. I wonder if there is a way to determine if this is design or bug....

Both don't fix the problem:
<action application="export"
data="effective_caller_id_number=${caller_id_number}"/

<action application="export"
data="origination_caller_id_number=${caller_id_number}"/


On 10/07/10 00:44, Steven Ayre wrote:
> It's also possible that it's a difference between enterprise and
> normal bridge syntax - I've never used enterprise so don't know too
> much about it.
> 
> -Steve
> 
> 
> On 7 October 2010 08:41, Steven Ayre <steveayre at gmail.com> wrote:
>> Have you tried using effective_* with export?
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_export
>>
>> -Steve
>>
>>
>>
>> On 6 October 2010 19:24, Mario <mario_fs at mgtech.com> wrote:
>>> I have the inbound caller ID working for a bridge but can't reconcile
>>> why effective_caller_id.. does not work. I spent a lot of time the last
>>> few days reading the wiki and other web info. Since I can't find the
>>> answer anywhere can anyone shed some light on this?
>>>
>>> The wiki states:
>>> effective_caller_id_number
>>> Sets the effective callerid number. This is automatically exported to
>>> the B-leg; however, it is not valid in an origination string. In other
>>> words, set this before calling bridge, otherwise use
>>> origination_caller_id_number
>>>
>>> Does NOT work before the bridge:
>>>        <!--action application="set"
>>> data="effective_caller_id_number=${caller_id_number}"/>
>>>        <action application="set"
>>> data="effective_caller_id_name=${caller_id_name}"/-->
>>>
>>> Does NOT work before the bridge:
>>>        <!--action application="set"
>>> data="origination_caller_id_number=${caller_id_number}"/>
>>>        <action application="set"
>>> data="origination_caller_id_name=${caller_id_name}"/-->
>>>        <!--action application="set"
>>>
>>> ONLY way IT WORKS (must be added before dialstring):
>>>        <action application="bridge"
>>> data="<originate_timeout=15,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/MGT@${domain_name}:_:sofia/gateway/uuid1/19162223333:_:sofia/gateway/uuid1/19164445555"/>
>>>
>>>
>>> The caller ID is lost in an extension, although I still don't know why
>>> at least I got it back.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From anthony.minessale at gmail.com  Thu Oct  7 08:52:41 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 7 Oct 2010 10:52:41 -0500
Subject: [Freeswitch-users] FreeSWITCH parameters for voice quality
In-Reply-To: <1286450460556-5610915.post@n2.nabble.com>
References: <1286450460556-5610915.post@n2.nabble.com>
Message-ID: <AANLkTi=aVU7Ojv_xYQEDcvvrztq-FH20_mOHGk_2btLF@mail.gmail.com>

The defaults should be ideal.

On Thu, Oct 7, 2010 at 6:21 AM, rex.alex <rajesh.npnr at yahoo.com> wrote:
> Hello,
>
> I have FreeSWITCH on Cent OS 5.4 physical server and I am using X-Lite soft
> phone on Windows client to make external calling through a registered
> gateway. But, I am facing issues like voice breakage and static noise on
> calls. I am monitoring the network connectivity but before that I want to
> make sure that all FreeSWITCH parameters have been configured correctly.
>
> Please let me know as far as FreeSWITCH is concern what are the parameters
> that I have to check for these kind of issues?
>
> Thanks,
> Rex
> ________________________________
> View this message in context: FreeSWITCH parameters for voice quality
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From sobolewski at gmail.com  Thu Oct  7 08:52:50 2010
From: sobolewski at gmail.com (Piotr Sobolewski)
Date: Thu, 7 Oct 2010 17:52:50 +0200
Subject: [Freeswitch-users] RPID in 183 Session progress
Message-ID: <AANLkTik-as7poMiPivJkzRm4v0YcR1Y69WOUDoz6yKJz@mail.gmail.com>

Hi

I have strange situation where FreeSWITCH ads Remote-party-ID to 183
session progress received from upstream GW.
Diagram below demonstrates this issue.

UA1 ------> PROXY ------> FS  (INVITE ruri=777 with RPID=123)
UA1 <------ PROXY <------- FS  (100 Trying)
FS ----> GW   (INVITE with RPID=123)
UA1 <----( 183 session with RPID=777)---- FS <-----(183 session
without RPID)------ GW

I've tried both
  <action application="set" data="sip_cid_type=none"/>
and
  <action application="export" data="sip_cid_type=none"/>
in the dialplan but without success (still have RPID in the 183)

Is it possible to disable adding of RPID to  SIP response 183 Session Progress?


-- 
Piotr Sobolewski
sobolewski at gmail.com


From anthony.minessale at gmail.com  Thu Oct  7 08:57:10 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 7 Oct 2010 10:57:10 -0500
Subject: [Freeswitch-users] FreeSwitch Out of threads!
In-Reply-To: <1286454943.5850.70.camel@luna.tc.commsmundi.com>
References: <1286266778.5790.22.camel@luna.tc.commsmundi.com>
	<AANLkTing67qXyj3msDQaxun2XOgocRDVYXvPh9hsZWFW@mail.gmail.com>
	<AANLkTimYA=XM4pPJRrUOA_er5WkhZfqhJQO0E4E6LKgj@mail.gmail.com>
	<1286271927.5790.33.camel@luna.tc.commsmundi.com>
	<AANLkTi=Gis9k=pHQ1si5cXeOCjk6EskjN0ri3Nak9YVE@mail.gmail.com>
	<AANLkTingwTJpYLtXChhGyzw6jPLq2ot_82bdptmRZR18@mail.gmail.com>
	<1286290551.5790.58.camel@luna.tc.commsmundi.com>
	<AANLkTinhiuJUQA26WevUPXiOtMHb=5bXKTR3N2dv3vrS@mail.gmail.com>
	<AANLkTimr9mqzT6UUULWHvim5bWGUwLakYs_1=J8tHPFX@mail.gmail.com>
	<1286293526.5790.71.camel@luna.tc.commsmundi.com>
	<AANLkTi=hTTkfXY_o4ZKNHt21x4EJvq-bBXCU2YzkeyFc@mail.gmail.com>
	<AANLkTino=xF7HAMXqaKmubswR26ytU4sNkGWB4Ch8ngz@mail.gmail.com>
	<1286454943.5850.70.camel@luna.tc.commsmundi.com>
Message-ID: <AANLkTimWGcFyKg0UNp8J_8bf5dkwJkT_zE5Qj7u79KCu@mail.gmail.com>

Think of FreeSWITCH as our personal solution to our unsolvable bugs over there.


On Thu, Oct 7, 2010 at 7:35 AM, Fran?ois Delawarde
<fdelawarde at wirelessmundi.com> wrote:
> I can't reproduce so far after updating to HEAD, I'll post a Jira issue
> if it happens again.
>
> How do you guys do to always resolve all my problems in a few hours or
> days at max? I won't believe it anymore if you say you used to be an
> Asterisk bug marshal (I'm used to wait months for issues to be attended
> there)... :-)
>
> Thanks,
> Fran?ois.
>
>
> On Tue, 2010-10-05 at 11:17 -0500, Anthony Minessale wrote:
>> Can you update first to this revision of HEAD or later?
>>
>> commit 80883ebeb81c73a3dc5c4ee65050d57e2c6ccf00
>> Author: Anthony Minessale <anthm at freeswitch.org>
>> Date: ? Tue Oct 5 11:11:21 2010 -0500
>>
>> I added a small patch that I think may be related to your problem.
>>
>>
>> On Tue, Oct 5, 2010 at 10:59 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > if you are on centos or something like it.
>> > you can run this from the FS build root.
>> >
>> > ./support-d/fscore_pb gcore <your nick>
>> >
>> > while FS is running in this state.
>> >
>> > That will give you a URL you can tell me, it will upload the trace I
>> > need to see right to our pastebin.
>> >
>> > If you can get it that way, and add our ssh key from
>> > ./support-d/shinzon.pub, install it and email or irc pm me the
>> > hostname. ?And I can examine it.
>> >
>> >
>> > On Tue, Oct 5, 2010 at 10:45 AM, Fran?ois Delawarde
>> > <fdelawarde at wirelessmundi.com> wrote:
>> >> GIT from 1st of October.
>> >>
>> >> Fran?ois.
>> >>
>> >> On Tue, 2010-10-05 at 10:28 -0500, Anthony Minessale wrote:
>> >>> BTW,
>> >>>
>> >>> If you are any less than this revision don't bother to report it until
>> >>> you have tested latest GIT
>> >>>
>> >>> commit b02c69bba9f46cad37225a0986cae068f24dcd81
>> >>> Author: Anthony Minessale <anthm at freeswitch.org>
>> >>> Date: ? Wed Sep 22 18:14:24 2010 -0500
>> >>>
>> >>>
>> >>>
>> >>> On Tue, Oct 5, 2010 at 10:26 AM, Anthony Minessale
>> >>> <anthony.minessale at gmail.com> wrote:
>> >>> > If there is a bug why are you not opening it on JIRA?
>> >>> > http://jira.freeswitch.org
>> >>> >
>> >>> > We are doing work on improving the event system to be faster and use
>> >>> > less memory.
>> >>> >
>> >>> > If it caused some problems, that is not uncommon, software development
>> >>> > requires testing. ?The right thing to do is to report it not to make
>> >>> > underhanded comments about rethinking your strategy.
>> >>> >
>> >>> > Ideally I would like to visit the machine while it's in that state or
>> >>> > otherwise I would like you to get a gcore while it's int that state,
>> >>> > open it in gdb and produce a "thread apply all bt" trace in text
>> >>> > format and attach it to the JIRA.
>> >>> >
>> >>> >
>> >>> >
>> >>> >
>> >>> > On Tue, Oct 5, 2010 at 9:55 AM, Fran?ois Delawarde
>> >>> > <fdelawarde at wirelessmundi.com> wrote:
>> >>> >> I wouldn't jump to conclusions that fast, it's just a bug with current
>> >>> >> git head, probably only happening in very specific cases.
>> >>> >>
>> >>> >> If I learned anything from hollywood movies, it would be that bugs or
>> >>> >> bad people always end up being tracked down and destroyed.
>> >>> >>
>> >>> >> Fran?ois.
>> >>> >>
>> >>> >>
>> >>> >> On Tue, 2010-10-05 at 20:35 +0600, Nyamul Hassan wrote:
>> >>> >>> We learned from another thread about how Sangoma regularly tests 32 E1
>> >>> >>> setups in the lab, and were preparing a concept paper to show a
>> >>> >>> telecom that traditional vendors can be replaced using open source
>> >>> >>> even for large scale implementations. Does this mean we should rethink
>> >>> >>> our strategy?
>> >>> >>>
>> >>> >>> Regards
>> >>> >>> HASSAN
>> >>> >>>
>> >>> >>>
>> >>> >>> On 2010-10-05, Moises Silva <moises.silva at gmail.com> wrote:
>> >>> >>> > On Tue, Oct 5, 2010 at 5:45 AM, Fran?ois Delawarde
>> >>> >>> > <fdelawarde at wirelessmundi.com> wrote:
>> >>> >>> >> I maintain "only" 100 sessions at all times, with a duration of 20-30s,
>> >>> >>> >> and no more than 10cps, so it is NOT a stress test, come on!
>> >>> >>> >>
>> >>> >>> >> I mean this would not even load an Asterisk server, and loading the
>> >>> >>> >> server was not an objective, just to maintain low-medium load over a
>> >>> >>> >> period of time. It worked very well during 5h.
>> >>> >>> >>
>> >>> >>> >> Why would I need to ulimit the crap out of my server then? :-)
>> >>> >>> >
>> >>> >>> > Don't feel alone, you're not the only one.
>> >>> >>> >
>> >>> >>> > We're getting the same pair of messages. We know what it means, we
>> >>> >>> > have just not found the cause of it. In our case happens quite fast
>> >>> >>> > (about an hour IIRC) when doing stress testing of about 900 calls at a
>> >>> >>> > time lasting 10 seconds each at a rate of about 40cps.
>> >>> >>> >
>> >>> >>> > Moises Silva
>> >>> >>> > Senior Software Engineer
>> >>> >>> > Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
>> >>> >>> > L3R 9R6 Canada
>> >>> >>> > t. 1 905 474 1990 x128 | e. moy at sangoma.com
>> >>> >>> >
>> >>> >>> > _______________________________________________
>> >>> >>> > FreeSWITCH-users mailing list
>> >>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >>> > http://www.freeswitch.org
>> >>> >>> >
>> >>> >>>
>> >>> >>
>> >>> >>
>> >>> >> _______________________________________________
>> >>> >> FreeSWITCH-users mailing list
>> >>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >> 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
>> >>> >
>> >>>
>> >>>
>> >>>
>> >>> --
>> >>> 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
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> 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
>> >
>>
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From anthony.minessale at gmail.com  Thu Oct  7 09:03:27 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 7 Oct 2010 11:03:27 -0500
Subject: [Freeswitch-users] RPID in 183 Session progress
In-Reply-To: <AANLkTik-as7poMiPivJkzRm4v0YcR1Y69WOUDoz6yKJz@mail.gmail.com>
References: <AANLkTik-as7poMiPivJkzRm4v0YcR1Y69WOUDoz6yKJz@mail.gmail.com>
Message-ID: <AANLkTimbZpmhqbMJpkkgxU+dkyapcMUsAzyBu08-r4Mt@mail.gmail.com>

sip_cid_in_1xx=false for the channel var version
profile param pass-callee-id=false for the profile wide version.


On Thu, Oct 7, 2010 at 10:52 AM, Piotr Sobolewski <sobolewski at gmail.com> wrote:
> Hi
>
> I have strange situation where FreeSWITCH ads Remote-party-ID to 183
> session progress received from upstream GW.
> Diagram below demonstrates this issue.
>
> UA1 ------> PROXY ------> FS ?(INVITE ruri=777 with RPID=123)
> UA1 <------ PROXY <------- FS ?(100 Trying)
> FS ----> GW ? (INVITE with RPID=123)
> UA1 <----( 183 session with RPID=777)---- FS <-----(183 session
> without RPID)------ GW
>
> I've tried both
> ?<action application="set" data="sip_cid_type=none"/>
> and
> ?<action application="export" data="sip_cid_type=none"/>
> in the dialplan but without success (still have RPID in the 183)
>
> Is it possible to disable adding of RPID to ?SIP response 183 Session Progress?
>
>
> --
> Piotr Sobolewski
> sobolewski at gmail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From djbinter at gmail.com  Thu Oct  7 09:09:03 2010
From: djbinter at gmail.com (DJB International)
Date: Thu, 7 Oct 2010 09:09:03 -0700
Subject: [Freeswitch-users] RPID in 183 Session progress
In-Reply-To: <AANLkTik-as7poMiPivJkzRm4v0YcR1Y69WOUDoz6yKJz@mail.gmail.com>
References: <AANLkTik-as7poMiPivJkzRm4v0YcR1Y69WOUDoz6yKJz@mail.gmail.com>
Message-ID: <AANLkTi=efMNWJoTh3XsTQH9pLdnj7oE-kesQqvTRkbp3@mail.gmail.com>

Set pass-callee-id=false in your profile

-djbinter


On Thu, Oct 7, 2010 at 8:52 AM, Piotr Sobolewski <sobolewski at gmail.com>wrote:

> Hi
>
> I have strange situation where FreeSWITCH ads Remote-party-ID to 183
> session progress received from upstream GW.
> Diagram below demonstrates this issue.
>
> UA1 ------> PROXY ------> FS  (INVITE ruri=777 with RPID=123)
> UA1 <------ PROXY <------- FS  (100 Trying)
> FS ----> GW   (INVITE with RPID=123)
> UA1 <----( 183 session with RPID=777)---- FS <-----(183 session
> without RPID)------ GW
>
> I've tried both
>  <action application="set" data="sip_cid_type=none"/>
> and
>  <action application="export" data="sip_cid_type=none"/>
> in the dialplan but without success (still have RPID in the 183)
>
> Is it possible to disable adding of RPID to  SIP response 183 Session
> Progress?
>
>
> --
> Piotr Sobolewski
> sobolewski at gmail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sobolewski at gmail.com  Thu Oct  7 09:15:31 2010
From: sobolewski at gmail.com (Piotr Sobolewski)
Date: Thu, 7 Oct 2010 18:15:31 +0200
Subject: [Freeswitch-users] RPID in 183 Session progress
In-Reply-To: <AANLkTimbZpmhqbMJpkkgxU+dkyapcMUsAzyBu08-r4Mt@mail.gmail.com>
References: <AANLkTik-as7poMiPivJkzRm4v0YcR1Y69WOUDoz6yKJz@mail.gmail.com>
	<AANLkTimbZpmhqbMJpkkgxU+dkyapcMUsAzyBu08-r4Mt@mail.gmail.com>
Message-ID: <AANLkTimOHE-4rOCrEMWUnydKtjs8vBBKq+EKPrYYi1th@mail.gmail.com>

Thank you very much, it worked.


On Thu, Oct 7, 2010 at 6:03 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> sip_cid_in_1xx=false for the channel var version
> profile param pass-callee-id=false for the profile wide version.
>
>
> On Thu, Oct 7, 2010 at 10:52 AM, Piotr Sobolewski <sobolewski at gmail.com> wrote:
>> Hi
>>
>> I have strange situation where FreeSWITCH ads Remote-party-ID to 183
>> session progress received from upstream GW.
>> Diagram below demonstrates this issue.
>>
>> UA1 ------> PROXY ------> FS ?(INVITE ruri=777 with RPID=123)
>> UA1 <------ PROXY <------- FS ?(100 Trying)
>> FS ----> GW ? (INVITE with RPID=123)
>> UA1 <----( 183 session with RPID=777)---- FS <-----(183 session
>> without RPID)------ GW
>>
>> I've tried both
>> ?<action application="set" data="sip_cid_type=none"/>
>> and
>> ?<action application="export" data="sip_cid_type=none"/>
>> in the dialplan but without success (still have RPID in the 183)
>>
>> Is it possible to disable adding of RPID to ?SIP response 183 Session Progress?
>>
>>
>> --
>> Piotr Sobolewski
>> sobolewski at gmail.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Piotr Sobolewski
sobolewski at gmail.com


From sobolewski at gmail.com  Thu Oct  7 09:15:23 2010
From: sobolewski at gmail.com (Piotr Sobolewski)
Date: Thu, 7 Oct 2010 18:15:23 +0200
Subject: [Freeswitch-users] RPID in 183 Session progress
In-Reply-To: <AANLkTimbZpmhqbMJpkkgxU+dkyapcMUsAzyBu08-r4Mt@mail.gmail.com>
References: <AANLkTik-as7poMiPivJkzRm4v0YcR1Y69WOUDoz6yKJz@mail.gmail.com>
	<AANLkTimbZpmhqbMJpkkgxU+dkyapcMUsAzyBu08-r4Mt@mail.gmail.com>
Message-ID: <AANLkTi=_tDgtjGoLXfs4wkXkxefF-pKFDBXrB9g0-dFq@mail.gmail.com>

Thank you very much, it worked.


On Thu, Oct 7, 2010 at 6:03 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> sip_cid_in_1xx=false for the channel var version
> profile param pass-callee-id=false for the profile wide version.
>
>
> On Thu, Oct 7, 2010 at 10:52 AM, Piotr Sobolewski <sobolewski at gmail.com> wrote:
>> Hi
>>
>> I have strange situation where FreeSWITCH ads Remote-party-ID to 183
>> session progress received from upstream GW.
>> Diagram below demonstrates this issue.
>>
>> UA1 ------> PROXY ------> FS ?(INVITE ruri=777 with RPID=123)
>> UA1 <------ PROXY <------- FS ?(100 Trying)
>> FS ----> GW ? (INVITE with RPID=123)
>> UA1 <----( 183 session with RPID=777)---- FS <-----(183 session
>> without RPID)------ GW
>>
>> I've tried both
>> ?<action application="set" data="sip_cid_type=none"/>
>> and
>> ?<action application="export" data="sip_cid_type=none"/>
>> in the dialplan but without success (still have RPID in the 183)
>>
>> Is it possible to disable adding of RPID to ?SIP response 183 Session Progress?
>>
>>
>> --
>> Piotr Sobolewski
>> sobolewski at gmail.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Piotr Sobolewski
sobolewski at gmail.com


From adminjew at gmail.com  Thu Oct  7 10:15:19 2010
From: adminjew at gmail.com (Yitzchok)
Date: Thu, 7 Oct 2010 13:15:19 -0400
Subject: [Freeswitch-users] mod_managed on linux centos
In-Reply-To: <AANLkTin0HvwpYfOCoH3upSGvA+UFiirjRoBic72yovqg@mail.gmail.com>
References: <AANLkTi=xt9CZBPZLnBX56e2Rjdc_FRf_=er6Yy7w7ZuN@mail.gmail.com>
	<AANLkTi=S=tpzczDM8WSm6N8AeTkN3XNTPda6048sRoJ9@mail.gmail.com>
	<4C801B6E.4050103@puzzled.xs4all.nl>
	<AANLkTimzjbLHbObgsQKqX3t-hfaZWr95UGMi17YxZFHS@mail.gmail.com>
	<4C838C3B.3070202@puzzled.xs4all.nl>
	<AANLkTinP+UfvkM0dmBycy5GqivAXNHGF9RnTBNvV0gV4@mail.gmail.com>
	<AANLkTik1CY5-6DUvsLg512bGRr66xLF5H8Gi7WYRenHx@mail.gmail.com>
	<1286205151753-5599563.post@n2.nabble.com>
	<AANLkTinKemVBa5HNSrJk6EOfWYM5wd7CyL4L-JX-BawQ@mail.gmail.com>
	<AANLkTinF2yvyA6ZerxjFoOi+ti8DAGHAyEYfBFrRGiR1@mail.gmail.com>
	<1286377066483-5607396.post@n2.nabble.com>
	<AANLkTin0HvwpYfOCoH3upSGvA+UFiirjRoBic72yovqg@mail.gmail.com>
Message-ID: <AANLkTikxXcmebpQStBeCceFG_-_QYaum16FZq7WmybSG@mail.gmail.com>

It looks like these errors don't happen in Mono *2.8* when I use the
original code. (leaving mono_thread_detach(mono_thread_current()); in there)

Though now I see that *sometimes *when I update a dll file it doesn't work
and I have to restart freeswitch and then it works.

Yitzchok


On Wed, Oct 6, 2010 at 11:40 AM, Yitzchok <adminjew at gmail.com> wrote:

> I tried that and it seems to work.
>
>
> ---diff from git head---
>
>
> @@ -385,9 +385,7 @@ SWITCH_STANDARD_API(managedrun_api_function)
>    } else {
>   stream->write_function(stream, "-ERR ExecuteBackground returned false
> (unknown module or exception?).\n");
>   }
> -#ifndef _MANAGED
> - mono_thread_detach(mono_thread_current());
> -#endif
> +
>   return SWITCH_STATUS_SUCCESS;
>  }
>
> @@ -403,9 +401,7 @@ SWITCH_STANDARD_API(managed_api_function)
>   if (!(executeDelegate(cmd, stream, stream->param_event))) {
>   switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Execute failed
> for %s (unknown module or exception).\n", cmd);
>   }
> -#ifndef _MANAGED
> - mono_thread_detach(mono_thread_current());
> -#endif
> +
>   return SWITCH_STATUS_SUCCESS;
>  }
>
> @@ -421,9 +417,7 @@ SWITCH_STANDARD_APP(managed_app_function)
>   if (!(runDelegate(data, session))) {
>   switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Application run
> failed for %s (unknown module or exception).\n", data);
>   }
> -#ifndef _MANAGED
> - mono_thread_detach(mono_thread_current());
> -#endif
> +
>  }
>
>  SWITCH_STANDARD_API(managedreload_api_function)
> @@ -438,9 +432,7 @@ SWITCH_STANDARD_API(managedreload_api_function)
>   if (!(reloadDelegate(cmd))) {
>   switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Execute failed
> for %s (unknown module or exception).\n", cmd);
>   }
> -#ifndef _MANAGED
> - mono_thread_detach(mono_thread_current());
> -#endif
> +
>   return SWITCH_STATUS_SUCCESS;
>  }
>
>
>
> -----
>
> Yitzchok
>
>
>
> On Wed, Oct 6, 2010 at 10:57 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
>
>>
>> Not sure how this makes sense yet. Yitzchok would you try this with only
>> the
>> mono_thread_detach calls removed leaving the mono_thread_attach in place
>> and
>> see what happens.
>>
>> Thanks
>> Jeff
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/mod-managed-on-linux-centos-tp5485480p5607396.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From msc at freeswitch.org  Thu Oct  7 10:42:43 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 7 Oct 2010 10:42:43 -0700
Subject: [Freeswitch-users] Radius AAA
In-Reply-To: <4CADC2E4.90608@gmail.com>
References: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>
	<AANLkTi=GiQaO-48BP=T8c-A+09sJMKzQH7BX=ewBdB8R@mail.gmail.com>
	<AANLkTikw8LGDqQki16D30b8JL0eJid2UOTs-bEaZSc2k@mail.gmail.com>
	<4CADC2E4.90608@gmail.com>
Message-ID: <AANLkTin03N2YNMYP0=r6L_3AjCGx-ip-5NCHDd6ObDK4@mail.gmail.com>

2 gold stars for Nazim!
-MC

On Thu, Oct 7, 2010 at 5:53 AM, Nazim Aghabayov
<nazim.aghabayov at gmail.com>wrote:

>  Hello Tihomir,
>
> I've just updated the Mod_rad_auth wiki with installation instructions.
> Thanks a lot for the mod. I use it in production and it works just great!
>
> Regards,
> Nazim
>
> On 10/07/2010 01:48 PM, Tihomir Culjaga wrote:
> > On Thu, Oct 7, 2010 at 1:15 AM, Michael Collins<msc at freeswitch.org>
>  wrote:
> >
> >> I'm afraid no such comprehensive documentation exists. Submissions to
> our
> >> wiki are most welcomed. This is all we have at the moment:
> >>
> >> http://wiki.freeswitch.org/wiki/Mod_rad_auth
> >>
> >> Be sure to install freeradius2 (server) and freeradius-client before
> trying
> >> to install mod_auth_rad.
> >>
> >> -MC
> >>
> >>
> >>
> > correct,
> >
> > Im the author of the module and im going to provide the documentation and
> > how-to.
> >
> >
> > in brief, this module does radius auth (not accounting). It is based on
> > freeradius-client library and as such this is the only dependency.
> >
> > you can specify your own list of VSAs to be included in the packet along
> > with the standard ones that are being used.
> >
> >
> >           name:       just a description
> >           value:      direct input or variable
> >           pec:        vendor ID (0 for default, 9 for cisco...)
> >           expr:       1 for channel variable, 2 for direct input (string)
> >           direction:  in for radius-request, out for radius-response
> >
> >
> > Im not going to describe it here... its better i do it on the wiki
> itself...
> >
> >
> >
> >
> > T.
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Thu Oct  7 11:00:46 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 7 Oct 2010 20:00:46 +0200
Subject: [Freeswitch-users] Radius AAA
In-Reply-To: <4CADC2E4.90608@gmail.com>
References: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>
	<AANLkTi=GiQaO-48BP=T8c-A+09sJMKzQH7BX=ewBdB8R@mail.gmail.com>
	<AANLkTikw8LGDqQki16D30b8JL0eJid2UOTs-bEaZSc2k@mail.gmail.com>
	<4CADC2E4.90608@gmail.com>
Message-ID: <AANLkTik5c9yEKhvzsKNJJNnrvKwbAuVLoOk6jRgonbGX@mail.gmail.com>

On Thu, Oct 7, 2010 at 2:53 PM, Nazim Aghabayov
<nazim.aghabayov at gmail.com>wrote:

>  Hello Tihomir,
>
> I've just updated the Mod_rad_auth wiki with installation instructions.
> Thanks a lot for the mod. I use it in production and it works just great!
>
>

wow, thats great.... you guys just saved me an hour or two of documenting...
Perhaps, we can add an example of a dictionary file and how to add your own
Vendor ID definitions

Also, the radius dictionary you point need to have all vendor IDs specified
... i found out that sometimes it has issues with $INCLUDEs.

<param name="dictionary" value="/usr/local/etc/radiusclient/dictionary.all"/>


This is for auth... I have modified mod_xml_cdr as well and added radius and
odbc interfaces as well. Now im able to store CDRs on 4 interfacs
simultaneously.
regarding radius accounting, the logic for VSAs is the same as for
mod_rad_auth.


I can provide diffs or patches if you like ... and i can give it here to be
stored in git.

Interested ?

T.
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From tculjaga at gmail.com  Thu Oct  7 11:03:35 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 7 Oct 2010 20:03:35 +0200
Subject: [Freeswitch-users] mod_h323 - worst audio quality compare to
 other h323
In-Reply-To: <alpine.DEB.2.00.1010052002410.18429@icf.org.ru>
References: <AANLkTim0JR1Cghq_7m0E686DtRwEfBDShFHieuGTRgcb@mail.gmail.com>
	<alpine.DEB.2.00.1010052002410.18429@icf.org.ru>
Message-ID: <AANLkTi=o-sh-HaGuP_pUcQ7Ste7ss7oa-=QXo7Wkdr_O@mail.gmail.com>

> Stm>
> Stm>Highest priority, for mod_h323 Audio problem:
> Stm>===================================
>
> enable late negotiation and put fs in transconding mode.
>
>
Shamun, did you try it?
What is the result ?
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From fs-list at communicatefreely.net  Thu Oct  7 11:48:23 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Thu, 07 Oct 2010 14:48:23 -0400
Subject: [Freeswitch-users] Dead lua scripts
Message-ID: <4CAE15F7.9090807@communicatefreely.net>

Hello,

I was building some lua apps, and clearly went wrong somewhere.

I now have several channels in the hangup state, all from the extension I was testing, to the script
I was testing.  There are lots of errors in the script at this point, so I'm sure it's something I did.

However, these channels sent the CPU to 120% (according to top), and I'm not at a load average of 4.
 Fortunately, Freeswitch handles this rather gracefully and call quality seems to be okay in spite
of it.

Is there a way to kill these threads without killing the other calls?  I tried using uuid_kill, but
it came back as no such channel.  I can see the channels in "show channels", but the uuids listed
don't exist according to uuid_exists.

Any suggestions?

Thanks!

-Tim


From nazim.aghabayov at gmail.com  Thu Oct  7 12:11:26 2010
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Fri, 08 Oct 2010 00:11:26 +0500
Subject: [Freeswitch-users] Radius AAA
In-Reply-To: <AANLkTik5c9yEKhvzsKNJJNnrvKwbAuVLoOk6jRgonbGX@mail.gmail.com>
References: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>	<AANLkTi=GiQaO-48BP=T8c-A+09sJMKzQH7BX=ewBdB8R@mail.gmail.com>	<AANLkTikw8LGDqQki16D30b8JL0eJid2UOTs-bEaZSc2k@mail.gmail.com>	<4CADC2E4.90608@gmail.com>
	<AANLkTik5c9yEKhvzsKNJJNnrvKwbAuVLoOk6jRgonbGX@mail.gmail.com>
Message-ID: <4CAE1B5E.6010203@gmail.com>

  Great new features! Of course I'm interested.
I've even borrowed some of your code from mod_xml_cdr to implement a simple
radius accounting as a plugin for Xmlcdrd: 
http://wiki.freeswitch.org/wiki/Xmlcdrd
Well, shame on me. It's not documented yet. Hoping I'll document it soon )

Nazim

On 10/07/2010 11:00 PM, Tihomir Culjaga wrote:
>
> wow, thats great.... you guys just saved me an hour or two of documenting...
> Perhaps, we can add an example of a dictionary file and how to add your own
> Vendor ID definitions
>
> Also, the radius dictionary you point need to have all vendor IDs specified
> ... i found out that sometimes it has issues with $INCLUDEs.
>
> <param name="dictionary" value="/usr/local/etc/radiusclient/dictionary.all"/>
>
>
> This is for auth... I have modified mod_xml_cdr as well and added radius and
> odbc interfaces as well. Now im able to store CDRs on 4 interfacs
> simultaneously.
> regarding radius accounting, the logic for VSAs is the same as for
> mod_rad_auth.
>
>
> I can provide diffs or patches if you like ... and i can give it here to be
> stored in git.
>
> Interested ?
>
> T.


From Peter.Hinman at ParcelPool.com  Thu Oct  7 12:25:14 2010
From: Peter.Hinman at ParcelPool.com (Peter Hinman)
Date: Thu, 07 Oct 2010 13:25:14 -0600
Subject: [Freeswitch-users] ODBC and MSSQL
In-Reply-To: <1286459189732-5611425.post@n2.nabble.com>
References: <4CAB9F5A.2090503@ParcelPool.com>	<1286331931285-5605569.post@n2.nabble.com>	<4CACE7C5.6060208@ParcelPool.com>
	<1286459189732-5611425.post@n2.nabble.com>
Message-ID: <4CAE1E9A.10708@ParcelPool.com>

  I'm certainly interested.  It may take me a couple of days to get to 
it.  I'll let you know how it goes for me.

Peter

On 10/7/2010 7:46 AM, Jeff Lenk wrote:
> Hi Peter,
>
> If your so inclined please test the patch here:
> http://jira.freeswitch.org/browse/FS-2050
>
> and please let me know how it works for you.
>
> After patching and building
> remove commented ->  <!--  -->
>
>
> -Jeff



From tculjaga at gmail.com  Thu Oct  7 12:24:58 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 7 Oct 2010 21:24:58 +0200
Subject: [Freeswitch-users] Radius AAA
In-Reply-To: <4CAE1B5E.6010203@gmail.com>
References: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>
	<AANLkTi=GiQaO-48BP=T8c-A+09sJMKzQH7BX=ewBdB8R@mail.gmail.com>
	<AANLkTikw8LGDqQki16D30b8JL0eJid2UOTs-bEaZSc2k@mail.gmail.com>
	<4CADC2E4.90608@gmail.com>
	<AANLkTik5c9yEKhvzsKNJJNnrvKwbAuVLoOk6jRgonbGX@mail.gmail.com>
	<4CAE1B5E.6010203@gmail.com>
Message-ID: <AANLkTim9LP=m8UdVYrupxRQTU=_HqV79QNSLyPrB+qUY@mail.gmail.com>

On Thu, Oct 7, 2010 at 9:11 PM, Nazim Aghabayov
<nazim.aghabayov at gmail.com>wrote:

>  Great new features! Of course I'm interested.
> I've even borrowed some of your code from mod_xml_cdr to implement a simple
> radius accounting as a plugin for Xmlcdrd:
> http://wiki.freeswitch.org/wiki/Xmlcdrd
> Well, shame on me. It's not documented yet. Hoping I'll document it soon )
>
>
i don't remember if i actually uploaded mod_xml_cdr patches ... hmmm

anyhow i use odbc to send CDRS to a DB for statistics purpose (calculate
ASR/ACD/ROUTING) and radius interface for billing purpose as this hooks to
an external billing.
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From tculjaga at gmail.com  Thu Oct  7 12:30:58 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 7 Oct 2010 21:30:58 +0200
Subject: [Freeswitch-users] ClueCon video
In-Reply-To: <AANLkTi=EFO9qoKgATO0P6gwKeAB1WVC7CC4Op=vBd+gu@mail.gmail.com>
References: <AANLkTi=1KS+h7FAB8TrYA6BMtpOuEw8eBxP2U4e1YVYC@mail.gmail.com>
	<AANLkTi=EFO9qoKgATO0P6gwKeAB1WVC7CC4Op=vBd+gu@mail.gmail.com>
Message-ID: <AANLkTinbNuSJYeiJh+oNRYfXKo8kRfnDXErfycj7fPw1@mail.gmail.com>

On Thu, Oct 7, 2010 at 12:18 AM, Ognjen Seslija <oseslija at gmail.com> wrote:

> Who was the role model for the clown?
>
> On Wed, Oct 6, 2010 at 6:06 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> http://www.freeswitch.org/node/287
>>
>>

... actually there is a bug in the video :)))
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From nazim.aghabayov at gmail.com  Thu Oct  7 12:39:39 2010
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Fri, 08 Oct 2010 00:39:39 +0500
Subject: [Freeswitch-users] Radius AAA
In-Reply-To: <AANLkTim9LP=m8UdVYrupxRQTU=_HqV79QNSLyPrB+qUY@mail.gmail.com>
References: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>	<AANLkTi=GiQaO-48BP=T8c-A+09sJMKzQH7BX=ewBdB8R@mail.gmail.com>	<AANLkTikw8LGDqQki16D30b8JL0eJid2UOTs-bEaZSc2k@mail.gmail.com>	<4CADC2E4.90608@gmail.com>	<AANLkTik5c9yEKhvzsKNJJNnrvKwbAuVLoOk6jRgonbGX@mail.gmail.com>	<4CAE1B5E.6010203@gmail.com>
	<AANLkTim9LP=m8UdVYrupxRQTU=_HqV79QNSLyPrB+qUY@mail.gmail.com>
Message-ID: <4CAE21FB.3050700@gmail.com>

  Pardon me ) I was mean to say "imported some of your code from 
mod_rad_auth".
Thank you once again for the module, it's extremely configurable and 
easy to use.

Nazim.

On 10/08/2010 12:24 AM, Tihomir Culjaga wrote:
> i don't remember if i actually uploaded mod_xml_cdr patches ... hmmm
>
> anyhow i use odbc to send CDRS to a DB for statistics purpose (calculate
> ASR/ACD/ROUTING) and radius interface for billing purpose as this hooks to
> an external billing.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From peter.schrock at gmail.com  Thu Oct  7 12:40:58 2010
From: peter.schrock at gmail.com (Peter Schrock)
Date: Thu, 7 Oct 2010 12:40:58 -0700
Subject: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57E0E68E2E@cooper>
References: <AANLkTingeYfPT_b6dG3bKQOVjxYsmBTKhDuLw_3BKGac@mail.gmail.com>
	<A499E7B3-9727-4822-B2AB-8FDDCA3213FA@avgs.ca>
	<AANLkTinOUhDgzCB2rTL20_RzGq3=uS7zOKE064DOR0v6@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57E0E68E2E@cooper>
Message-ID: <AANLkTinHyjVKiidD4d2X3d3hKDWDcRpokhVG8pU3dcUt@mail.gmail.com>

John Covici,

To make sure I understand correctly, do you me the copy found in:
/usr/local/freeswitch/libs/esl
or somewhere else?
Also, is there a wiki that discusses how to make those parameters agreeable?

Peter Olsson,
Thanks for clarifying that. I guess my question now would be (cause I am
trying to use the book as much as possible to help me understand FS) is why
would the book be insistent on using fs_cli and how many people actually use
it. Is there a wiki discussing how to manage the mod_event_socket so that
fs_cli will function the way it should instead popping up errors like it has
been.

Also, in general, if one were to set up FS in background mode, how do you
stop it from being in background mode?

PeterS

On Wed, Oct 6, 2010 at 11:14 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  fs_cli is a tool to connect to a running instance of FS, on the same
> computer, or another computer. It communicates with FS using the
> mod_event_socket loaded within FS. The server socket in FS listens for
> requests on port 8021 by default, this can be changed in
> event_socket.conf.xml
>
>
>
> /Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Peter Schrock
> *Skickat:* den 7 oktober 2010 07:55
> *Till:* FreeSWITCH Users Help
> *?mne:* Re: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4
>
>
>
> In answer to your first question, yes. I guess my next question would be,
> if I don't need fs_cli, then what is it for? I guess I am assuming to access
> FS when it is running in background. What would be the difference between
> running it in the foreground vs background?
>
> Also, I ran FS without -nc and it showed that mod_event_socket loaded
> successfully. So why won't it with fs_cli?
>
>
>
> PeterS
>
> On Wed, Oct 6, 2010 at 5:29 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>
> Hi,
>
>
>
> Did you start FreeSWITCH before? fs_cli is only used to connect to the
> console of a running FreeSWITCH instance.
>
>
>
> If you did, try starting it without the -nc switch, it should stay in the
> foreground and tell you whats wrong, if you have a console you can also
> check that mod_event_socket is properly loaded (load mod_event_socket)
>
>
>
> Mathieu Rene
>
> Avant-Garde Solutions Inc
>
> Office: + 1 (514) 664-1044 x100
>
> Cell: +1 (514) 664-1044 x200
>
> mrene at avgs.ca
>
>
>
>
>
>
>
>
>
> On 2010-10-06, at 8:23 PM, Peter Schrock wrote:
>
>
>
>  Okay, so I managed to get FS working the other day and I even managed to
> test a call and test my voicemail. All seemed to be working smoothly until,
> because of the rain here, my power went out and I had to reboot my computer.
> I logged in through the terminal, set up FS in background went to fs_cliand I get this error message:
>
> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
> Error]
>
> I managed to figure out that in the file "fs_cli.c" on line 1206 you find
> the code for displaying this error message. The problem is that I don't know
> why this error message is occurring. Does anyone have any helpful hints as
> to what I should look at to resolve this problem?
>
> I even tried going to the git tree and make current, but that gave me
> problems that forced me to turn off mod_spandsp and mod_skyopen in the
> modules.conf, which I had running earlier. Any thoughts?
>
>
>
> PeterS
>
>
>
> PS
>
> I am not sure if this is of any help, but in addition to the error line
> above, it also posted this info:
>
> Usage: fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x
> command] [profile]
>
>
>
>   -?,-h --help                    Usage Information
>
>   -H, --host=hostname             Host to connect
>
>   -P, --port=port                 Port to connect (1 - 65535)
>
>   -u, --user=user at domain          user at domain
>
>   -p, --password=password         Password
>
>   -x, --execute=command           Execute Command and Exit
>
>   -l, --loglevel=command          Log Level
>
>   -q, --quiet                     Disable logging
>
>   -r, --retry                     Retry connection on failure
>
>   -R, --reconnect                 Reconnect if disconnected
>
>   -d, --debug=level               Debug Level (0 - 7)
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> !DSPAM:4cad626f32932643394250!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From moises.silva at gmail.com  Thu Oct  7 12:51:31 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 7 Oct 2010 13:51:31 -0600
Subject: [Freeswitch-users] outbound calling error: multiple PRIs with
 Sangoma + FS
In-Reply-To: <AANLkTimn7P64Exf9JTwdLZB8Vam8wBLuvrLS5MY+xAwT@mail.gmail.com>
References: <AANLkTimn7P64Exf9JTwdLZB8Vam8wBLuvrLS5MY+xAwT@mail.gmail.com>
Message-ID: <AANLkTin01O6WPqXfXN+aiaPcOd+Ww1HiFwzcbUEeMz2Z@mail.gmail.com>

On Thu, Oct 7, 2010 at 6:52 AM, Neil Patel <neilp at cs.stanford.edu> wrote:
> ...and here is the output from FS CLI. Note that the PRI line connected to
> port 2 (g2) works great; it is g1 that is producing the NO_ROUTE_DESTINATION
> error:
>
> freeswitch at otalo> originate openzap/smg_prid/a/958655XXXX at g1 &echo
> 2010-10-07 18:08:25.273814 [WARNING] ozmod_sangoma_boost.c:344 TX EVENT:
> CALL_START:(80) [w1g1] CSid=5 Seq=41 Cn=[N/A] Cd=[958655XXXX] Ci=[N/A]
> Rdnis=[]
> 2010-10-07 18:08:25.306941 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT
> (N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=5 Seq=34
> 2010-10-07 18:08:25.307953 [NOTICE] switch_channel.c:675 New Channel
> OpenZAP/1:1/958655XXXX at g1 [c4305297-3a6e-4bd8-a25c-71e96da6b77d]
> 2010-10-07 18:08:25.411216 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT
> (N): CALL_STOPPED:(85) [w1g1] Rc=3 CSid=5 Seq=35
> 2010-10-07 18:08:25.411216 [NOTICE] mod_openzap.c:1935 Hangup
> OpenZAP/1:1/958655XXXX at g1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
>
> -ERR NO_ROUTE_DESTINATION
>
> 2010-10-07 18:08:25.412218 [NOTICE] switch_core_session.c:1188 Session 16
> (OpenZAP/1:1/958655XXXX at g1) Ended
> 2010-10-07 18:08:25.412218 [NOTICE] switch_core_session.c:1190 Close Channel
> OpenZAP/1:1/958655XXXX at g1 [CS_DESTROY]
> 2010-10-07 18:08:25.441373 [WARNING] sangoma_boost_client.c:221 TX EVENT
> (N): CALL_STOPPED_ACK:(86) [w1g1] Rc=0 CSid=0 Seq=42
> freeswitch at otalo> originate openzap/smg_prid/a/958655XXXX at g2 &echo
> 2010-10-07 18:08:30.199197 [WARNING] ozmod_sangoma_boost.c:344 TX EVENT:
> CALL_START:(80) [w1g1] CSid=6 Seq=43 Cn=[N/A] Cd=[958655XXXX] Ci=[N/A]
> Rdnis=[]
> 2010-10-07 18:08:30.263078 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT
> (N): CALL_START_ACK:(81) [w2g1] Rc=0 CSid=6 Seq=36
> 2010-10-07 18:08:30.264081 [NOTICE] switch_channel.c:675 New Channel
> OpenZAP/1:31/958655XXXX at g2 [fe1dad03-1086-4d56-bc55-62563e170a78]
> 2010-10-07 18:08:30.493273 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT:
> CALL PROGRESS:(50) [w2g1] CSid=6 Seq=37 Cn=[N/A] Cd=[N/A] Ci=[N/A] Rdnis=[]
> 2010-10-07 18:08:30.493273 [NOTICE] mod_openzap.c:1962 Pre-Answer
> OpenZAP/1:31/958655XXXX at g2!
>
> +OK fe1dad03-1086-4d56-bc55-62563e170a78
>
> I don't notice much going on in /var/log/sangoma_mgd.log or
> /var/log/messages, but then again I'm not sure what to look for. Any
> guidance would be greatly appreciated!
> Neil
>

Hi Neil,

Any reason you are using deprecated software? FreeTDM replaced openzap
and libsng_isdn replaced boost.

See:

http://wiki.freeswitch.org/wiki/FreeTDM
http://wiki.sangoma.com/FreeTDM-Sangoma-ISDN-Library

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From anthony.minessale at gmail.com  Thu Oct  7 14:32:36 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 7 Oct 2010 16:32:36 -0500
Subject: [Freeswitch-users] Dead lua scripts
In-Reply-To: <4CAE15F7.9090807@communicatefreely.net>
References: <4CAE15F7.9090807@communicatefreely.net>
Message-ID: <AANLkTinKznsWB6Zzf3cexnqHjesgQ_gsK++KkKD_h+U0@mail.gmail.com>

look for endless loops in your code that are not using session:ready

On Thu, Oct 7, 2010 at 1:48 PM, Tim St. Pierre
<fs-list at communicatefreely.net> wrote:
> Hello,
>
> I was building some lua apps, and clearly went wrong somewhere.
>
> I now have several channels in the hangup state, all from the extension I was testing, to the script
> I was testing. ?There are lots of errors in the script at this point, so I'm sure it's something I did.
>
> However, these channels sent the CPU to 120% (according to top), and I'm not at a load average of 4.
> ?Fortunately, Freeswitch handles this rather gracefully and call quality seems to be okay in spite
> of it.
>
> Is there a way to kill these threads without killing the other calls? ?I tried using uuid_kill, but
> it came back as no such channel. ?I can see the channels in "show channels", but the uuids listed
> don't exist according to uuid_exists.
>
> Any suggestions?
>
> Thanks!
>
> -Tim
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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
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FreeSWITCH Developer Conference
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googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From covici at ccs.covici.com  Thu Oct  7 14:40:41 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 07 Oct 2010 17:40:41 -0400
Subject: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4
In-Reply-To: <AANLkTinHyjVKiidD4d2X3d3hKDWDcRpokhVG8pU3dcUt@mail.gmail.com>
References: <AANLkTingeYfPT_b6dG3bKQOVjxYsmBTKhDuLw_3BKGac@mail.gmail.com>
	<A499E7B3-9727-4822-B2AB-8FDDCA3213FA@avgs.ca>
	<AANLkTinOUhDgzCB2rTL20_RzGq3=uS7zOKE064DOR0v6@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57E0E68E2E@cooper>
	<AANLkTinHyjVKiidD4d2X3d3hKDWDcRpokhVG8pU3dcUt@mail.gmail.com>
Message-ID: <5211.1286487641@ccs.covici.com>

Yep, that is the file and you can look in the configs for mod_esl  and
make sure the port, password, etc. match.

Peter Schrock <peter.schrock at gmail.com> wrote:

> John Covici,
> 
> To make sure I understand correctly, do you me the copy found in:
> /usr/local/freeswitch/libs/esl
> or somewhere else?
> Also, is there a wiki that discusses how to make those parameters agreeable?
> 
> Peter Olsson,
> Thanks for clarifying that. I guess my question now would be (cause I am
> trying to use the book as much as possible to help me understand FS) is why
> would the book be insistent on using fs_cli and how many people actually use
> it. Is there a wiki discussing how to manage the mod_event_socket so that
> fs_cli will function the way it should instead popping up errors like it has
> been.
> 
> Also, in general, if one were to set up FS in background mode, how do you
> stop it from being in background mode?
> 
> PeterS
> 
> On Wed, Oct 6, 2010 at 11:14 PM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
> 
> >  fs_cli is a tool to connect to a running instance of FS, on the same
> > computer, or another computer. It communicates with FS using the
> > mod_event_socket loaded within FS. The server socket in FS listens for
> > requests on port 8021 by default, this can be changed in
> > event_socket.conf.xml
> >
> >
> >
> > /Peter
> >
> >
> >
> > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Peter Schrock
> > *Skickat:* den 7 oktober 2010 07:55
> > *Till:* FreeSWITCH Users Help
> > *?mne:* Re: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4
> >
> >
> >
> > In answer to your first question, yes. I guess my next question would be,
> > if I don't need fs_cli, then what is it for? I guess I am assuming to access
> > FS when it is running in background. What would be the difference between
> > running it in the foreground vs background?
> >
> > Also, I ran FS without -nc and it showed that mod_event_socket loaded
> > successfully. So why won't it with fs_cli?
> >
> >
> >
> > PeterS
> >
> > On Wed, Oct 6, 2010 at 5:29 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> >
> > Hi,
> >
> >
> >
> > Did you start FreeSWITCH before? fs_cli is only used to connect to the
> > console of a running FreeSWITCH instance.
> >
> >
> >
> > If you did, try starting it without the -nc switch, it should stay in the
> > foreground and tell you whats wrong, if you have a console you can also
> > check that mod_event_socket is properly loaded (load mod_event_socket)
> >
> >
> >
> > Mathieu Rene
> >
> > Avant-Garde Solutions Inc
> >
> > Office: + 1 (514) 664-1044 x100
> >
> > Cell: +1 (514) 664-1044 x200
> >
> > mrene at avgs.ca
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > On 2010-10-06, at 8:23 PM, Peter Schrock wrote:
> >
> >
> >
> >  Okay, so I managed to get FS working the other day and I even managed to
> > test a call and test my voicemail. All seemed to be working smoothly until,
> > because of the rain here, my power went out and I had to reboot my computer.
> > I logged in through the terminal, set up FS in background went to fs_cliand I get this error message:
> >
> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
> > Error]
> >
> > I managed to figure out that in the file "fs_cli.c" on line 1206 you find
> > the code for displaying this error message. The problem is that I don't know
> > why this error message is occurring. Does anyone have any helpful hints as
> > to what I should look at to resolve this problem?
> >
> > I even tried going to the git tree and make current, but that gave me
> > problems that forced me to turn off mod_spandsp and mod_skyopen in the
> > modules.conf, which I had running earlier. Any thoughts?
> >
> >
> >
> > PeterS
> >
> >
> >
> > PS
> >
> > I am not sure if this is of any help, but in addition to the error line
> > above, it also posted this info:
> >
> > Usage: fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x
> > command] [profile]
> >
> >
> >
> >   -?,-h --help                    Usage Information
> >
> >   -H, --host=hostname             Host to connect
> >
> >   -P, --port=port                 Port to connect (1 - 65535)
> >
> >   -u, --user=user at domain          user at domain
> >
> >   -p, --password=password         Password
> >
> >   -x, --execute=command           Execute Command and Exit
> >
> >   -l, --loglevel=command          Log Level
> >
> >   -q, --quiet                     Disable logging
> >
> >   -r, --retry                     Retry connection on failure
> >
> >   -R, --reconnect                 Reconnect if disconnected
> >
> >   -d, --debug=level               Debug Level (0 - 7)
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > !DSPAM:4cad626f32932643394250!
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From jaybinks at gmail.com  Thu Oct  7 14:48:47 2010
From: jaybinks at gmail.com (jay binks)
Date: Fri, 8 Oct 2010 07:48:47 +1000
Subject: [Freeswitch-users] Freeswitch & SNMP ( again ,
 but with a difference )
In-Reply-To: <65d96fc81003050144k76ffdf70rea30b109e2b19392@mail.gmail.com>
References: <ec2ce17e1003040038q153b5b8dgdc0137b3f152dca3@mail.gmail.com>
	<65d96fc81003041244t363450f2p8bf853c788d2295d@mail.gmail.com>
	<ec2ce17e1003041425p4e70995crf1e7b447a981e283@mail.gmail.com>
	<65d96fc81003050144k76ffdf70rea30b109e2b19392@mail.gmail.com>
Message-ID: <AANLkTimv8tLJuaQf2=7_c+8UxLtOa_HPtOz7NkHoeD2Z@mail.gmail.com>

sorry to bug you, but
did you manage to get anywhere with this ??

I think we need to try and expand this sort of stuff in the wiki...
maybe Michael Collins would like stuff like this for a few miscellaneous
recopies for his cook book ? :)

Jay


On Fri, Mar 5, 2010 at 7:44 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

>
>
> On Thu, Mar 4, 2010 at 11:25 PM, jay binks <jaybinks at gmail.com> wrote:
>
>> HUH, well there you go..   exactly what I was after...
>> someone has done exactly what I was thinking.
>>
>> I know you've sent it to me by email, but lets get this in the wiki.
>> Ill put this up from your email, but if you have more id encourage you to
>> share what youve done.
>>
>>
>
> let me finish the current project ..."wholesale routing machine". and i
> will put everything on wiki
>
>
>
> T.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From msc at freeswitch.org  Thu Oct  7 17:23:46 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 7 Oct 2010 17:23:46 -0700
Subject: [Freeswitch-users] Radius AAA
In-Reply-To: <4CAE21FB.3050700@gmail.com>
References: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>
	<AANLkTi=GiQaO-48BP=T8c-A+09sJMKzQH7BX=ewBdB8R@mail.gmail.com>
	<AANLkTikw8LGDqQki16D30b8JL0eJid2UOTs-bEaZSc2k@mail.gmail.com>
	<4CADC2E4.90608@gmail.com>
	<AANLkTik5c9yEKhvzsKNJJNnrvKwbAuVLoOk6jRgonbGX@mail.gmail.com>
	<4CAE1B5E.6010203@gmail.com>
	<AANLkTim9LP=m8UdVYrupxRQTU=_HqV79QNSLyPrB+qUY@mail.gmail.com>
	<4CAE21FB.3050700@gmail.com>
Message-ID: <AANLkTimRvzM-6Z-2bEtB7LFzsdD337awphpuqNpVLmQ5@mail.gmail.com>

I don't suppose one of you RADIUS experts would like to join the FS conf
call on a Wednesday and do a presentation on this? You would probably need
to do a primer on RADIUS in general to get everyone up to speed.

Please let me know.

Thanks,
MC

On Thu, Oct 7, 2010 at 12:39 PM, Nazim Aghabayov
<nazim.aghabayov at gmail.com>wrote:

>  Pardon me ) I was mean to say "imported some of your code from
> mod_rad_auth".
> Thank you once again for the module, it's extremely configurable and
> easy to use.
>
> Nazim.
>
> On 10/08/2010 12:24 AM, Tihomir Culjaga wrote:
> > i don't remember if i actually uploaded mod_xml_cdr patches ... hmmm
> >
> > anyhow i use odbc to send CDRS to a DB for statistics purpose (calculate
> > ASR/ACD/ROUTING) and radius interface for billing purpose as this hooks
> to
> > an external billing.
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Oct  7 17:30:56 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 7 Oct 2010 17:30:56 -0700
Subject: [Freeswitch-users] Mono 2.8 released and need to update
	mod_managed
In-Reply-To: <AANLkTi=_aLLty1F_iFkf_vQRuHzWTYJrZVc-7Zkos=Dh@mail.gmail.com>
References: <AANLkTin3rM+NaW1_O2d4HaDL9yuESGmZvd8v29Kxh6Y8@mail.gmail.com>
	<AANLkTimoS2=NaMhg2dVVEuyEjwEqagLie3KDVNHYv=8B@mail.gmail.com>
	<AANLkTi=_aLLty1F_iFkf_vQRuHzWTYJrZVc-7Zkos=Dh@mail.gmail.com>
Message-ID: <AANLkTimi+bXcwYGk575ATaxGC88MMn35j6P5m54oOMGM@mail.gmail.com>

Yitzchok,

Thanks for working on this, we appreciate it when people in the community
step up and help out. BTW, this particular topic is a little bit intense for
the users list - it is better for the -dev list.

Thanks,
MC

On Wed, Oct 6, 2010 at 8:20 PM, Yitzchok <adminjew at gmail.com> wrote:

> I got it to work and attached is the patch file with the changes I made to
> get it to work. (Jeff this includes the changes from my other post)
>
> This code changes has to be reviewed and I think you should checkout *mono_assembly_name_free
> (name); *I don't call that but maybe it should be called somewhere in the
> code.
>
> Thanks
>
> Yitzchok
>
>
>
> On Wed, Oct 6, 2010 at 9:48 PM, Yitzchok <adminjew at gmail.com> wrote:
>
>> It seems like glib itself is easy to get around by removing the glig
>> includes and changing g_free(x) to mono_free(x)
>>
>> But I am getting stuck by this one.
>>
>>
>>    - The MonoAssemblyName struct is no longer fully visible: to access
>>    its fields
>>
>> you need to use the newly-provided accessors. Note also that it can't be
>> allocated on the stack anymore and you'll need to create and destroy it with
>> the following API:
>>
>> MonoAssemlyName *aname = mono_assembly_name_new ("mscorlib");
>> mono_assembly_name_free (aname);
>>
>>
>>
>> (from http://www.mono-project.com/Embedding_Mono)
>>
>> Yitzchok
>>
>>
>>
>> On Wed, Oct 6, 2010 at 9:00 PM, Yitzchok <adminjew at gmail.com> wrote:
>>
>>> Mono 2.8 was released today with support for C# 4 and more
>>> http://www.mono-project.com/Release_Notes_Mono_2.8
>>>
>>> <http://www.mono-project.com/Release_Notes_Mono_2.8>It seems like there
>>> was some changes in the way applications embed mono (which includes removing
>>> the reference to glib and replacing it with eglib from what I understand) in
>>> this link you can find information for the changes made with embedding mono
>>> http://www.mono-project.com/Embedding_Mono
>>>
>>> I am trying to see if I can make it work but since I am just a C# windows
>>> developer (not a c or c++ dev) what I am doing is just hacking around and
>>> don't know if I will even get it to work but I think a c++ linux developer
>>> might get it to work with only a little work (the info needed I think is in
>>> the link above).
>>>
>>> So I am wondering if anyone is interested to get it to work.
>>>
>>> <http://www.mono-project.com/Embedding_Mono>
>>>
>>> Yitzchok
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Oct  7 17:36:22 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 7 Oct 2010 17:36:22 -0700
Subject: [Freeswitch-users] sip client
In-Reply-To: <AANLkTingm-baqQgx1VxBFcR5hWYe0qRhLZ5-pNxUJxsN@mail.gmail.com>
References: <AANLkTimnafsORnCFib+3FCpp=gp0ekhoevGggDzKLfmp@mail.gmail.com>
	<AANLkTi=bsnUVd+L5BFLtQ65viAFSeKuAv_RhNyqTMNGp@mail.gmail.com>
	<AANLkTik75hyXxBCFhAvob9_vjUkyGa4N_MRSqfYJ=dQg@mail.gmail.com>
	<AANLkTingm-baqQgx1VxBFcR5hWYe0qRhLZ5-pNxUJxsN@mail.gmail.com>
Message-ID: <AANLkTimuROE51KnW8kz7q1h6JX0OszYHvGrzGhr5dxft@mail.gmail.com>

GLWT.

I've had success on Android with Sipdroid. Others have used Acrobits on
iPhone. YMMV.
-MV

On Wed, Oct 6, 2010 at 10:06 PM, budi wibowo <bwibowo at gmail.com> wrote:

> thx, that looks great but i'm looking the ready to use product that can run
> on major mobile phone (blackbery iphone android winmo symbian)
> voipswitch is good, but the server running on windows platform. i prefer
> the server running on unix platform.
> any info please share
>
>
> regards
>
> budi
>
>
>
>
> On Fri, Oct 1, 2010 at 7:19 AM, jesse <chat2jesse at gmail.com> wrote:
>
>> Try open source client sip communicator that is in java.
>> On Sep 30, 2010 2:53 PM, "budi wibowo" <bwibowo at gmail.com> wrote:
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Oct  7 18:10:36 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 7 Oct 2010 18:10:36 -0700
Subject: [Freeswitch-users] Config issue after router/Firewall change
In-Reply-To: <AANLkTi=VTDYarQ6Jf+46b44cUc_vXOn-3EfNr3Rz8Jjs@mail.gmail.com>
References: <AANLkTi=8p5S40+JV3cAitrudYdXmbP3jcM5zXG2BeNEK@mail.gmail.com>
	<0ea901cb6579$8b77bf00$a2673d00$@com>
	<AANLkTi=VTDYarQ6Jf+46b44cUc_vXOn-3EfNr3Rz8Jjs@mail.gmail.com>
Message-ID: <AANLkTikpDrpADV+w=ihQNc9-4ObyEO5oRk+6sA_oaqce@mail.gmail.com>

On Wed, Oct 6, 2010 at 10:35 AM, Malay Thakershi <mthakershi at gmail.com>wrote:

> Router is pfsense
>

The pfsense guys have decent documentation and an IRC channel. You might
want to check in with them to make sure that you have all of the settings
that you need to have for pfsense to work with VoIP.

-MC
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From msc at freeswitch.org  Thu Oct  7 18:14:08 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 7 Oct 2010 18:14:08 -0700
Subject: [Freeswitch-users] Possible bug with bypass_media=true
	functionality
In-Reply-To: <AANLkTimqMMVvKR2PojPMnnnp-0KtWsHj43inGy43PDAN@mail.gmail.com>
References: <AANLkTimqMMVvKR2PojPMnnnp-0KtWsHj43inGy43PDAN@mail.gmail.com>
Message-ID: <AANLkTikr_E4wOkLU8t2jMyaT+E-Jec+6fg3oHxCWn66S@mail.gmail.com>

I don't know if it's a bug or not, but I think you probably will need to
create a dialplan and sample Lua script to demonstrate the behavior. Post
them on pastebin and link to this thread. We will see if we can reproduce
the symptoms and diagnose from there.

-MC

On Thu, Oct 7, 2010 at 7:37 AM, Pete Kelly <pkelly at gmail.com> wrote:

> Hi, I posted a question to this list yesterday because I was having
> problems with my calls ending as soon as they were bridged. I was
> originating 2 outbound calls from a lua script, then bridging them. I
> managed to get round the problem of the calls shutting down immediately by
> initiating calls using the originate command, thusly:
>
> originate sofia/lpmedia/2000 at 10.15.20.122 '&lua(click2call_call.lua)'
>
> This causes the lua script to initiate with a session already present and
> then a bridge to a new call (legB) completes successfully.
>
> Within the lua script I am bridging to leg B like this:
>
> session:execute("bridge", "[bypass_media_after_bridge=true]sofia/lpmedia/
> 91979197 at 10.15.20.122")
>
> This works, but the reINVITEs which get sent out after legB is established
> contain the wrong sip request URI.
>
> - The SIP URI in the reINVITE to Leg A is to the MEDIA IP of leg B (the SIP
> packet is sent to the correct IP, just the request URI contains the media
> IP)
> - The SIP URI in the reINVITE to Leg B is to the MEDIA IP of leg A... again
> the SIP packet is sent to the correct IP, it's just the request URI which
> contains the media IP.
>
> In some instances the request URI doesn't contain the username either, it
> is just INVITE sip:<MEDIA_IP>
>
> I am using the latest sources of freeswitch from git
>
> Is this a bug I need to report? It is causing me to receive 404 and 402
> errors on the reinvite as the request URI is invalid.
>
> Pete
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tgraziano at myitdepartment.net  Thu Oct  7 18:18:29 2010
From: tgraziano at myitdepartment.net (Tony Graziano)
Date: Thu, 7 Oct 2010 21:18:29 -0400
Subject: [Freeswitch-users] Config issue after router/Firewall change
In-Reply-To: <AANLkTi=VTDYarQ6Jf+46b44cUc_vXOn-3EfNr3Rz8Jjs@mail.gmail.com>
References: <AANLkTi=8p5S40+JV3cAitrudYdXmbP3jcM5zXG2BeNEK@mail.gmail.com>
	<0ea901cb6579$8b77bf00$a2673d00$@com>
	<AANLkTi=VTDYarQ6Jf+46b44cUc_vXOn-3EfNr3Rz8Jjs@mail.gmail.com>
Message-ID: <AANLkTik1WCsety1XkiUD9toee3qXEVF-Pk2pad1jS0az@mail.gmail.com>

Ensure the siproxd package is not installed. Make sure the outbound
rule for your voice lan is set for manual nat.

On 10/6/10, Malay Thakershi <mthakershi at gmail.com> wrote:
> Router is pfsense
>
> On Wed, Oct 6, 2010 at 12:11 PM, Peder <peder at networkoblivion.com> wrote:
>> What kind of firewall is it now and was it before? ?Turn off SIP ALG if
>> there is one (unless it is a PIX/ASA or Fortinet as those are the only two
>> that seem to handle it ok).
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Malay
>> Thakershi
>> Sent: Wednesday, October 06, 2010 11:17 AM
>> To: FreeSWITCH Users Help
>> Subject: [Freeswitch-users] Config issue after router/Firewall change
>>
>> We changed firewall our FS server is behind.
>>
>> All off sudden, no voice heard on answered calls.
>>
>> We made sure all rules are ditto as older router/firewall. We even
>> tried opening up everything but result is same. Called Vitelity: they
>> say according to them calls are OK.
>>
>> Please help with you suggestions.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device

======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgraziano at voice.myitdepartment.net
Fax: 434.984.8431

Email: tgraziano at myitdepartment.net

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpdesk at voice.myitdepartment.net
Fax: 434.984.8427

Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/

Why do mathematicians always confuse Halloween and Christmas?
Because 31 Oct = 25 Dec.


From msc at freeswitch.org  Thu Oct  7 18:24:45 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 7 Oct 2010 18:24:45 -0700
Subject: [Freeswitch-users] Question about effective caller ID &
 origination caller ID
In-Reply-To: <4CADEA81.5060007@mgtech.com>
References: <4CACBEDE.8060805@mgtech.com>
	<AANLkTikVj0YXKTfWEvhcFn6RLAicK4rSKt-3eXwXNSs+@mail.gmail.com>
	<AANLkTikYPAT2=V7euHq4+R+E1v4VnV=eJ7YnCf7kV0Ma@mail.gmail.com>
	<4CADEA81.5060007@mgtech.com>
Message-ID: <AANLkTi=wR8v=zUxe3P4PsJeyVkA4i7P_5rgYBTsfU2XD@mail.gmail.com>

Thanks for reporting. I will test and confirm the behavior of normal vs.
enterprise and make sure it is properly documented on the wiki.

-MC

On Thu, Oct 7, 2010 at 8:42 AM, Mario <mario_fs at mgtech.com> wrote:

> Thanks for responding Steve, I am beginning to wonder if this is by
> design or a bug. I am closer to the end of building the system for our
> small office and I thought this would be an easy one. Not so, I already
> checked enterprise (:_:) vs (,) and that is where the problem is. I
> tried the exports you suggested and had high hopes but no dice. Seems
> when using :_: everything is lost and has to be passed as part of the
> dialstring.  BTW, I need :_: because the SPA962 phones I use now don't
> use SUBSCRIBE only NOTIFY for the mailbox so I have to use multiple
> registrations since we are not big and share mailboxes. Thanks for the
> help. I wonder if there is a way to determine if this is design or bug....
>
> Both don't fix the problem:
> <action application="export"
> data="effective_caller_id_number=${caller_id_number}"/
>
> <action application="export"
> data="origination_caller_id_number=${caller_id_number}"/
>
>
> On 10/07/10 00:44, Steven Ayre wrote:
> > It's also possible that it's a difference between enterprise and
> > normal bridge syntax - I've never used enterprise so don't know too
> > much about it.
> >
> > -Steve
> >
> >
> > On 7 October 2010 08:41, Steven Ayre <steveayre at gmail.com> wrote:
> >> Have you tried using effective_* with export?
> >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_export
> >>
> >> -Steve
> >>
> >>
> >>
> >> On 6 October 2010 19:24, Mario <mario_fs at mgtech.com> wrote:
> >>> I have the inbound caller ID working for a bridge but can't reconcile
> >>> why effective_caller_id.. does not work. I spent a lot of time the last
> >>> few days reading the wiki and other web info. Since I can't find the
> >>> answer anywhere can anyone shed some light on this?
> >>>
> >>> The wiki states:
> >>> effective_caller_id_number
> >>> Sets the effective callerid number. This is automatically exported to
> >>> the B-leg; however, it is not valid in an origination string. In other
> >>> words, set this before calling bridge, otherwise use
> >>> origination_caller_id_number
> >>>
> >>> Does NOT work before the bridge:
> >>>        <!--action application="set"
> >>> data="effective_caller_id_number=${caller_id_number}"/>
> >>>        <action application="set"
> >>> data="effective_caller_id_name=${caller_id_name}"/-->
> >>>
> >>> Does NOT work before the bridge:
> >>>        <!--action application="set"
> >>> data="origination_caller_id_number=${caller_id_number}"/>
> >>>        <action application="set"
> >>> data="origination_caller_id_name=${caller_id_name}"/-->
> >>>        <!--action application="set"
> >>>
> >>> ONLY way IT WORKS (must be added before dialstring):
> >>>        <action application="bridge"
> >>>
> data="<originate_timeout=15,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/MGT@
> ${domain_name}:_:sofia/gateway/uuid1/19162223333:_:sofia/gateway/uuid1/19164445555"/>
> >>>
> >>>
> >>> The caller ID is lost in an extension, although I still don't know why
> >>> at least I got it back.
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From nazim.aghabayov at gmail.com  Thu Oct  7 18:55:58 2010
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Fri, 08 Oct 2010 06:55:58 +0500
Subject: [Freeswitch-users] Radius AAA
In-Reply-To: <AANLkTimRvzM-6Z-2bEtB7LFzsdD337awphpuqNpVLmQ5@mail.gmail.com>
References: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>	<AANLkTi=GiQaO-48BP=T8c-A+09sJMKzQH7BX=ewBdB8R@mail.gmail.com>	<AANLkTikw8LGDqQki16D30b8JL0eJid2UOTs-bEaZSc2k@mail.gmail.com>	<4CADC2E4.90608@gmail.com>	<AANLkTik5c9yEKhvzsKNJJNnrvKwbAuVLoOk6jRgonbGX@mail.gmail.com>	<4CAE1B5E.6010203@gmail.com>	<AANLkTim9LP=m8UdVYrupxRQTU=_HqV79QNSLyPrB+qUY@mail.gmail.com>	<4CAE21FB.3050700@gmail.com>
	<AANLkTimRvzM-6Z-2bEtB7LFzsdD337awphpuqNpVLmQ5@mail.gmail.com>
Message-ID: <4CAE7A2E.5040202@gmail.com>

  Despite the fact that I know very little about the Radius I'll gladly 
join and share my experience with integration of 3d party billing 
solution with a FreeSWITCH.

Thanks,
Nazim

On 10/08/2010 05:23 AM, Michael Collins wrote:
> I don't suppose one of you RADIUS experts would like to join the FS conf
> call on a Wednesday and do a presentation on this? You would probably need
> to do a primer on RADIUS in general to get everyone up to speed.
>
> Please let me know.
>
> Thanks,
> MC
>
> On Thu, Oct 7, 2010 at 12:39 PM, Nazim Aghabayov
> <nazim.aghabayov at gmail.com>wrote:
>
>>   Pardon me ) I was mean to say "imported some of your code from
>> mod_rad_auth".
>> Thank you once again for the module, it's extremely configurable and
>> easy to use.
>>
>> Nazim.
>>
>> On 10/08/2010 12:24 AM, Tihomir Culjaga wrote:
>>> i don't remember if i actually uploaded mod_xml_cdr patches ... hmmm
>>>
>>> anyhow i use odbc to send CDRS to a DB for statistics purpose (calculate
>>> ASR/ACD/ROUTING) and radius interface for billing purpose as this hooks
>> to
>>> an external billing.
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Thu Oct  7 18:55:36 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 7 Oct 2010 20:55:36 -0500
Subject: [Freeswitch-users] Possible bug with bypass_media=true
	functionality
In-Reply-To: <AANLkTikr_E4wOkLU8t2jMyaT+E-Jec+6fg3oHxCWn66S@mail.gmail.com>
References: <AANLkTimqMMVvKR2PojPMnnnp-0KtWsHj43inGy43PDAN@mail.gmail.com>
	<AANLkTikr_E4wOkLU8t2jMyaT+E-Jec+6fg3oHxCWn66S@mail.gmail.com>
Message-ID: <AANLkTimBv1opo=7vZRJ7pnQcgd42J6ERX5CgShS=Wi5F@mail.gmail.com>

Most likely its some nat related issue.
On Oct 7, 2010 8:19 PM, "Michael Collins" <msc at freeswitch.org> wrote:
> I don't know if it's a bug or not, but I think you probably will need to
> create a dialplan and sample Lua script to demonstrate the behavior. Post
> them on pastebin and link to this thread. We will see if we can reproduce
> the symptoms and diagnose from there.
>
> -MC
>
> On Thu, Oct 7, 2010 at 7:37 AM, Pete Kelly <pkelly at gmail.com> wrote:
>
>> Hi, I posted a question to this list yesterday because I was having
>> problems with my calls ending as soon as they were bridged. I was
>> originating 2 outbound calls from a lua script, then bridging them. I
>> managed to get round the problem of the calls shutting down immediately
by
>> initiating calls using the originate command, thusly:
>>
>> originate sofia/lpmedia/2000 at 10.15.20.122 '&lua(click2call_call.lua)'
>>
>> This causes the lua script to initiate with a session already present and
>> then a bridge to a new call (legB) completes successfully.
>>
>> Within the lua script I am bridging to leg B like this:
>>
>> session:execute("bridge", "[bypass_media_after_bridge=true]sofia/lpmedia/
>> 91979197 at 10.15.20.122")
>>
>> This works, but the reINVITEs which get sent out after legB is
established
>> contain the wrong sip request URI.
>>
>> - The SIP URI in the reINVITE to Leg A is to the MEDIA IP of leg B (the
SIP
>> packet is sent to the correct IP, just the request URI contains the media
>> IP)
>> - The SIP URI in the reINVITE to Leg B is to the MEDIA IP of leg A...
again
>> the SIP packet is sent to the correct IP, it's just the request URI which
>> contains the media IP.
>>
>> In some instances the request URI doesn't contain the username either, it
>> is just INVITE sip:<MEDIA_IP>
>>
>> I am using the latest sources of freeswitch from git
>>
>> Is this a bug I need to report? It is causing me to receive 404 and 402
>> errors on the reinvite as the request URI is invalid.
>>
>> Pete
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
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From mario_fs at mgtech.com  Thu Oct  7 19:10:19 2010
From: mario_fs at mgtech.com (Mario)
Date: Thu, 07 Oct 2010 19:10:19 -0700
Subject: [Freeswitch-users] Can shared mailbox signal multiple MWIs?
Message-ID: <4CAE7D8B.60707@mgtech.com>

If possible, how can FS light multiple userid/phones NOT using multiple
registrations MWI when a message is in a shared mailbox?

I have had a lot of trouble with multiple registrations and can't get
SLA/SCA to work on the SPA962s. Use of :_: was required in the bridge
and that was a lot of headaches. All due to use of shared mailboxes. I
can control what mailbox is played back via an extension but I can't
find a way to have the MWI indicator come on multiple userid/phones for
a shared mailbox on SPA962s.

If I can get past this I can finally dump shared lines, they were also a
pain on the SPA9000 I am trying to replace. Thanks!
Mario


From fs-list at communicatefreely.net  Thu Oct  7 19:15:12 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Thu, 07 Oct 2010 22:15:12 -0400
Subject: [Freeswitch-users] Dead lua scripts
In-Reply-To: <AANLkTinKznsWB6Zzf3cexnqHjesgQ_gsK++KkKD_h+U0@mail.gmail.com>
References: <4CAE15F7.9090807@communicatefreely.net>
	<AANLkTinKznsWB6Zzf3cexnqHjesgQ_gsK++KkKD_h+U0@mail.gmail.com>
Message-ID: <4CAE7EB0.2070600@communicatefreely.net>

Okay, I think I see where I'm going wrong with Lua.  I didn't realize that the script persists even
if the user hangs up.

When this does happen, is there any way to kill the thread and make the script stop?

I had to completely restart freeswitch to make these calls go away.  Is there a less disruptive way
to terminate the scripts?



Anthony Minessale wrote:
> look for endless loops in your code that are not using session:ready
> 
> On Thu, Oct 7, 2010 at 1:48 PM, Tim St. Pierre
> <fs-list at communicatefreely.net> wrote:


From pkelly at gmail.com  Fri Oct  8 00:15:18 2010
From: pkelly at gmail.com (Pete Kelly)
Date: Fri, 8 Oct 2010 08:15:18 +0100
Subject: [Freeswitch-users] Possible bug with bypass_media=true
	functionality
In-Reply-To: <AANLkTimBv1opo=7vZRJ7pnQcgd42J6ERX5CgShS=Wi5F@mail.gmail.com>
References: <AANLkTimqMMVvKR2PojPMnnnp-0KtWsHj43inGy43PDAN@mail.gmail.com>
	<AANLkTikr_E4wOkLU8t2jMyaT+E-Jec+6fg3oHxCWn66S@mail.gmail.com>
	<AANLkTimBv1opo=7vZRJ7pnQcgd42J6ERX5CgShS=Wi5F@mail.gmail.com>
Message-ID: <AANLkTi=hCcki-wNPsfXCjp2d0UJom76N5-H_e4gksMsH@mail.gmail.com>

On 8 October 2010 02:55, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> Most likely its some nat related issue.
>
I am seeing the same problem on a local network (calling 2 devices
registered to a local opensips proxy) and on a live setup calling tier 1
carriers with no NAT.

Would it help if I posted the sample scripts as requested?

Pete



> On Oct 7, 2010 8:19 PM, "Michael Collins" <msc at freeswitch.org> wrote:
> > I don't know if it's a bug or not, but I think you probably will need to
> > create a dialplan and sample Lua script to demonstrate the behavior. Post
> > them on pastebin and link to this thread. We will see if we can reproduce
> > the symptoms and diagnose from there.
> >
> > -MC
> >
> > On Thu, Oct 7, 2010 at 7:37 AM, Pete Kelly <pkelly at gmail.com> wrote:
> >
> >> Hi, I posted a question to this list yesterday because I was having
> >> problems with my calls ending as soon as they were bridged. I was
> >> originating 2 outbound calls from a lua script, then bridging them. I
> >> managed to get round the problem of the calls shutting down immediately
> by
> >> initiating calls using the originate command, thusly:
> >>
> >> originate sofia/lpmedia/2000 at 10.15.20.122 '&lua(click2call_call.lua)'
> >>
> >> This causes the lua script to initiate with a session already present
> and
> >> then a bridge to a new call (legB) completes successfully.
> >>
> >> Within the lua script I am bridging to leg B like this:
> >>
> >> session:execute("bridge",
> "[bypass_media_after_bridge=true]sofia/lpmedia/
> >> 91979197 at 10.15.20.122")
> >>
> >> This works, but the reINVITEs which get sent out after legB is
> established
> >> contain the wrong sip request URI.
> >>
> >> - The SIP URI in the reINVITE to Leg A is to the MEDIA IP of leg B (the
> SIP
> >> packet is sent to the correct IP, just the request URI contains the
> media
> >> IP)
> >> - The SIP URI in the reINVITE to Leg B is to the MEDIA IP of leg A...
> again
> >> the SIP packet is sent to the correct IP, it's just the request URI
> which
> >> contains the media IP.
> >>
> >> In some instances the request URI doesn't contain the username either,
> it
> >> is just INVITE sip:<MEDIA_IP>
> >>
> >> I am using the latest sources of freeswitch from git
> >>
> >> Is this a bug I need to report? It is causing me to receive 404 and 402
> >> errors on the reinvite as the request URI is invalid.
> >>
> >> Pete
> >>
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peter.olsson at visionutveckling.se  Fri Oct  8 00:16:08 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 8 Oct 2010 09:16:08 +0200
Subject: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4
In-Reply-To: <AANLkTinHyjVKiidD4d2X3d3hKDWDcRpokhVG8pU3dcUt@mail.gmail.com>
References: <AANLkTingeYfPT_b6dG3bKQOVjxYsmBTKhDuLw_3BKGac@mail.gmail.com>
	<A499E7B3-9727-4822-B2AB-8FDDCA3213FA@avgs.ca>
	<AANLkTinOUhDgzCB2rTL20_RzGq3=uS7zOKE064DOR0v6@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57E0E68E2E@cooper>
	<AANLkTinHyjVKiidD4d2X3d3hKDWDcRpokhVG8pU3dcUt@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57E0E6928E@cooper>

I'm pretty sure that fs_cli is used by everyone who uses FS, since we want the FreeSWITCH process itself to run in background.

I'm not really getting the problem here. Install FS with default config, start FS in background, then run fs_cli to connect to the running instance - is it really that hard?

If running in background, you can stop with "freeswitch -stop" or "fs_cli -x shutdown"

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Peter Schrock
Skickat: den 7 oktober 2010 21:41
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4

John Covici,

To make sure I understand correctly, do you me the copy found in:
/usr/local/freeswitch/libs/esl
or somewhere else?
Also, is there a wiki that discusses how to make those parameters agreeable?

Peter Olsson,
Thanks for clarifying that. I guess my question now would be (cause I am trying to use the book as much as possible to help me understand FS) is why would the book be insistent on using fs_cli and how many people actually use it. Is there a wiki discussing how to manage the mod_event_socket so that fs_cli will function the way it should instead popping up errors like it has been.

Also, in general, if one were to set up FS in background mode, how do you stop it from being in background mode?

PeterS
On Wed, Oct 6, 2010 at 11:14 PM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
fs_cli is a tool to connect to a running instance of FS, on the same computer, or another computer. It communicates with FS using the mod_event_socket loaded within FS. The server socket in FS listens for requests on port 8021 by default, this can be changed in event_socket.conf.xml

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Peter Schrock
Skickat: den 7 oktober 2010 07:55
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4

In answer to your first question, yes. I guess my next question would be, if I don't need fs_cli, then what is it for? I guess I am assuming to access FS when it is running in background. What would be the difference between running it in the foreground vs background?
Also, I ran FS without -nc and it showed that mod_event_socket loaded successfully. So why won't it with fs_cli?

PeterS
On Wed, Oct 6, 2010 at 5:29 PM, Mathieu Rene <mrene_lists at avgs.ca<mailto:mrene_lists at avgs.ca>> wrote:
Hi,

Did you start FreeSWITCH before? fs_cli is only used to connect to the console of a running FreeSWITCH instance.

If you did, try starting it without the -nc switch, it should stay in the foreground and tell you whats wrong, if you have a console you can also check that mod_event_socket is properly loaded (load mod_event_socket)

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca<mailto:mrene at avgs.ca>




On 2010-10-06, at 8:23 PM, Peter Schrock wrote:

Okay, so I managed to get FS working the other day and I even managed to test a call and test my voicemail. All seemed to be working smoothly until, because of the rain here, my power went out and I had to reboot my computer. I logged in through the terminal, set up FS in background went to fs_cli and I get this error message:
[ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection Error]
I managed to figure out that in the file "fs_cli.c" on line 1206 you find the code for displaying this error message. The problem is that I don't know why this error message is occurring. Does anyone have any helpful hints as to what I should look at to resolve this problem?
I even tried going to the git tree and make current, but that gave me problems that forced me to turn off mod_spandsp and mod_skyopen in the modules.conf, which I had running earlier. Any thoughts?

PeterS

PS
I am not sure if this is of any help, but in addition to the error line above, it also posted this info:
Usage: fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x command] [profile]

  -?,-h --help                    Usage Information
  -H, --host=hostname             Host to connect
  -P, --port=port                 Port to connect (1 - 65535)
  -u, --user=user at domain          user at domain
  -p, --password=password         Password
  -x, --execute=command           Execute Command and Exit
  -l, --loglevel=command          Log Level
  -q, --quiet                     Disable logging
  -r, --retry                     Retry connection on failure
  -R, --reconnect                 Reconnect if disconnected
  -d, --debug=level               Debug Level (0 - 7)

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From abid_freeswitch at live.com  Fri Oct  8 01:24:36 2010
From: abid_freeswitch at live.com (Abid Saleem)
Date: Fri, 8 Oct 2010 14:24:36 +0600
Subject: [Freeswitch-users] FS Components
Message-ID: <COL109-W232E0055DD6B7F7E508364F4500@phx.gbl>


Hi,
Could someone through some light on what are the functions FS can provide out of the below.
SIP Server/Proxy ServerLocation/Presence ServerRedirect Server
Thanks.
Regards-----------Abid SaleemSr. Product Manager Terminus Technologies 		 	   		  
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From mnhassan at usa.net  Fri Oct  8 01:32:48 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Fri, 8 Oct 2010 14:32:48 +0600
Subject: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57E0E6928E@cooper>
References: <AANLkTingeYfPT_b6dG3bKQOVjxYsmBTKhDuLw_3BKGac@mail.gmail.com>
	<A499E7B3-9727-4822-B2AB-8FDDCA3213FA@avgs.ca>
	<AANLkTinOUhDgzCB2rTL20_RzGq3=uS7zOKE064DOR0v6@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57E0E68E2E@cooper>
	<AANLkTinHyjVKiidD4d2X3d3hKDWDcRpokhVG8pU3dcUt@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57E0E6928E@cooper>
Message-ID: <AANLkTikr9c8a34hmyqqeSZCHgeKqZo6kc0gVRku=tGmR@mail.gmail.com>

Did you check if the FreeSWITCH process is running?

Regards
HASSAN


On 2010-10-08, Peter Olsson <peter.olsson at visionutveckling.se> wrote:
> I'm pretty sure that fs_cli is used by everyone who uses FS, since we want
> the FreeSWITCH process itself to run in background.
>
> I'm not really getting the problem here. Install FS with default config,
> start FS in background, then run fs_cli to connect to the running instance -
> is it really that hard?
>
> If running in background, you can stop with "freeswitch -stop" or "fs_cli -x
> shutdown"
>
> /Peter
>
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Peter Schrock
> Skickat: den 7 oktober 2010 21:41
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4
>
> John Covici,
>
> To make sure I understand correctly, do you me the copy found in:
> /usr/local/freeswitch/libs/esl
> or somewhere else?
> Also, is there a wiki that discusses how to make those parameters agreeable?
>
> Peter Olsson,
> Thanks for clarifying that. I guess my question now would be (cause I am
> trying to use the book as much as possible to help me understand FS) is why
> would the book be insistent on using fs_cli and how many people actually use
> it. Is there a wiki discussing how to manage the mod_event_socket so that
> fs_cli will function the way it should instead popping up errors like it has
> been.
>
> Also, in general, if one were to set up FS in background mode, how do you
> stop it from being in background mode?
>
> PeterS
> On Wed, Oct 6, 2010 at 11:14 PM, Peter Olsson
> <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>
> wrote:
> fs_cli is a tool to connect to a running instance of FS, on the same
> computer, or another computer. It communicates with FS using the
> mod_event_socket loaded within FS. The server socket in FS listens for
> requests on port 8021 by default, this can be changed in
> event_socket.conf.xml
>
> /Peter
>
> Fr?n:
> freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>]
> F?r Peter Schrock
> Skickat: den 7 oktober 2010 07:55
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4
>
> In answer to your first question, yes. I guess my next question would be, if
> I don't need fs_cli, then what is it for? I guess I am assuming to access FS
> when it is running in background. What would be the difference between
> running it in the foreground vs background?
> Also, I ran FS without -nc and it showed that mod_event_socket loaded
> successfully. So why won't it with fs_cli?
>
> PeterS
> On Wed, Oct 6, 2010 at 5:29 PM, Mathieu Rene
> <mrene_lists at avgs.ca<mailto:mrene_lists at avgs.ca>> wrote:
> Hi,
>
> Did you start FreeSWITCH before? fs_cli is only used to connect to the
> console of a running FreeSWITCH instance.
>
> If you did, try starting it without the -nc switch, it should stay in the
> foreground and tell you whats wrong, if you have a console you can also
> check that mod_event_socket is properly loaded (load mod_event_socket)
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca<mailto:mrene at avgs.ca>
>
>
>
>
> On 2010-10-06, at 8:23 PM, Peter Schrock wrote:
>
> Okay, so I managed to get FS working the other day and I even managed to
> test a call and test my voicemail. All seemed to be working smoothly until,
> because of the rain here, my power went out and I had to reboot my computer.
> I logged in through the terminal, set up FS in background went to fs_cli and
> I get this error message:
> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
> Error]
> I managed to figure out that in the file "fs_cli.c" on line 1206 you find
> the code for displaying this error message. The problem is that I don't know
> why this error message is occurring. Does anyone have any helpful hints as
> to what I should look at to resolve this problem?
> I even tried going to the git tree and make current, but that gave me
> problems that forced me to turn off mod_spandsp and mod_skyopen in the
> modules.conf, which I had running earlier. Any thoughts?
>
> PeterS
>
> PS
> I am not sure if this is of any help, but in addition to the error line
> above, it also posted this info:
> Usage: fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x
> command] [profile]
>
>   -?,-h --help                    Usage Information
>   -H, --host=hostname             Host to connect
>   -P, --port=port                 Port to connect (1 - 65535)
>   -u, --user=user at domain          user at domain
>   -p, --password=password         Password
>   -x, --execute=command           Execute Command and Exit
>   -l, --loglevel=command          Log Level
>   -q, --quiet                     Disable logging
>   -r, --retry                     Retry connection on failure
>   -R, --reconnect                 Reconnect if disconnected
>   -d, --debug=level               Debug Level (0 - 7)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4cae242c32931240012030!
>

-- 
Sent from my mobile device


From ale975 at gmail.com  Fri Oct  8 02:25:12 2010
From: ale975 at gmail.com (Ale)
Date: Fri, 8 Oct 2010 11:25:12 +0200
Subject: [Freeswitch-users] Fax hint request
In-Reply-To: <AANLkTimTbh_wCYd4UDE2LXe1NTxWj59g_ttC5cbAdy79@mail.gmail.com>
References: <AANLkTim8NhxTcypcA9kuT4vnmRNObZ2JbZzgVNdGpsgW@mail.gmail.com>
	<AANLkTimmjxQjfzyb=+ZxvJzFDR5EduyM8pLXBXbX+62p@mail.gmail.com>
	<0eb201cb6579$de02ba40$9a082ec0$@com>
	<AANLkTimTbh_wCYd4UDE2LXe1NTxWj59g_ttC5cbAdy79@mail.gmail.com>
Message-ID: <AANLkTinv4NDrK0MJR4P0e_1++-Ak9XE6ALvw8uKDt8E2@mail.gmail.com>

After fs sent first invite, the remote party (my operator) inside 183
reply change the contact ip, this one send rtp to fs, but it does not
reply, and after seconds the call fail.

i'm new to fs, not seeing any response, I think I made some stupid
mistake, but really any small hint ?


On Thu, Oct 7, 2010 at 9:34 AM, Ale <ale975 at gmail.com> wrote:
> Hi Peder
>
> On Wed, Oct 6, 2010 at 7:13 PM, Peder <peder at networkoblivion.com> wrote:
>> What are you trying to do? ?And what is the problem? ?This works for me to
>> receive a fax:
>>
>
> I'm trying to send fax, i've not tested rxfax. I've a little script
> that send fax via originate command, i also tried to send manually via
> cli with the same result.
>
> If i set enable-t38 to false all seems to works, but when i enable it,
> all originate fail with the error show in the first mail. It's
> absolutely reproducible, so i believe to make something wrong, but i
> does not understand what could be. I'm trying to read the
> documentation but configuration seems to be correct.
>
> If you want a complete console output, a dump or any other information
> feel free to ask, i would be happy if someone point me on the right
> direction.
>
> Many thanx
> Ale
>


From riedinger at sns.eu  Fri Oct  8 03:40:00 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Fri, 08 Oct 2010 12:40:00 +0200
Subject: [Freeswitch-users] Problematic Behaviour of FS regarding ptime
	negotiation
Message-ID: <4CAEF500.1050202@sns.eu>

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From david.ponzone at ipeva.fr  Fri Oct  8 03:48:50 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 8 Oct 2010 12:48:50 +0200
Subject: [Freeswitch-users] Problematic Behaviour of FS regarding ptime
	negotiation
In-Reply-To: <4CAEF500.1050202@sns.eu>
References: <4CAEF500.1050202@sns.eu>
Message-ID: <3E86F0B9-A2F1-4F7A-979D-2C11F84EC1D7@ipeva.fr>

Jan, no ptime means 20ms.
That's the RFC.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 08/10/2010 ? 12:40, Jan Riedinger a ?crit :

> I'm terminating various destination by various carriers. After  
> migrating one customer to Freeswitch, we observed problems for the  
> termination of a specific route for a specific carrier. I tried to  
> examine the problem in detail and I think it's related to problems  
> regarding the ptime negotiation. I think Freeswitch doesn't breach  
> any RFC, but I'm not sure, if the behaviour is optimal.
>
> The SDP of the Caller INVITE-Message at time 1160,056 in the  
> attached trace doesn't include any ptime setting. Nevertheless  
> Freeswitch includes a ptime=20 media attribute in the forwarded  
> INVITE message at time 1160,065. The ringing SDP sent by the callee  
> at time 1161,948 again doesn't include any ptime setting.  
> Nevertheless, Freeswitch includes in the Session Progress SDP (at  
> time 1164,240) a ptime=20 media atrribute. Why try Freeswitch to  
> force the usage of ptime=20 for the communication?
>
> The OK SDP at time 1164,240 again doesn't contain a ptime media  
> attribute. Nevertheless, the Freeswitch add a ptime=20 media  
> attribute forwarded to the caller at 1164,256.
>
> It seems that the callee is sending in the following with a frame  
> size of 60 bytes - it never claimed to use ptime=20 and according  
> the RFC 3264 it SHOULD send with ptime=20 because of the received  
> INVITE message specification, but it DON'T HAVE to send with ptime=20.
>
> At next Freeswitch tries to fix "the issue". In the logfile I found:
> e686b430-5d2d-488b-8b58-0fca1965eea7 2010-10-07 15:20:25.673206  
> [WARNING] mod_sofia.c:1033 We were told to use ptime 20 but what  
> they meant to say was 60
> This issue has so far been identified to happen on the following  
> broken platforms/devices:
> Linksys/Sipura aka Cisco
> ShoreTel
> Sonus/L3
> We will try to fix it but some of the devices on this list are so  
> broken,
> who knows what will happen..
> This log message isn't correct. The callee never specified anything  
> about the usage of a specific ptime. Furthermore, according RFC  
> 3264  the ptime doesn't specify the frame size, which will be used  
> to send packages by the side, which specify it in the SDP. In the  
> RFC 3264 is written:
>    If the ptime attribute is present for a stream, it indicates the
>    desired packetization interval that the offerer would like to
>    receive.
>
>    ...
>
>    There is now requirement that the packetization interval be the  
> same in each direction for a particular stream.
>
> IMHO that means, that it isn't possible in principle that a device  
> is lying about it's ptime usage, because it only specify by the  
> media attribute the packetization it likes to receive and doesn't  
> specify the packetization it will use itself.
>
> For fixing "the problem" Freeswitch sends a re-INVITE message at  
> 1164,777. This message includes in the message header "X-Broken- 
> PTIME: Adv=20; Sent=60", and ptime = 60 media attribute.
> The callee fails to process this re-INVITE and drops the call.
>
> I made the trace after I set the newly introduced parameter  
> "passthru_ptime_mismatch=true" (it's documented in the Wiki since  
> yesterday). Does it make sense, that Freeswitch tries to fix any  
> ptime setting if this variable is set to true?
>
> If someone wants to examine this issue more detailed, I can provide  
> the Wireshark-cap file of the call and the debug output of Freeswitch.
>
>
> Thank you in advance
>            Jan
>
>
>
>
>
> -- 
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                         E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>
> AG Charlottenburg - HRB 71973
> <Ptime Problem Call  
> Trace.tif>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at ipeva.fr  Fri Oct  8 04:06:19 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 8 Oct 2010 13:06:19 +0200
Subject: [Freeswitch-users] Problematic Behaviour of FS regarding ptime
	negotiation
In-Reply-To: <4CAEF500.1050202@sns.eu>
References: <4CAEF500.1050202@sns.eu>
Message-ID: <E18C5FFA-F9BD-43F0-9697-1BCC7A7176C3@ipeva.fr>

Jan,

to answer to 2 others questions you ask:

Why FS tries to enforce 20 ?
Well, the default is 20ms for most codecs, except perhaps G723, and no  
explicit ptime means 20ms for most codecs.
So your carrier is sending no ptime, meaning they want 20.
FS agrees and send back 20 (and explictly, because smart people always  
do things explictly and avoid relying on default values/behaviours).

So, yes, the message displayed by FS is correct at some point:
FS asked for 20ms, and your carrier is sending 60ms.

Now, I see your point: perhaps the phrase is not very clear.
I think the issue is (and Anthony or Brian will correct me on this if  
required) that FS tries to negotiate the same ptime on both  
directions, because what the RFC says about asymmetrical ptimes is  
scary, AFAIK. I heard people reporting major issues trying to do this.
Ok the RFC allows it, but as usual, it was probably badly implemented  
by most vendors, and anyway, there is no real benefit.
So FS tries to stay simple.
I think that's what FS means by "We were told": the other party asked  
us for 20ms, and as we like to keep things simple, we also asked for  
20ms, and they send back 60ms, those p....bast.... :)

Basically, I think what you are asking is a new parameter that would  
instruct FS to stop trying to re-packetize and accept asymmetrical  
ptimes.
About the message, you can get rid of it with rtp-autofix- 
timing=false, but use it at your own risk.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 08/10/2010 ? 12:40, Jan Riedinger a ?crit :

> I'm terminating various destination by various carriers. After  
> migrating one customer to Freeswitch, we observed problems for the  
> termination of a specific route for a specific carrier. I tried to  
> examine the problem in detail and I think it's related to problems  
> regarding the ptime negotiation. I think Freeswitch doesn't breach  
> any RFC, but I'm not sure, if the behaviour is optimal.
>
> The SDP of the Caller INVITE-Message at time 1160,056 in the  
> attached trace doesn't include any ptime setting. Nevertheless  
> Freeswitch includes a ptime=20 media attribute in the forwarded  
> INVITE message at time 1160,065. The ringing SDP sent by the callee  
> at time 1161,948 again doesn't include any ptime setting.  
> Nevertheless, Freeswitch includes in the Session Progress SDP (at  
> time 1164,240) a ptime=20 media atrribute. Why try Freeswitch to  
> force the usage of ptime=20 for the communication?
>
> The OK SDP at time 1164,240 again doesn't contain a ptime media  
> attribute. Nevertheless, the Freeswitch add a ptime=20 media  
> attribute forwarded to the caller at 1164,256.
>
> It seems that the callee is sending in the following with a frame  
> size of 60 bytes - it never claimed to use ptime=20 and according  
> the RFC 3264 it SHOULD send with ptime=20 because of the received  
> INVITE message specification, but it DON'T HAVE to send with ptime=20.
>
> At next Freeswitch tries to fix "the issue". In the logfile I found:
> e686b430-5d2d-488b-8b58-0fca1965eea7 2010-10-07 15:20:25.673206  
> [WARNING] mod_sofia.c:1033 We were told to use ptime 20 but what  
> they meant to say was 60
> This issue has so far been identified to happen on the following  
> broken platforms/devices:
> Linksys/Sipura aka Cisco
> ShoreTel
> Sonus/L3
> We will try to fix it but some of the devices on this list are so  
> broken,
> who knows what will happen..
> This log message isn't correct. The callee never specified anything  
> about the usage of a specific ptime. Furthermore, according RFC  
> 3264  the ptime doesn't specify the frame size, which will be used  
> to send packages by the side, which specify it in the SDP. In the  
> RFC 3264 is written:
>    If the ptime attribute is present for a stream, it indicates the
>    desired packetization interval that the offerer would like to
>    receive.
>
>    ...
>
>    There is now requirement that the packetization interval be the  
> same in each direction for a particular stream.
>
> IMHO that means, that it isn't possible in principle that a device  
> is lying about it's ptime usage, because it only specify by the  
> media attribute the packetization it likes to receive and doesn't  
> specify the packetization it will use itself.
>
> For fixing "the problem" Freeswitch sends a re-INVITE message at  
> 1164,777. This message includes in the message header "X-Broken- 
> PTIME: Adv=20; Sent=60", and ptime = 60 media attribute.
> The callee fails to process this re-INVITE and drops the call.
>
> I made the trace after I set the newly introduced parameter  
> "passthru_ptime_mismatch=true" (it's documented in the Wiki since  
> yesterday). Does it make sense, that Freeswitch tries to fix any  
> ptime setting if this variable is set to true?
>
> If someone wants to examine this issue more detailed, I can provide  
> the Wireshark-cap file of the call and the debug output of Freeswitch.
>
>
> Thank you in advance
>            Jan
>
>
>
>
>
> -- 
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                         E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>
> AG Charlottenburg - HRB 71973
> <Ptime Problem Call  
> Trace.tif>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From pkelly at gmail.com  Fri Oct  8 04:09:00 2010
From: pkelly at gmail.com (Pete Kelly)
Date: Fri, 8 Oct 2010 12:09:00 +0100
Subject: [Freeswitch-users] Possible bug with bypass_media=true
	functionality
In-Reply-To: <AANLkTi=hCcki-wNPsfXCjp2d0UJom76N5-H_e4gksMsH@mail.gmail.com>
References: <AANLkTimqMMVvKR2PojPMnnnp-0KtWsHj43inGy43PDAN@mail.gmail.com>
	<AANLkTikr_E4wOkLU8t2jMyaT+E-Jec+6fg3oHxCWn66S@mail.gmail.com>
	<AANLkTimBv1opo=7vZRJ7pnQcgd42J6ERX5CgShS=Wi5F@mail.gmail.com>
	<AANLkTi=hCcki-wNPsfXCjp2d0UJom76N5-H_e4gksMsH@mail.gmail.com>
Message-ID: <AANLkTi==RMLLUU6ewQZHa3x=cPT40beoXGRX3mNR5NH-@mail.gmail.com>

Hi

I've checked again locally against some other reinvites, and it seems it is
normal to use the media IP in the reinvite sip request URI.

The actual problem I am having, is that on the live environment only, the
reinvite request URI does not contain a username, which causes a 404 0r a
402 to be generated.

I've tried this with a number of downstream carriers with the same result,
so it seems freeswitch is deciding not to include the username for some
reason. I can't replicate this locally unfortunately, even using direct IP
dialling.

I am using this command to initiate the scenario from the fs_cli

originate
{origination_uuid=74f1bee7-9078-4d0d-b686-c2dc4334e1a,origination_caller_id_number=TestTest}sofia/lpmedia/
1001 at 10.15.20.14:5063 '&lua(scenario.lua)'

the lpmedia scenario file is here: http://pastebin.com/0NYnxZFH
the lua script is here: http://pastebin.com/f46N4WnM

If anyone could shed any light I would appreciate it... this is the last
step in what I am trying to achieve!

Thanks

Pete

On 8 October 2010 08:15, Pete Kelly <pkelly at gmail.com> wrote:

> On 8 October 2010 02:55, Anthony Minessale <anthony.minessale at gmail.com>wrote:
>
>> Most likely its some nat related issue.
>>
> I am seeing the same problem on a local network (calling 2 devices
> registered to a local opensips proxy) and on a live setup calling tier 1
> carriers with no NAT.
>
> Would it help if I posted the sample scripts as requested?
>
> Pete
>
>
>
>>  On Oct 7, 2010 8:19 PM, "Michael Collins" <msc at freeswitch.org> wrote:
>> > I don't know if it's a bug or not, but I think you probably will need to
>> > create a dialplan and sample Lua script to demonstrate the behavior.
>> Post
>> > them on pastebin and link to this thread. We will see if we can
>> reproduce
>> > the symptoms and diagnose from there.
>> >
>> > -MC
>> >
>> > On Thu, Oct 7, 2010 at 7:37 AM, Pete Kelly <pkelly at gmail.com> wrote:
>> >
>> >> Hi, I posted a question to this list yesterday because I was having
>> >> problems with my calls ending as soon as they were bridged. I was
>> >> originating 2 outbound calls from a lua script, then bridging them. I
>> >> managed to get round the problem of the calls shutting down immediately
>> by
>> >> initiating calls using the originate command, thusly:
>> >>
>> >> originate sofia/lpmedia/2000 at 10.15.20.122 '&lua(click2call_call.lua)'
>> >>
>> >> This causes the lua script to initiate with a session already present
>> and
>> >> then a bridge to a new call (legB) completes successfully.
>> >>
>> >> Within the lua script I am bridging to leg B like this:
>> >>
>> >> session:execute("bridge",
>> "[bypass_media_after_bridge=true]sofia/lpmedia/
>> >> 91979197 at 10.15.20.122")
>> >>
>> >> This works, but the reINVITEs which get sent out after legB is
>> established
>> >> contain the wrong sip request URI.
>> >>
>> >> - The SIP URI in the reINVITE to Leg A is to the MEDIA IP of leg B (the
>> SIP
>> >> packet is sent to the correct IP, just the request URI contains the
>> media
>> >> IP)
>> >> - The SIP URI in the reINVITE to Leg B is to the MEDIA IP of leg A...
>> again
>> >> the SIP packet is sent to the correct IP, it's just the request URI
>> which
>> >> contains the media IP.
>> >>
>> >> In some instances the request URI doesn't contain the username either,
>> it
>> >> is just INVITE sip:<MEDIA_IP>
>> >>
>> >> I am using the latest sources of freeswitch from git
>> >>
>> >> Is this a bug I need to report? It is causing me to receive 404 and 402
>> >> errors on the reinvite as the request URI is invalid.
>> >>
>> >> Pete
>> >>
>> >>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From riedinger at sns.eu  Fri Oct  8 04:16:47 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Fri, 08 Oct 2010 13:16:47 +0200
Subject: [Freeswitch-users] Problematic Behaviour of FS regarding ptime
 negotiation
In-Reply-To: <3E86F0B9-A2F1-4F7A-979D-2C11F84EC1D7@ipeva.fr>
References: <4CAEF500.1050202@sns.eu>
	<3E86F0B9-A2F1-4F7A-979D-2C11F84EC1D7@ipeva.fr>
Message-ID: <4CAEFD9F.8090407@sns.eu>

  Propably you refer RFC 2327:

      Note that RTP audio formats typically do not include information
      about the number of samples per packet.  If a non-default (as
      defined in the RTP Audio/Video Profile) packetisation is required,
      the "ptime" attribute is used as given below.

or  RFC 4566:
          Note: RTP audio formats typically do not include information
          about the number of samples per packet.  If a non-default (as
          defined in the RTP Audio/Video Profile) packetisation is
          required, the "ptime" attribute is used as given above.


IHMO that means, that the default packetisation CAN be used, if nothing 
is specified, but it doesn't mean that it SHOULD be used. But even if a 
ptime is specified, the RFC only says that the specified ptime SHOULD be 
used, what is different from that it HAVE to be used. Thus it doesn't 
breach any RFC, if a device is using another packetisation nevertheless.

BR
    Jan


Am 08.10.2010 12:48, schrieb David Ponzone:
> Jan, no ptime means 20ms.
> That's the RFC.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IP eva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 08/10/2010 ? 12:40, Jan Riedinger a ?crit :
>
>> I'm terminating various destination by various carriers. After 
>> migrating one customer to Freeswitch, we observed problems for the 
>> termination of a specific route for a specific carrier. I tried to 
>> examine the problem in detail and I think it's related to problems 
>> regarding the ptime negotiation. I think Freeswitch doesn't breach 
>> any RFC, but I'm not sure, if the behaviour is optimal.
>>
>> The SDP of the Caller INVITE-Message at time 1160,056 in the attached 
>> trace doesn't include any ptime setting. Nevertheless Freeswitch 
>> includes a ptime=20 media attribute in the forwarded INVITE message 
>> at time 1160,065. The ringing SDP sent by the callee at time 1161,948 
>> again doesn't include any ptime setting. Nevertheless, Freeswitch 
>> includes in the Session Progress SDP (at time 1164,240) a ptime=20 
>> media atrribute. Why try Freeswitch to force the usage of ptime=20 
>> for the communication?
>>
>> The OK SDP at time 1164,240 again doesn't contain a ptime media 
>> attribute. Nevertheless, the Freeswitch add a ptime=20 media 
>> attribute forwarded to the caller at 1164,256.
>>
>> It seems that the callee is sending in the following with a frame 
>> size of 60 bytes - it never claimed to use ptime=20 and according the 
>> RFC 3264 it SHOULD send with ptime=20 because of the received INVITE 
>> message specification, but it DON'T HAVE to send with ptime=20.
>>
>> At next Freeswitch tries to fix "the issue". In the logfile I found:
>>
>>     e686b430-5d2d-488b-8b58-0fca1965eea7 2010-10-07 15:20:25.673206
>>     [WARNING] mod_sofia.c:1033 We were told to use ptime 20 but what
>>     they meant to say was 60
>>     This issue has so far been identified to happen on the following
>>     broken platforms/devices:
>>     Linksys/Sipura aka Cisco
>>     ShoreTel
>>     Sonus/L3
>>     We will try to fix it but some of the devices on this list are so
>>     broken,
>>     who knows what will happen..
>>
>> This log message isn't correct. The callee never specified anything 
>> about the usage of a specific ptime. Furthermore, according RFC 3264  
>> the ptime doesn't specify the frame size, which will be used to send 
>> packages by the side, which specify it in the SDP. In the RFC 3264 is 
>> written:
>>     If the ptime attribute is present for a stream, it indicates the
>>     desired packetization interval that the offerer would like to
>>      receive .
>>
>>     ...
>>
>>     There is now requirement that the packetization interval be the same in each direction for a particular stream.
>>
>> IMHO that means, that it isn't possible in principle that a device is 
>> lying about it's ptime usage, because it only specify by the media 
>> attribute the packetization it likes to receive and doesn't specify 
>> the packetization it will use itself.
>>
>> For fixing "the problem" Freeswitch sends a re-INVITE message at 
>> 1164,777. This message includes in the message header 
>> "X-Broken-PTIME: Adv=20; Sent=60", and ptime = 60 media attribute.
>> The callee fails to process this re-INVITE and drops the call.
>>
>> I made the trace after I set the newly introduced parameter 
>> "passthru_ptime_mismatch=true" (it's documented in the Wiki since 
>> yesterday). Does it make sense, that Freeswitch tries to fix any 
>> ptime setting if this variable is set to true?
>>
>> If someone wants to examine this issue more detailed, I can provide 
>> the Wireshark-cap file of the call and the debug output of Freeswitch.
>>
>>
>> Thank you in advance
>>            Jan
>>
>>
>>
>>
>>
>> -- 
>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>                                          E-Mail:riedinger at sns.eu  <mailto:riedinger at sns.eu>
>> SNS Consult GmbH                        ICQ   :  163-237-041
>> S?dwestkorso 49a                        MSN   :jan at sns-consult.de  <mailto:jan at sns-consult.de>
>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>
>> AG Charlottenburg - HRB 71973
>> <Ptime Problem Call 
>> Trace.tif>_______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                         E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973

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From david.ponzone at ipeva.fr  Fri Oct  8 04:27:07 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 8 Oct 2010 13:27:07 +0200
Subject: [Freeswitch-users] Problematic Behaviour of FS regarding ptime
	negotiation
In-Reply-To: <4CAEFD9F.8090407@sns.eu>
References: <4CAEF500.1050202@sns.eu>
	<3E86F0B9-A2F1-4F7A-979D-2C11F84EC1D7@ipeva.fr>
	<4CAEFD9F.8090407@sns.eu>
Message-ID: <F920C4A6-EC3D-4E40-85E6-14C25DC37067@ipeva.fr>

It's exactly where RFCs start to be stupid, IMHO.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 08/10/2010 ? 13:16, Jan Riedinger a ?crit :

> Propably you refer RFC 2327:
>      Note that RTP audio formats typically do not include information
>      about the number of samples per packet.  If a non-default (as
>      defined in the RTP Audio/Video Profile) packetisation is  
> required,
>      the "ptime" attribute is used as given below.
>
> or  RFC 4566:
>          Note: RTP audio formats typically do not include information
>          about the number of samples per packet.  If a non-default (as
>          defined in the RTP Audio/Video Profile) packetisation is
>          required, the "ptime" attribute is used as given above.
>
> IHMO that means, that the default packetisation CAN be used, if  
> nothing is specified, but it doesn't mean that it SHOULD be used.  
> But even if a ptime is specified, the RFC only says that the  
> specified ptime SHOULD be used, what is different from that it HAVE  
> to be used. Thus it doesn't breach any RFC, if a device is using  
> another packetisation nevertheless.
>
> BR
>    Jan
>
>
> Am 08.10.2010 12:48, schrieb David Ponzone:
>>
>> Jan, no ptime means 20ms.
>> That's the RFC.
>>
>> David Ponzone   Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client  IP eva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ? l'intention exclusive de ses destinataires. Toute  
>> utilisation ou diffusion non autoris?e est interdite. Tout message  
>> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
>> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
>> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>>
>> Le 08/10/2010 ? 12:40, Jan Riedinger a ?crit :
>>
>>> I'm terminating various destination by various carriers. After  
>>> migrating one customer to Freeswitch, we observed problems for the  
>>> termination of a specific route for a specific carrier. I tried to  
>>> examine the problem in detail and I think it's related to problems  
>>> regarding the ptime negotiation. I think Freeswitch doesn't breach  
>>> any RFC, but I'm not sure, if the behaviour is optimal.
>>>
>>> The SDP of the Caller INVITE-Message at time 1160,056 in the  
>>> attached trace doesn't include any ptime setting. Nevertheless  
>>> Freeswitch includes a ptime=20 media attribute in the forwarded  
>>> INVITE message at time 1160,065. The ringing SDP sent by the  
>>> callee at time 1161,948 again doesn't include any ptime setting.  
>>> Nevertheless, Freeswitch includes in the Session Progress SDP (at  
>>> time 1164,240) a ptime=20 media atrribute. Why try Freeswitch to  
>>> force the usage of ptime=20 for the communication?
>>>
>>> The OK SDP at time 1164,240 again doesn't contain a ptime media  
>>> attribute. Nevertheless, the Freeswitch add a ptime=20 media  
>>> attribute forwarded to the caller at 1164,256.
>>>
>>> It seems that the callee is sending in the following with a frame  
>>> size of 60 bytes - it never claimed to use ptime=20 and according  
>>> the RFC 3264 it SHOULD send with ptime=20 because of the received  
>>> INVITE message specification, but it DON'T HAVE to send with  
>>> ptime=20.
>>>
>>> At next Freeswitch tries to fix "the issue". In the logfile I found:
>>> e686b430-5d2d-488b-8b58-0fca1965eea7 2010-10-07 15:20:25.673206  
>>> [WARNING] mod_sofia.c:1033 We were told to use ptime 20 but what  
>>> they meant to say was 60
>>> This issue has so far been identified to happen on the following  
>>> broken platforms/devices:
>>> Linksys/Sipura aka Cisco
>>> ShoreTel
>>> Sonus/L3
>>> We will try to fix it but some of the devices on this list are so  
>>> broken,
>>> who knows what will happen..
>>> This log message isn't correct. The callee never specified  
>>> anything about the usage of a specific ptime. Furthermore,  
>>> according RFC 3264  the ptime doesn't specify the frame size,  
>>> which will be used to send packages by the side, which specify it  
>>> in the SDP. In the RFC 3264 is written:
>>>    If the ptime attribute is present for a stream, it indicates the
>>>    desired packetization interval that the offerer would like to
>>>     receive .
>>>
>>>    ...
>>>
>>>    There is now requirement that the packetization interval be the  
>>> same in each direction for a particular stream.
>>>
>>> IMHO that means, that it isn't possible in principle that a device  
>>> is lying about it's ptime usage, because it only specify by the  
>>> media attribute the packetization it likes to receive and doesn't  
>>> specify the packetization it will use itself.
>>>
>>> For fixing "the problem" Freeswitch sends a re-INVITE message at  
>>> 1164,777. This message includes in the message header "X-Broken- 
>>> PTIME: Adv=20; Sent=60", and ptime = 60 media attribute.
>>> The callee fails to process this re-INVITE and drops the call.
>>>
>>> I made the trace after I set the newly introduced parameter  
>>> "passthru_ptime_mismatch=true" (it's documented in the Wiki since  
>>> yesterday). Does it make sense, that Freeswitch tries to fix any  
>>> ptime setting if this variable is set to true?
>>>
>>> If someone wants to examine this issue more detailed, I can  
>>> provide the Wireshark-cap file of the call and the debug output of  
>>> Freeswitch.
>>>
>>>
>>> Thank you in advance
>>>            Jan
>>>
>>>
>>>
>>>
>>>
>>> -- 
>>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>>                                         E-Mail:  riedinger at sns.eu
>>> SNS Consult GmbH                        ICQ   :  163-237-041
>>> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
>>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>>
>>> AG Charlottenburg - HRB 71973
>>> <Ptime Problem Call  
>>> Trace.tif>_______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> -- 
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                         E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>
> AG Charlottenburg - HRB 71973
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From riedinger at sns.eu  Fri Oct  8 04:39:25 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Fri, 08 Oct 2010 13:39:25 +0200
Subject: [Freeswitch-users] Problematic Behaviour of FS regarding ptime
 negotiation
In-Reply-To: <E18C5FFA-F9BD-43F0-9697-1BCC7A7176C3@ipeva.fr>
References: <4CAEF500.1050202@sns.eu>
	<E18C5FFA-F9BD-43F0-9697-1BCC7A7176C3@ipeva.fr>
Message-ID: <4CAF02ED.60508@sns.eu>



Am 08.10.2010 13:06, schrieb David Ponzone:
> Jan,
>
> to answer to 2 others questions you ask:
>
> Why FS tries to enforce 20 ?
> Well, the default is 20ms for most codecs, except perhaps G723, and no 
> explicit ptime means 20ms for most codecs.
> So your carrier is sending no ptime, meaning they want 20.
> FS agrees and send back 20 (and explictly, because smart people always 
> do things explictly and avoid relying on default values/behaviours).
I think this isn't correct. If you work with a codec list in a Cisco and 
set any byte / frame size values for the codecs of the codec list, the 
Cisco doesn't specify any ptime in the initial INVITE message, even if 
for all codecs of the codec list the same frame size is specified. Thus 
it's risky at this point, if FS assums that the caller wants to use 20 ms

>
> So, yes, the message displayed by FS is correct at some point:
> FS asked for 20ms, and your carrier is sending 60ms.
>
But the usage of 60 ms nevertheless, is ok according the RFC.
> Now, I see your point: perhaps the phrase is not very clear.
> I think the issue is (and Anthony or Brian will correct me on this if 
> required) that FS tries to negotiate the same ptime on both 
> directions, because what the RFC says about asymmetrical ptimes is 
> scary, AFAIK. I heard people reporting major issues trying to do this.
I configured for a long time a payload of 40 or 60 bytes on my Ciscos, 
because of the disadvantageous TCP/IP header overhead, if you go with 20 
bytes. I asked my business partners to do it in the same way. However, 
often the didn't change their standard config and continued to use 20 
bytes. I had trouble by this asymmetry only once out of more than 200 
configured interconnects.
> Ok the RFC allows it, but as usual, it was probably badly implemented 
> by most vendors, and anyway, there is no real benefit.
> So FS tries to stay simple.
> I think that's what FS means by "We were told": the other party asked 
> us for 20ms, and as we like to keep things simple, we also asked for 
> 20ms, and they send back 60ms, those p....bast.... :)
>
As you see in the trace graph attached to my previous e-mail, the 
re-INVITE of FFS results in an "internal server error" at the 
terminating GW. Of course this shouldn't be the case and doesn't comply 
with the RFC, but this problem is caused by the efforts of FS to fix a 
problem, which doesn' exist - at least according the RFC.

> Basically, I think what you are asking is a new parameter that would 
> instruct FS to stop trying to re-packetize and accept asymmetrical ptimes.
> About the message, you can get rid of it with 
> rtp-autofix-timing=false, but use it at your own risk.
>
Is it possible to use rtp-autofix-timing just for a specific carrier? If 
I specify it in the default profile, it is used for all carriers. 
Maybe/Probably I'm wrong, but according my current knowledge I have to 
use another non standard IP port, if I want to use another profile just 
for this specific carrier.

BR
     Jan

> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IP eva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 08/10/2010 ? 12:40, Jan Riedinger a ?crit :
>
>> I'm terminating various destination by various carriers. After 
>> migrating one customer to Freeswitch, we observed problems for the 
>> termination of a specific route for a specific carrier. I tried to 
>> examine the problem in detail and I think it's related to problems 
>> regarding the ptime negotiation. I think Freeswitch doesn't breach 
>> any RFC, but I'm not sure, if the behaviour is optimal.
>>
>> The SDP of the Caller INVITE-Message at time 1160,056 in the attached 
>> trace doesn't include any ptime setting. Nevertheless Freeswitch 
>> includes a ptime=20 media attribute in the forwarded INVITE message 
>> at time 1160,065. The ringing SDP sent by the callee at time 1161,948 
>> again doesn't include any ptime setting. Nevertheless, Freeswitch 
>> includes in the Session Progress SDP (at time 1164,240) a ptime=20 
>> media atrribute. Why try Freeswitch to force the usage of ptime=20 
>> for the communication?
>>
>> The OK SDP at time 1164,240 again doesn't contain a ptime media 
>> attribute. Nevertheless, the Freeswitch add a ptime=20 media 
>> attribute forwarded to the caller at 1164,256.
>>
>> It seems that the callee is sending in the following with a frame 
>> size of 60 bytes - it never claimed to use ptime=20 and according the 
>> RFC 3264 it SHOULD send with ptime=20 because of the received INVITE 
>> message specification, but it DON'T HAVE to send with ptime=20.
>>
>> At next Freeswitch tries to fix "the issue". In the logfile I found:
>>
>>     e686b430-5d2d-488b-8b58-0fca1965eea7 2010-10-07 15:20:25.673206
>>     [WARNING] mod_sofia.c:1033 We were told to use ptime 20 but what
>>     they meant to say was 60
>>     This issue has so far been identified to happen on the following
>>     broken platforms/devices:
>>     Linksys/Sipura aka Cisco
>>     ShoreTel
>>     Sonus/L3
>>     We will try to fix it but some of the devices on this list are so
>>     broken,
>>     who knows what will happen..
>>
>> This log message isn't correct. The callee never specified anything 
>> about the usage of a specific ptime. Furthermore, according RFC 3264  
>> the ptime doesn't specify the frame size, which will be used to send 
>> packages by the side, which specify it in the SDP. In the RFC 3264 is 
>> written:
>>     If the ptime attribute is present for a stream, it indicates the
>>     desired packetization interval that the offerer would like to
>>      receive .
>>
>>     ...
>>
>>     There is now requirement that the packetization interval be the same in each direction for a particular stream.
>>
>> IMHO that means, that it isn't possible in principle that a device is 
>> lying about it's ptime usage, because it only specify by the media 
>> attribute the packetization it likes to receive and doesn't specify 
>> the packetization it will use itself.
>>
>> For fixing "the problem" Freeswitch sends a re-INVITE message at 
>> 1164,777. This message includes in the message header 
>> "X-Broken-PTIME: Adv=20; Sent=60", and ptime = 60 media attribute.
>> The callee fails to process this re-INVITE and drops the call.
>>
>> I made the trace after I set the newly introduced parameter 
>> "passthru_ptime_mismatch=true" (it's documented in the Wiki since 
>> yesterday). Does it make sense, that Freeswitch tries to fix any 
>> ptime setting if this variable is set to true?
>>
>> If someone wants to examine this issue more detailed, I can provide 
>> the Wireshark-cap file of the call and the debug output of Freeswitch.
>>
>>
>> Thank you in advance
>>            Jan
>>
>>
>>
>>
>>
>> -- 
>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>                                          E-Mail:riedinger at sns.eu  <mailto:riedinger at sns.eu>
>> SNS Consult GmbH                        ICQ   :  163-237-041
>> S?dwestkorso 49a                        MSN   :jan at sns-consult.de  <mailto:jan at sns-consult.de>
>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>
>> AG Charlottenburg - HRB 71973
>> <Ptime Problem Call 
>> Trace.tif>_______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                         E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973

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From david.ponzone at ipeva.fr  Fri Oct  8 04:57:15 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 8 Oct 2010 13:57:15 +0200
Subject: [Freeswitch-users] Problematic Behaviour of FS regarding ptime
	negotiation
In-Reply-To: <4CAF02ED.60508@sns.eu>
References: <4CAEF500.1050202@sns.eu>
	<E18C5FFA-F9BD-43F0-9697-1BCC7A7176C3@ipeva.fr>
	<4CAF02ED.60508@sns.eu>
Message-ID: <D364FEBB-0B8F-4CD0-AA95-D7BE30E3C24A@ipeva.fr>

I am out of my league, I would need to dig into the RFCs so I prefer  
to wait for comments from the people who wrote the code, because I am  
sure they have an opinion on how the RFCs should be read or why they  
did what they did.

For the autofix-timing thing, I don't think there is a way to change  
that per gateway or with a channel variable.
So yes, you have to use another IP or same IP with non-standard port.
But you just need 2: one profile with it, one without.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 08/10/2010 ? 13:39, Jan Riedinger a ?crit :

>
>
> Am 08.10.2010 13:06, schrieb David Ponzone:
>>
>> Jan,
>>
>> to answer to 2 others questions you ask:
>>
>> Why FS tries to enforce 20 ?
>> Well, the default is 20ms for most codecs, except perhaps G723, and  
>> no explicit ptime means 20ms for most codecs.
>> So your carrier is sending no ptime, meaning they want 20.
>> FS agrees and send back 20 (and explictly, because smart people  
>> always do things explictly and avoid relying on default values/ 
>> behaviours).
> I think this isn't correct. If you work with a codec list in a Cisco  
> and set any byte / frame size values for the codecs of the codec  
> list, the Cisco doesn't specify any ptime in the initial INVITE  
> message, even if for all codecs of the codec list the same frame  
> size is specified. Thus it's risky at this point, if FS assums that  
> the caller wants to use 20 ms
>
>>
>> So, yes, the message displayed by FS is correct at some point:
>> FS asked for 20ms, and your carrier is sending 60ms.
>>
> But the usage of 60 ms nevertheless, is ok according the RFC.
>> Now, I see your point: perhaps the phrase is not very clear.
>> I think the issue is (and Anthony or Brian will correct me on this  
>> if required) that FS tries to negotiate the same ptime on both  
>> directions, because what the RFC says about asymmetrical ptimes is  
>> scary, AFAIK. I heard people reporting major issues trying to do  
>> this.
> I configured for a long time a payload of 40 or 60 bytes on my  
> Ciscos, because of the disadvantageous TCP/IP header overhead, if  
> you go with 20 bytes. I asked my business partners to do it in the  
> same way. However, often the didn't change their standard config and  
> continued to use 20 bytes. I had trouble by this asymmetry only once  
> out of more than 200 configured interconnects.
>> Ok the RFC allows it, but as usual, it was probably badly  
>> implemented by most vendors, and anyway, there is no real benefit.
>> So FS tries to stay simple.
>> I think that's what FS means by "We were told": the other party  
>> asked us for 20ms, and as we like to keep things simple, we also  
>> asked for 20ms, and they send back 60ms, those p....bast.... :)
>>
> As you see in the trace graph attached to my previous e-mail, the re- 
> INVITE of FFS results in an "internal server error" at the  
> terminating GW. Of course this shouldn't be the case and doesn't  
> comply with the RFC, but this problem is caused by the efforts of FS  
> to fix a problem, which doesn' exist - at least according the RFC.
>
>> Basically, I think what you are asking is a new parameter that  
>> would instruct FS to stop trying to re-packetize and accept  
>> asymmetrical ptimes.
>> About the message, you can get rid of it with rtp-autofix- 
>> timing=false, but use it at your own risk.
>>
> Is it possible to use rtp-autofix-timing just for a specific  
> carrier? If I specify it in the default profile, it is used for all  
> carriers. Maybe/Probably I'm wrong, but according my current  
> knowledge I have to use another non standard IP port, if I want to  
> use another profile just for this specific carrier.
>
> BR
>     Jan
>
>> David Ponzone   Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client  IP eva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ? l'intention exclusive de ses destinataires. Toute  
>> utilisation ou diffusion non autoris?e est interdite. Tout message  
>> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
>> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
>> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>>
>> Le 08/10/2010 ? 12:40, Jan Riedinger a ?crit :
>>
>>> I'm terminating various destination by various carriers. After  
>>> migrating one customer to Freeswitch, we observed problems for the  
>>> termination of a specific route for a specific carrier. I tried to  
>>> examine the problem in detail and I think it's related to problems  
>>> regarding the ptime negotiation. I think Freeswitch doesn't breach  
>>> any RFC, but I'm not sure, if the behaviour is optimal.
>>>
>>> The SDP of the Caller INVITE-Message at time 1160,056 in the  
>>> attached trace doesn't include any ptime setting. Nevertheless  
>>> Freeswitch includes a ptime=20 media attribute in the forwarded  
>>> INVITE message at time 1160,065. The ringing SDP sent by the  
>>> callee at time 1161,948 again doesn't include any ptime setting.  
>>> Nevertheless, Freeswitch includes in the Session Progress SDP (at  
>>> time 1164,240) a ptime=20 media atrribute. Why try Freeswitch to  
>>> force the usage of ptime=20 for the communication?
>>>
>>> The OK SDP at time 1164,240 again doesn't contain a ptime media  
>>> attribute. Nevertheless, the Freeswitch add a ptime=20 media  
>>> attribute forwarded to the caller at 1164,256.
>>>
>>> It seems that the callee is sending in the following with a frame  
>>> size of 60 bytes - it never claimed to use ptime=20 and according  
>>> the RFC 3264 it SHOULD send with ptime=20 because of the received  
>>> INVITE message specification, but it DON'T HAVE to send with  
>>> ptime=20.
>>>
>>> At next Freeswitch tries to fix "the issue". In the logfile I found:
>>> e686b430-5d2d-488b-8b58-0fca1965eea7 2010-10-07 15:20:25.673206  
>>> [WARNING] mod_sofia.c:1033 We were told to use ptime 20 but what  
>>> they meant to say was 60
>>> This issue has so far been identified to happen on the following  
>>> broken platforms/devices:
>>> Linksys/Sipura aka Cisco
>>> ShoreTel
>>> Sonus/L3
>>> We will try to fix it but some of the devices on this list are so  
>>> broken,
>>> who knows what will happen..
>>> This log message isn't correct. The callee never specified  
>>> anything about the usage of a specific ptime. Furthermore,  
>>> according RFC 3264  the ptime doesn't specify the frame size,  
>>> which will be used to send packages by the side, which specify it  
>>> in the SDP. In the RFC 3264 is written:
>>>    If the ptime attribute is present for a stream, it indicates the
>>>    desired packetization interval that the offerer would like to
>>>     receive .
>>>
>>>    ...
>>>
>>>    There is now requirement that the packetization interval be the  
>>> same in each direction for a particular stream.
>>>
>>> IMHO that means, that it isn't possible in principle that a device  
>>> is lying about it's ptime usage, because it only specify by the  
>>> media attribute the packetization it likes to receive and doesn't  
>>> specify the packetization it will use itself.
>>>
>>> For fixing "the problem" Freeswitch sends a re-INVITE message at  
>>> 1164,777. This message includes in the message header "X-Broken- 
>>> PTIME: Adv=20; Sent=60", and ptime = 60 media attribute.
>>> The callee fails to process this re-INVITE and drops the call.
>>>
>>> I made the trace after I set the newly introduced parameter  
>>> "passthru_ptime_mismatch=true" (it's documented in the Wiki since  
>>> yesterday). Does it make sense, that Freeswitch tries to fix any  
>>> ptime setting if this variable is set to true?
>>>
>>> If someone wants to examine this issue more detailed, I can  
>>> provide the Wireshark-cap file of the call and the debug output of  
>>> Freeswitch.
>>>
>>>
>>> Thank you in advance
>>>            Jan
>>>
>>>
>>>
>>>
>>>
>>> -- 
>>> Jan Riedinger                           Phone :  +49-30-39 73 19 66
>>> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>>>                                         E-Mail:  riedinger at sns.eu
>>> SNS Consult GmbH                        ICQ   :  163-237-041
>>> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
>>> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>>>
>>> AG Charlottenburg - HRB 71973
>>> <Ptime Problem Call  
>>> Trace.tif>_______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> -- 
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                         E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>
> AG Charlottenburg - HRB 71973
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From steveayre at gmail.com  Fri Oct  8 04:56:58 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 8 Oct 2010 12:56:58 +0100
Subject: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4
In-Reply-To: <AANLkTingeYfPT_b6dG3bKQOVjxYsmBTKhDuLw_3BKGac@mail.gmail.com>
References: <AANLkTingeYfPT_b6dG3bKQOVjxYsmBTKhDuLw_3BKGac@mail.gmail.com>
Message-ID: <AANLkTi=YZsrcSWbrnUrZ6+P_fnXpQVAROCNc+9okF0X9@mail.gmail.com>

1. Check FS is running
2. Check the port you're connecting to matches the one in event_socket.conf.xml
3. Check that event_socket.conf.xml binding to the same IP you're
connecting to (e.g. 0.0.0.0 if you're connecting from a remote
machine, since the default 127.0.0.1 won't work then)
4. Use netstat on the FS server to verify FreeSWITCH is actually
listening on the port you're trying to connect to (I've had a syntax
error in the config file make the module fail to load in the past
which left FS running but with no ESL socket). (If the module fails to
load there'll also be an error in the log file).
5. Check a firewall isn't blocking access to the port

-Steve



On 7 October 2010 01:23, Peter Schrock <peter.schrock at gmail.com> wrote:
> Okay, so I managed to get FS working the other day and I even managed to
> test a call and test my voicemail. All seemed to be working smoothly until,
> because of the rain here, my power went out and I had to reboot my computer.
> I logged in through the terminal, set up FS in background went to fs_cli and
> I get this error message:
> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
> Error]
> I managed to figure out that in the file "fs_cli.c" on line 1206 you find
> the code for displaying this error message. The problem is that I don't know
> why this error message is occurring. Does anyone have any helpful hints as
> to what I should look at to resolve this problem?
> I even tried going to the git tree and make current, but that gave me
> problems that forced me to turn off mod_spandsp and mod_skyopen in the
> modules.conf, which I had running earlier. Any thoughts?
> PeterS
> PS
> I am not sure if this is of any help, but in addition to the error line
> above, it also posted this info:
> Usage: fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x
> command] [profile]
> ??-?,-h --help ? ? ? ? ? ? ? ? ? ?Usage Information
> ??-H, --host=hostname ? ? ? ? ? ? Host to connect
> ??-P, --port=port ? ? ? ? ? ? ? ? Port to connect (1 - 65535)
> ??-u, --user=user at domain ? ? ? ? ?user at domain
> ??-p, --password=password ? ? ? ? Password
> ??-x, --execute=command ? ? ? ? ? Execute Command and Exit
> ??-l, --loglevel=command ? ? ? ? ?Log Level
> ??-q, --quiet ? ? ? ? ? ? ? ? ? ? Disable logging
> ??-r, --retry ? ? ? ? ? ? ? ? ? ? Retry connection on failure
> ??-R, --reconnect ? ? ? ? ? ? ? ? Reconnect if disconnected
> ??-d, --debug=level ? ? ? ? ? ? ? Debug Level (0 - 7)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From fs-list at communicatefreely.net  Fri Oct  8 07:00:52 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 08 Oct 2010 10:00:52 -0400
Subject: [Freeswitch-users] Cannot create any sessions at this time
Message-ID: <4CAF2414.2040900@communicatefreely.net>

Hello,

At some point during the night, something happened to our freeswitch blocking all outgoing calls.
When I looked at the console, I got this message "Cannot create any sessions at this time".  There
were no active channels according to show channels, and for whatever reason, I thought to try fsctl
hupall and then fsctl resume.  That fixed everything, but I have no idea how it got to that state.
Everything was working fine when I went to bed yesterday.

I couldn't find anything on the wiki about the error message or the resume command.

What does this error message mean, (what can put it into this condition), and what did the resume
command do that made it okay again?

Just so I can avoid this in the future when we move our 300 office users over to this box!

Thanks!

-Tim


From anthony.minessale at gmail.com  Fri Oct  8 07:11:22 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Oct 2010 09:11:22 -0500
Subject: [Freeswitch-users] Cannot create any sessions at this time
In-Reply-To: <4CAF2414.2040900@communicatefreely.net>
References: <4CAF2414.2040900@communicatefreely.net>
Message-ID: <AANLkTikYZFDnkXGh1OOyupXqD+7H+s6dy7tz68jbHRfL@mail.gmail.com>

you would need to supply the actual logs.
but it's possible you inadvertently entered fsctl pause or you had the
old config where F10 was mapped to that command.


On Fri, Oct 8, 2010 at 9:00 AM, Tim St. Pierre
<fs-list at communicatefreely.net> wrote:
> Hello,
>
> At some point during the night, something happened to our freeswitch blocking all outgoing calls.
> When I looked at the console, I got this message "Cannot create any sessions at this time". ?There
> were no active channels according to show channels, and for whatever reason, I thought to try fsctl
> hupall and then fsctl resume. ?That fixed everything, but I have no idea how it got to that state.
> Everything was working fine when I went to bed yesterday.
>
> I couldn't find anything on the wiki about the error message or the resume command.
>
> What does this error message mean, (what can put it into this condition), and what did the resume
> command do that made it okay again?
>
> Just so I can avoid this in the future when we move our 300 office users over to this box!
>
> Thanks!
>
> -Tim
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From anthony.minessale at gmail.com  Fri Oct  8 07:18:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Oct 2010 09:18:30 -0500
Subject: [Freeswitch-users] Problematic Behaviour of FS regarding ptime
	negotiation
In-Reply-To: <D364FEBB-0B8F-4CD0-AA95-D7BE30E3C24A@ipeva.fr>
References: <4CAEF500.1050202@sns.eu>
	<E18C5FFA-F9BD-43F0-9697-1BCC7A7176C3@ipeva.fr>
	<4CAF02ED.60508@sns.eu>
	<D364FEBB-0B8F-4CD0-AA95-D7BE30E3C24A@ipeva.fr>
Message-ID: <AANLkTin9A+m6koQu5YnEVJs6bFBUzd1_vX_XhhFKf+Wz@mail.gmail.com>

Everything you described is how we behave.
We will not be changing it.

Not specifying the ptime is a giant performance hit because we cannot
initilize the timers.
Wanting to send 60 and not actually specifying it could also probably
explained away in some deep interpretation of the RFC but it's not
typical and it's plain foolish.

The only advice I can give you is to create a dedicated sip profile
for this call path and configure the codec negotiation to scrooge and
define G729 at 60i in your codec config.

This will make FS use 60ms g729 regardless of what it sees in the sdp,
this is not optimal for anyone but your one case which is why I say to
put it in a specific profile.


On Fri, Oct 8, 2010 at 6:57 AM, David Ponzone <david.ponzone at ipeva.fr> wrote:
> I am out of my league, I would need to dig into the RFCs so I prefer to wait
> for comments from the people who wrote the code, because I am sure they have
> an opinion on how the RFCs should be read or why they did what they did.
> For the autofix-timing thing, I don't think there is a way to change that
> per gateway or with a channel variable.
> So yes, you have to use another IP or same IP with non-standard port.
> But you just need 2: one profile with it, one without.
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 08/10/2010 ? 13:39, Jan Riedinger a ?crit :
>
>
> Am 08.10.2010 13:06, schrieb David Ponzone:
>
> Jan,
> to answer to 2 others questions you ask:
> Why FS tries to enforce 20 ?
> Well, the default is 20ms for most codecs, except perhaps G723, and no
> explicit ptime means 20ms for most codecs.
> So your carrier is sending no ptime, meaning they want 20.
> FS agrees and send back 20 (and explictly, because smart people always do
> things explictly and avoid relying on default values/behaviours).
>
> I think this isn't correct. If you work with a codec list in a Cisco and set
> any byte / frame size values for the codecs of the codec list, the Cisco
> doesn't specify any ptime in the initial INVITE message, even if for all
> codecs of the codec list the same frame size is specified. Thus it's risky
> at this point, if FS assums that the caller wants to use 20 ms
>
>
> So, yes, the message displayed by FS is correct at some point:
> FS asked for 20ms, and your carrier is sending 60ms.
>
> But the usage of 60 ms nevertheless, is ok according the RFC.
>
> Now, I see your point: perhaps the phrase is not very clear.
> I think the issue is (and Anthony or Brian will correct me on this if
> required) that FS tries to negotiate the same ptime on both directions,
> because what the RFC says about asymmetrical ptimes is scary, AFAIK. I heard
> people reporting major issues trying to do this.
>
> I configured for a long time a payload of 40 or 60 bytes on my Ciscos,
> because of the disadvantageous TCP/IP header overhead, if you go with 20
> bytes. I asked my business partners to do it in the same way. However, often
> the didn't change their standard config and continued to use 20 bytes. I had
> trouble by this asymmetry only once out of more than 200 configured
> interconnects.
>
> Ok the RFC allows it, but as usual, it was probably badly implemented by
> most vendors, and anyway, there is no real benefit.
>
> So FS tries to stay simple.
> I think that's what FS means by "We were told": the other party asked us for
> 20ms, and as we like to keep things simple, we also asked for 20ms, and they
> send back 60ms, those p....bast.... :)
>
> As you see in the trace graph attached to my previous e-mail, the re-INVITE
> of FFS results in an "internal server error" at the terminating GW. Of
> course this shouldn't be the case and doesn't comply with the RFC, but this
> problem is caused by the efforts of FS to fix a problem, which doesn' exist
> - at least according the RFC.
>
> Basically, I think what you are asking is a new parameter that would
> instruct FS to stop trying to re-packetize and accept asymmetrical ptimes.
> About the message, you can get rid of it with rtp-autofix-timing=false, but
> use it at your own risk.
>
> Is it possible to use rtp-autofix-timing just for a specific carrier? If I
> specify it in the default profile, it is used for all carriers.
> Maybe/Probably I'm wrong, but according my current knowledge I have to use
> another non standard IP port, if I want to use another profile just for this
> specific carrier.
>
> BR
> ? ? Jan
>
> David Ponzone ? Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client? IP eva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 08/10/2010 ? 12:40, Jan Riedinger a ?crit :
>
> I'm terminating various destination by various carriers. After migrating one
> customer to Freeswitch, we observed problems for the termination of a
> specific route for a specific carrier. I tried to examine the problem in
> detail and I think it's related to problems regarding the ptime negotiation.
> I think Freeswitch doesn't breach any RFC, but I'm not sure, if the
> behaviour is optimal.
>
> The SDP of the Caller INVITE-Message at time 1160,056 in the attached trace
> doesn't include any ptime setting. Nevertheless Freeswitch includes a
> ptime=20 media attribute in the forwarded INVITE message at time 1160,065.
> The ringing SDP sent by the callee at time 1161,948 again doesn't include
> any ptime setting. Nevertheless, Freeswitch includes in the Session Progress
> SDP (at time 1164,240) a ptime=20 media atrribute. Why try Freeswitch to
> force the usage of ptime=20 for the communication?
>
> The OK SDP at time 1164,240 again doesn't contain a ptime media attribute.
> Nevertheless, the Freeswitch add a ptime=20 media attribute forwarded to the
> caller at 1164,256.
>
> It seems that the callee is sending in the following with a frame size of 60
> bytes - it never claimed to use ptime=20 and according the RFC 3264 it
> SHOULD send with ptime=20 because of the received INVITE message
> specification, but it DON'T HAVE to send with ptime=20.
>
> At next Freeswitch tries to fix "the issue". In the logfile I found:
>
> e686b430-5d2d-488b-8b58-0fca1965eea7 2010-10-07 15:20:25.673206 [WARNING]
> mod_sofia.c:1033 We were told to use ptime 20 but what they meant to say was
> 60
> This issue has so far been identified to happen on the following broken
> platforms/devices:
> Linksys/Sipura aka Cisco
> ShoreTel
> Sonus/L3
> We will try to fix it but some of the devices on this list are so broken,
> who knows what will happen..
>
> This log message isn't correct. The callee never specified anything about
> the usage of a specific ptime. Furthermore, according RFC 3264? the ptime
> doesn't specify the frame size, which will be used to send packages by the
> side, which specify it in the SDP. In the RFC 3264 is written:
>
>    If the ptime attribute is present for a stream, it indicates the
>    desired packetization interval that the offerer would like to
>     receive .
>
>    ...
>
>    There is now requirement that the packetization interval be the same in
> each direction for a particular stream.
>
> IMHO that means, that it isn't possible in principle that a device is lying
> about it's ptime usage, because it only specify by the media attribute the
> packetization it likes to receive and doesn't specify the packetization it
> will use itself.
>
> For fixing "the problem" Freeswitch sends a re-INVITE message at 1164,777.
> This message includes in the message header "X-Broken-PTIME: Adv=20;
> Sent=60", and ptime = 60 media attribute.
> The callee fails to process this re-INVITE and drops the call.
>
> I made the trace after I set the newly introduced parameter
> "passthru_ptime_mismatch=true" (it's documented in the Wiki since
> yesterday). Does it make sense, that Freeswitch tries to fix any ptime
> setting if this variable is set to true?
>
> If someone wants to examine this issue more detailed, I can provide the
> Wireshark-cap file of the call and the debug output of Freeswitch.
>
>
> Thank you in advance
> ?????????? Jan
>
>
>
>
>
> --
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                         E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>
> AG Charlottenburg - HRB 71973
>
> <Ptime Problem Call
> Trace.tif>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> --
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                         E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>
> AG Charlottenburg - HRB 71973
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From covici at ccs.covici.com  Fri Oct  8 07:32:30 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 08 Oct 2010 10:32:30 -0400
Subject: [Freeswitch-users] segfault in swift library
Message-ID: <17117.1286548350@ccs.covici.com>

I have been trying to get mod_cepstral to work and so far, no joy at
all.  At first I was using the version 5 voice, but was told on the IRC
or maybe in the onference that I should go back to version 4.  So I
restore the directory, recompiled FSW and now I see in the logs the
following message:
Oct  7 06:16:15 ccs kernel: freeswitch[10398]: segfault at 627769 ip
b5ffda43 sp bfd48ce4 error 4 in libswift.so.4.2[b5fcf000+7e000]

No core was dumped -- can someone tell me what is going on, or how to
troubleshoot this?

Thanks in advance for any suggestions.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From lists at infosecurity.ch  Fri Oct  8 07:45:18 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Fri, 08 Oct 2010 16:45:18 +0200
Subject: [Freeswitch-users] bypass_media only if both peers have public IP:
	how to do it?
Message-ID: <4CAF2E7E.9060709@infosecurity.ch>

Hi,

i would like to configure a FS system so that only if both peers have a
public IP address they can speak each other.

The condition is that the client are mobile and can connect from
whatever network, so apply-nat-acl does not work because some mobile
carrier use fancy internal IP address such as 1.1.1.1 .

All clients connect with SIP/TLS and there is no SIP ALG that mangle the
SDP, so we can be sure that what FS see is exactly what the SIP client
has sent.

Given that, i would like FS to always do proxy_media unless BOTH two leg
of the call have a public IP address.

I know that's not extremely efficient and that in theory two SIP clients
with one behind NAT and one with public IP can speak each other, but
even in such conditions there may be unpredictable issues.

How the NAT handling of FS could be modified to do that and let
bypass_media only when both SIP client have a public IP address?

Fabio


From brian at freeswitch.org  Fri Oct  8 07:45:51 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 Oct 2010 09:45:51 -0500
Subject: [Freeswitch-users] Dead lua scripts
In-Reply-To: <4CAE7EB0.2070600@communicatefreely.net>
References: <4CAE15F7.9090807@communicatefreely.net>
	<AANLkTinKznsWB6Zzf3cexnqHjesgQ_gsK++KkKD_h+U0@mail.gmail.com>
	<4CAE7EB0.2070600@communicatefreely.net>
Message-ID: <F5C4BC8D-E276-4125-AAB3-655715875EFB@freeswitch.org>

Well anthony gave you a hint... wrap your whole loop in a while(session:ready) it will then end the script if the session hangs up.

/b

On Oct 7, 2010, at 9:15 PM, Tim St. Pierre wrote:

> I had to completely restart freeswitch to make these calls go away.  Is there a less disruptive way
> to terminate the scripts?



From riedinger at sns.eu  Fri Oct  8 08:03:56 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Fri, 08 Oct 2010 17:03:56 +0200
Subject: [Freeswitch-users] Problematic Behaviour of FS regarding ptime
 negotiation
In-Reply-To: <AANLkTin9A+m6koQu5YnEVJs6bFBUzd1_vX_XhhFKf+Wz@mail.gmail.com>
References: <4CAEF500.1050202@sns.eu>	<E18C5FFA-F9BD-43F0-9697-1BCC7A7176C3@ipeva.fr>	<4CAF02ED.60508@sns.eu>	<D364FEBB-0B8F-4CD0-AA95-D7BE30E3C24A@ipeva.fr>
	<AANLkTin9A+m6koQu5YnEVJs6bFBUzd1_vX_XhhFKf+Wz@mail.gmail.com>
Message-ID: <4CAF32DC.3040005@sns.eu>

  Hi Anthony,

you are writing: "Wanting to send 60 and not actually specifying it..."

According my interpretation of the RFC it's just not possible for 
gateway to specify the preferred ptime for SENDING. The ptime specifies 
the preferred frame size for RECEIVING.

In the RFC 3264 is written:

    If the ptime attribute is present for a stream, it indicates the
    desired packetization interval that the offerer would like to
    *RECEIVE*.


Indications that the RFC is to be interpreted in this way can be found 
under

http://www.cisco.com/en/US/docs/ios/12_3/sip/configuration/guide/chapter8.html#wp1064009

If you study the examples of this web site for asymmetric SDP you will 
find, that the ptime, which was requested by GW A is used from GW B for 
sending packets, whereas GW A itself uses the frame size for sending, 
which was requested by GW B!

BR
     Jan






Am 08.10.2010 16:18, schrieb Anthony Minessale:
> Everything you described is how we behave.
> We will not be changing it.
>
> Not specifying the ptime is a giant performance hit because we cannot
> initilize the timers.
> Wanting to send 60 and not actually specifying it could also probably
> explained away in some deep interpretation of the RFC but it's not
> typical and it's plain foolish.
>
> The only advice I can give you is to create a dedicated sip profile
> for this call path and configure the codec negotiation to scrooge and
> define G729 at 60i in your codec config.
>
> This will make FS use 60ms g729 regardless of what it sees in the sdp,
> this is not optimal for anyone but your one case which is why I say to
> put it in a specific profile.
>
>
> On Fri, Oct 8, 2010 at 6:57 AM, David Ponzone <david.ponzone at ipeva.fr> 
> wrote:
>> I am out of my league, I would need to dig into the RFCs so I prefer 
>> to wait
>> for comments from the people who wrote the code, because I am sure 
>> they have
>> an opinion on how the RFCs should be read or why they did what they did.
>> For the autofix-timing thing, I don't think there is a way to change that
>> per gateway or with a channel variable.
>> So yes, you have to use another IP or same IP with non-standard port.
>> But you just need 2: one profile with it, one without.
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou 
>> diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce 
>> message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>> Le 08/10/2010 ? 13:39, Jan Riedinger a ?crit :
>>
>>
>> Am 08.10.2010 13:06, schrieb David Ponzone:
>>
>> Jan,
>> to answer to 2 others questions you ask:
>> Why FS tries to enforce 20 ?
>> Well, the default is 20ms for most codecs, except perhaps G723, and no
>> explicit ptime means 20ms for most codecs.
>> So your carrier is sending no ptime, meaning they want 20.
>> FS agrees and send back 20 (and explictly, because smart people always do
>> things explictly and avoid relying on default values/behaviours).
>>
>> I think this isn't correct. If you work with a codec list in a Cisco 
>> and set
>> any byte / frame size values for the codecs of the codec list, the Cisco
>> doesn't specify any ptime in the initial INVITE message, even if for all
>> codecs of the codec list the same frame size is specified. Thus it's 
>> risky
>> at this point, if FS assums that the caller wants to use 20 ms
>>
>>
>> So, yes, the message displayed by FS is correct at some point:
>> FS asked for 20ms, and your carrier is sending 60ms.
>>
>> But the usage of 60 ms nevertheless, is ok according the RFC.
>>
>> Now, I see your point: perhaps the phrase is not very clear.
>> I think the issue is (and Anthony or Brian will correct me on this if
>> required) that FS tries to negotiate the same ptime on both directions,
>> because what the RFC says about asymmetrical ptimes is scary, AFAIK. 
>> I heard
>> people reporting major issues trying to do this.
>>
>> I configured for a long time a payload of 40 or 60 bytes on my Ciscos,
>> because of the disadvantageous TCP/IP header overhead, if you go with 20
>> bytes. I asked my business partners to do it in the same way. 
>> However, often
>> the didn't change their standard config and continued to use 20 
>> bytes. I had
>> trouble by this asymmetry only once out of more than 200 configured
>> interconnects.
>>
>> Ok the RFC allows it, but as usual, it was probably badly implemented by
>> most vendors, and anyway, there is no real benefit.
>>
>> So FS tries to stay simple.
>> I think that's what FS means by "We were told": the other party asked 
>> us for
>> 20ms, and as we like to keep things simple, we also asked for 20ms, 
>> and they
>> send back 60ms, those p....bast.... :)
>>
>> As you see in the trace graph attached to my previous e-mail, the 
>> re-INVITE
>> of FFS results in an "internal server error" at the terminating GW. Of
>> course this shouldn't be the case and doesn't comply with the RFC, 
>> but this
>> problem is caused by the efforts of FS to fix a problem, which doesn' 
>> exist
>> - at least according the RFC.
>>
>> Basically, I think what you are asking is a new parameter that would
>> instruct FS to stop trying to re-packetize and accept asymmetrical 
>> ptimes.
>> About the message, you can get rid of it with 
>> rtp-autofix-timing=false, but
>> use it at your own risk.
>>
>> Is it possible to use rtp-autofix-timing just for a specific carrier? 
>> If I
>> specify it in the default profile, it is used for all carriers.
>> Maybe/Probably I'm wrong, but according my current knowledge I have 
>> to use
>> another non standard IP port, if I want to use another profile just 
>> for this
>> specific carrier.
>>
>> BR
>>     Jan
>>
>> David Ponzone   Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client  IP eva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou 
>> diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce 
>> message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>> Le 08/10/2010 ? 12:40, Jan Riedinger a ?crit :
>>
>> I'm terminating various destination by various carriers. After 
>> migrating one
>> customer to Freeswitch, we observed problems for the termination of a
>> specific route for a specific carrier. I tried to examine the problem in
>> detail and I think it's related to problems regarding the ptime 
>> negotiation.
>> I think Freeswitch doesn't breach any RFC, but I'm not sure, if the
>> behaviour is optimal.
>>
>> The SDP of the Caller INVITE-Message at time 1160,056 in the attached 
>> trace
>> doesn't include any ptime setting. Nevertheless Freeswitch includes a
>> ptime=20 media attribute in the forwarded INVITE message at time 
>> 1160,065.
>> The ringing SDP sent by the callee at time 1161,948 again doesn't include
>> any ptime setting. Nevertheless, Freeswitch includes in the Session 
>> Progress
>> SDP (at time 1164,240) a ptime=20 media atrribute. Why try Freeswitch to
>> force the usage of ptime=20 for the communication?
>>
>> The OK SDP at time 1164,240 again doesn't contain a ptime media 
>> attribute.
>> Nevertheless, the Freeswitch add a ptime=20 media attribute forwarded 
>> to the
>> caller at 1164,256.
>>
>> It seems that the callee is sending in the following with a frame 
>> size of 60
>> bytes - it never claimed to use ptime=20 and according the RFC 3264 it
>> SHOULD send with ptime=20 because of the received INVITE message
>> specification, but it DON'T HAVE to send with ptime=20.
>>
>> At next Freeswitch tries to fix "the issue". In the logfile I found:
>>
>> e686b430-5d2d-488b-8b58-0fca1965eea7 2010-10-07 15:20:25.673206 [WARNING]
>> mod_sofia.c:1033 We were told to use ptime 20 but what they meant to 
>> say was
>> 60
>> This issue has so far been identified to happen on the following broken
>> platforms/devices:
>> Linksys/Sipura aka Cisco
>> ShoreTel
>> Sonus/L3
>> We will try to fix it but some of the devices on this list are so broken,
>> who knows what will happen..
>>
>> This log message isn't correct. The callee never specified anything about
>> the usage of a specific ptime. Furthermore, according RFC 3264  the ptime
>> doesn't specify the frame size, which will be used to send packages 
>> by the
>> side, which specify it in the SDP. In the RFC 3264 is written:
>>
>> If the ptime attribute is present for a stream, it indicates the
>> desired packetization interval that the offerer would like to
>> receive .
>>
>> ...
>>
>> There is now requirement that the packetization interval be the same in
>> each direction for a particular stream.
>>
>> IMHO that means, that it isn't possible in principle that a device is 
>> lying
>> about it's ptime usage, because it only specify by the media 
>> attribute the
>> packetization it likes to receive and doesn't specify the 
>> packetization it
>> will use itself.
>>
>> For fixing "the problem" Freeswitch sends a re-INVITE message at 
>> 1164,777.
>> This message includes in the message header "X-Broken-PTIME: Adv=20;
>> Sent=60", and ptime = 60 media attribute.
>> The callee fails to process this re-INVITE and drops the call.
>>
>> I made the trace after I set the newly introduced parameter
>> "passthru_ptime_mismatch=true" (it's documented in the Wiki since
>> yesterday). Does it make sense, that Freeswitch tries to fix any ptime
>> setting if this variable is set to true?
>>
>> If someone wants to examine this issue more detailed, I can provide the
>> Wireshark-cap file of the call and the debug output of Freeswitch.
>>
>>
>> Thank you in advance
>>            Jan
>>
>>
>>
>>
>>
>> --
>> Jan Riedinger Phone : +49-30-39 73 19 66
>> Dipl.-Inf. | Managing Director Fax : +49-30-39 73 19 64
>> E-Mail: riedinger at sns.eu
>> SNS Consult GmbH ICQ : 163-237-041
>> S?dwestkorso 49a MSN : jan at sns-consult.de
>> 14197 Berlin GERMANY Skype : Jan Riedinger
>>
>> AG Charlottenburg - HRB 71973
>>
>> <Ptime Problem Call
>> Trace.tif>_______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> --
>> Jan Riedinger Phone : +49-30-39 73 19 66
>> Dipl.-Inf. | Managing Director Fax : +49-30-39 73 19 64
>> E-Mail: riedinger at sns.eu
>> SNS Consult GmbH ICQ : 163-237-041
>> S?dwestkorso 49a MSN : jan at sns-consult.de
>> 14197 Berlin GERMANY Skype : Jan Riedinger
>>
>> AG Charlottenburg - HRB 71973
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>

-- 
Jan Riedinger Phone : +49-30-39 73 19 66
Dipl.-Inf. | Managing Director Fax : +49-30-39 73 19 64
E-Mail: riedinger at sns.eu
SNS Consult GmbH ICQ : 163-237-041
S?dwestkorso 49a MSN : jan at sns-consult.de
14197 Berlin GERMANY Skype : Jan Riedinger

AG Charlottenburg - HRB 71973
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From anthony.minessale at gmail.com  Fri Oct  8 08:14:22 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Oct 2010 10:14:22 -0500
Subject: [Freeswitch-users] Problematic Behaviour of FS regarding ptime
	negotiation
In-Reply-To: <4CAF32DC.3040005@sns.eu>
References: <4CAEF500.1050202@sns.eu>
	<E18C5FFA-F9BD-43F0-9697-1BCC7A7176C3@ipeva.fr>
	<4CAF02ED.60508@sns.eu>
	<D364FEBB-0B8F-4CD0-AA95-D7BE30E3C24A@ipeva.fr>
	<AANLkTin9A+m6koQu5YnEVJs6bFBUzd1_vX_XhhFKf+Wz@mail.gmail.com>
	<4CAF32DC.3040005@sns.eu>
Message-ID: <AANLkTim21cO0ua2e6-4cx0rX7MQ8E0eawM4OSOW2z6wH@mail.gmail.com>

I already stated my position and supplied you with a solution.
I am on to other issues now.


On Fri, Oct 8, 2010 at 10:03 AM, Jan Riedinger <riedinger at sns.eu> wrote:
> Hi Anthony,
>
> you are writing: "Wanting to send 60 and not actually specifying it..."
>
> According my interpretation of the RFC it's just not possible for gateway to
> specify the preferred ptime for SENDING. The ptime specifies the preferred
> frame size for RECEIVING.
>
> In the RFC 3264 is written:
>
> If the ptime attribute is present for a stream, it indicates the
> desired packetization interval that the offerer would like to
> RECEIVE.
>
> Indications that the RFC is to be interpreted in this way can be found under
>
> http://www.cisco.com/en/US/docs/ios/12_3/sip/configuration/guide/chapter8.html#wp1064009
>
> If you study the examples of this web site for asymmetric SDP you will find,
> that the ptime, which was requested by GW A is used from GW B for sending
> packets, whereas GW A itself uses the frame size for sending, which was
> requested by GW B!
>
> BR
> ??? Jan
>
>
>
>
>
>
> Am 08.10.2010 16:18, schrieb Anthony Minessale:
>
> Everything you described is how we behave.
> We will not be changing it.
>
> Not specifying the ptime is a giant performance hit because we cannot
> initilize the timers.
> Wanting to send 60 and not actually specifying it could also probably
> explained away in some deep interpretation of the RFC but it's not
> typical and it's plain foolish.
>
> The only advice I can give you is to create a dedicated sip profile
> for this call path and configure the codec negotiation to scrooge and
> define G729 at 60i in your codec config.
>
> This will make FS use 60ms g729 regardless of what it sees in the sdp,
> this is not optimal for anyone but your one case which is why I say to
> put it in a specific profile.
>
>
> On Fri, Oct 8, 2010 at 6:57 AM, David Ponzone <david.ponzone at ipeva.fr>
> wrote:
>
> I am out of my league, I would need to dig into the RFCs so I prefer to wait
> for comments from the people who wrote the code, because I am sure they have
> an opinion on how the RFCs should be read or why they did what they did.
> For the autofix-timing thing, I don't think there is a way to change that
> per gateway or with a channel variable.
> So yes, you have to use another IP or same IP with non-standard port.
> But you just need 2: one profile with it, one without.
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 08/10/2010 ? 13:39, Jan Riedinger a ?crit :
>
>
> Am 08.10.2010 13:06, schrieb David Ponzone:
>
> Jan,
> to answer to 2 others questions you ask:
> Why FS tries to enforce 20 ?
> Well, the default is 20ms for most codecs, except perhaps G723, and no
> explicit ptime means 20ms for most codecs.
> So your carrier is sending no ptime, meaning they want 20.
> FS agrees and send back 20 (and explictly, because smart people always do
> things explictly and avoid relying on default values/behaviours).
>
> I think this isn't correct. If you work with a codec list in a Cisco and set
> any byte / frame size values for the codecs of the codec list, the Cisco
> doesn't specify any ptime in the initial INVITE message, even if for all
> codecs of the codec list the same frame size is specified. Thus it's risky
> at this point, if FS assums that the caller wants to use 20 ms
>
>
> So, yes, the message displayed by FS is correct at some point:
> FS asked for 20ms, and your carrier is sending 60ms.
>
> But the usage of 60 ms nevertheless, is ok according the RFC.
>
> Now, I see your point: perhaps the phrase is not very clear.
> I think the issue is (and Anthony or Brian will correct me on this if
> required) that FS tries to negotiate the same ptime on both directions,
> because what the RFC says about asymmetrical ptimes is scary, AFAIK. I heard
> people reporting major issues trying to do this.
>
> I configured for a long time a payload of 40 or 60 bytes on my Ciscos,
> because of the disadvantageous TCP/IP header overhead, if you go with 20
> bytes. I asked my business partners to do it in the same way. However, often
> the didn't change their standard config and continued to use 20 bytes. I had
> trouble by this asymmetry only once out of more than 200 configured
> interconnects.
>
> Ok the RFC allows it, but as usual, it was probably badly implemented by
> most vendors, and anyway, there is no real benefit.
>
> So FS tries to stay simple.
> I think that's what FS means by "We were told": the other party asked us for
> 20ms, and as we like to keep things simple, we also asked for 20ms, and they
> send back 60ms, those p....bast.... :)
>
> As you see in the trace graph attached to my previous e-mail, the re-INVITE
> of FFS results in an "internal server error" at the terminating GW. Of
> course this shouldn't be the case and doesn't comply with the RFC, but this
> problem is caused by the efforts of FS to fix a problem, which doesn' exist
> - at least according the RFC.
>
> Basically, I think what you are asking is a new parameter that would
> instruct FS to stop trying to re-packetize and accept asymmetrical ptimes.
> About the message, you can get rid of it with rtp-autofix-timing=false, but
> use it at your own risk.
>
> Is it possible to use rtp-autofix-timing just for a specific carrier? If I
> specify it in the default profile, it is used for all carriers.
> Maybe/Probably I'm wrong, but according my current knowledge I have to use
> another non standard IP port, if I want to use another profile just for this
> specific carrier.
>
> BR
> ? ? Jan
>
> David Ponzone ? Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client? IP eva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 08/10/2010 ? 12:40, Jan Riedinger a ?crit :
>
> I'm terminating various destination by various carriers. After migrating one
> customer to Freeswitch, we observed problems for the termination of a
> specific route for a specific carrier. I tried to examine the problem in
> detail and I think it's related to problems regarding the ptime negotiation.
> I think Freeswitch doesn't breach any RFC, but I'm not sure, if the
> behaviour is optimal.
>
> The SDP of the Caller INVITE-Message at time 1160,056 in the attached trace
> doesn't include any ptime setting. Nevertheless Freeswitch includes a
> ptime=20 media attribute in the forwarded INVITE message at time 1160,065.
> The ringing SDP sent by the callee at time 1161,948 again doesn't include
> any ptime setting. Nevertheless, Freeswitch includes in the Session Progress
> SDP (at time 1164,240) a ptime=20 media atrribute. Why try Freeswitch to
> force the usage of ptime=20 for the communication?
>
> The OK SDP at time 1164,240 again doesn't contain a ptime media attribute.
> Nevertheless, the Freeswitch add a ptime=20 media attribute forwarded to the
> caller at 1164,256.
>
> It seems that the callee is sending in the following with a frame size of 60
> bytes - it never claimed to use ptime=20 and according the RFC 3264 it
> SHOULD send with ptime=20 because of the received INVITE message
> specification, but it DON'T HAVE to send with ptime=20.
>
> At next Freeswitch tries to fix "the issue". In the logfile I found:
>
> e686b430-5d2d-488b-8b58-0fca1965eea7 2010-10-07 15:20:25.673206 [WARNING]
> mod_sofia.c:1033 We were told to use ptime 20 but what they meant to say was
> 60
> This issue has so far been identified to happen on the following broken
> platforms/devices:
> Linksys/Sipura aka Cisco
> ShoreTel
> Sonus/L3
> We will try to fix it but some of the devices on this list are so broken,
> who knows what will happen..
>
> This log message isn't correct. The callee never specified anything about
> the usage of a specific ptime. Furthermore, according RFC 3264? the ptime
> doesn't specify the frame size, which will be used to send packages by the
> side, which specify it in the SDP. In the RFC 3264 is written:
>
> If the ptime attribute is present for a stream, it indicates the
> desired packetization interval that the offerer would like to
> receive .
>
> ...
>
> There is now requirement that the packetization interval be the same in
> each direction for a particular stream.
>
> IMHO that means, that it isn't possible in principle that a device is lying
> about it's ptime usage, because it only specify by the media attribute the
> packetization it likes to receive and doesn't specify the packetization it
> will use itself.
>
> For fixing "the problem" Freeswitch sends a re-INVITE message at 1164,777.
> This message includes in the message header "X-Broken-PTIME: Adv=20;
> Sent=60", and ptime = 60 media attribute.
> The callee fails to process this re-INVITE and drops the call.
>
> I made the trace after I set the newly introduced parameter
> "passthru_ptime_mismatch=true" (it's documented in the Wiki since
> yesterday). Does it make sense, that Freeswitch tries to fix any ptime
> setting if this variable is set to true?
>
> If someone wants to examine this issue more detailed, I can provide the
> Wireshark-cap file of the call and the debug output of Freeswitch.
>
>
> Thank you in advance
> ?????????? Jan
>
>
>
>
>
> --
> Jan Riedinger Phone : +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director Fax : +49-30-39 73 19 64
> E-Mail: riedinger at sns.eu
> SNS Consult GmbH ICQ : 163-237-041
> S?dwestkorso 49a MSN : jan at sns-consult.de
> 14197 Berlin GERMANY Skype : Jan Riedinger
>
> AG Charlottenburg - HRB 71973
>
> <Ptime Problem Call
> Trace.tif>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> --
> Jan Riedinger Phone : +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director Fax : +49-30-39 73 19 64
> E-Mail: riedinger at sns.eu
> SNS Consult GmbH ICQ : 163-237-041
> S?dwestkorso 49a MSN : jan at sns-consult.de
> 14197 Berlin GERMANY Skype : Jan Riedinger
>
> AG Charlottenburg - HRB 71973
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
>
> --
> Jan Riedinger Phone : +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director Fax : +49-30-39 73 19 64
> E-Mail: riedinger at sns.eu
> SNS Consult GmbH ICQ : 163-237-041
> S?dwestkorso 49a MSN : jan at sns-consult.de
> 14197 Berlin GERMANY Skype : Jan Riedinger
>
> AG Charlottenburg - HRB 71973
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From mario_fs at mgtech.com  Fri Oct  8 08:39:04 2010
From: mario_fs at mgtech.com (Mario)
Date: Fri, 08 Oct 2010 08:39:04 -0700
Subject: [Freeswitch-users] FS starts re-registering every minute after a
	few hours
Message-ID: <4CAF3B18.10206@mgtech.com>

I really could use some help on this one, this is one of the last major
issues before I switch to FS. I have expire-seconds set at 600, this
works for several hours. Then, for no apparent reason it changes to 60
(see trace). Most calls are missed due to this issue. Why would FS
change it and how can I keep it at 600? BTW retry is at 30. I spent
lot's of time with traces/investigating and the ITSP watched it and says
they are not changing anything. Thanks! Mario

PS If it's them how do I prove it?

send 865 bytes to udp/[111.222.1.36]:5060 at 15:07:22.220308:


   ------------------------------------------------------------------------
   REGISTER sip:sip.itsp1.com SIP/2.0
   Via: SIP/2.0/UDP 100.101.102.103:5080;rport;branch=hideBcm183mmt2UD
   Max-Forwards: 70
   From: <sip:uuid1 at sip.itsp1.com;transport=udp>;tag=hide37BeUD
   To: <sip:uuid1 at sip.itsp1.com;transport=udp>
   Call-ID: hide-a069-49d0-b342-hide34d28
   CSeq: 2914438 REGISTER
   Contact: <sip:gw+uuid1 at 100.101.102.103:5080;transport=udp;gw=uuid1>
   Expires: 60
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f499c0c 2010-10-07
10-27-31 -0500
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Proxy-Authorization: Digest username="uuid1", realm="xxxtel.com",
nonce="hide0000123270bed298a5fee464c4bb367750hide", algorithm=MD5,
uri="sip:sip.itsp1.com", response="70000a710c1b7a363899922e6hide"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 567 bytes from udp/[111.222.1.36]:5060 at 15:07:22.321019:
   ------------------------------------------------------------------------
   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/UDP
100.101.102.103:5080;rport=5080;branch=hideBcm183mmt2UD;received=111.222.333.127
   From: <sip:uuid1 at sip.itsp1.com;transport=udp>;tag=hide37BeUD
   To:
<sip:uuid1 at sip.itsp1.com;transport=udp>;tag=9214944d68c027e4c99a14a397962871.4d99
   Call-ID: hide-a069-49d0-b342-hide34d28
   CSeq: 2914438 REGISTER
   Proxy-Authenticate: Digest realm="xxxtel.com",
nonce="4caf33ab99012dadc6e7c84040e1b4ff8841a343edc5b0c3", stale=true
   Server: OpenSIPS (1.6.3-notls (i386/freebsd))
   Content-Length: 0

   ------------------------------------------------------------------------
send 865 bytes to udp/[111.222.1.36]:5060 at 15:07:22.321391:
   ------------------------------------------------------------------------
   REGISTER sip:sip.itsp1.com SIP/2.0
   Via: SIP/2.0/UDP 100.101.102.103:5080;rport;branch=z9hG4b99m5c33mrH3reS
   Max-Forwards: 70
   From: <sip:uuid1 at sip.itsp1.com;transport=udp>;tag=hide37BeUD
   To: <sip:uuid1 at sip.itsp1.com;transport=udp>
   Call-ID: hide-a069-49d0-b342-hide34d28
   CSeq: 2914439 REGISTER
   Contact: <sip:gw+uuid1 at 100.101.102.103:5080;transport=udp;gw=uuid1>
   Expires: 60
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f499c0c 2010-10-07
10-27-31 -0500
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Proxy-Authorization: Digest username="uuid1", realm="xxxtel.com",
nonce="4caf33ab00012dadc6e7c89940e1b4ff9941a343edc5b0c3", algorithm=MD5,
uri="sip:sip.itsp1.com", response="b8597203297d6db518b9996f55d3dc36"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 503 bytes from udp/[111.222.1.36]:5060 at 15:07:22.421206:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
100.101.102.103:5080;rport=5080;branch=z9hG4bKKm5c999rH3reS;received=111.222.333.127
   From: <sip:uuid1 at sip.itsp1.com;transport=udp>;tag=hide37BeUD
   To:
<sip:uuid1 at sip.itsp1.com;transport=udp>;tag=921495599c027e4c99a14a397962871.9262
   Call-ID: hide-a069-49d0-b342-hide34d28
   CSeq: 2914439 REGISTER
   Contact:
<sip:gw+uuid1 at 111.222.333.127:5080;transport=udp;gw=uuid1>;expires=60
   Server: OpenSIPS (1.6.3-notls (i386/freebsd))
   Content-Length: 0



From shamun.toha at gmail.com  Fri Oct  8 09:44:20 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Fri, 8 Oct 2010 18:44:20 +0200
Subject: [Freeswitch-users] Freeswitch - mod_spidermonkey_curl bug?
Message-ID: <AANLkTikoViwpeHePu_2oodByfXpe+9rz1=9yLHzGxz7G@mail.gmail.com>

Hello,

I have a very strange problem using mod_spidermonkey_curl in CentOS 5.5
x86_64 bit. Same setup i am using in other machine in Fedora 12, it works,
is there any bugs????????? I receive OK, but when i use to compare it fails
in logical statement.


FreeSWITCH Version 1.0.head (git-25ddef5 2010-09-24 01-55-54 +0200) Started.

Installation curl as:
==============
[root at a autoload_configs]# cat spidermonkey.conf.xml
<configuration name="spidermonkey.conf" description="Spider Monkey
JavaScript Plug-Ins">
  <modules>
    <load module="mod_spidermonkey_teletone"/>
    <load module="mod_spidermonkey_core_db"/>
    <load module="mod_spidermonkey_socket"/>
    <!--<load module="mod_spidermonkey_odbc"/>-->
*    <load module="mod_spidermonkey_curl" />*

  </modules>
</configuration>



.../scripts/helloworld.js:
============

use("CURL");
var menuselection = "";
var promptfordigits_dtmf_digits = "";
var invalidRetry = 0;

/* This is getting failed, in Centos, but in other machine it works */
function my_callback(string, arg)
{
  console_log("info", string);

  //string = 'OK'; // then it works!!
  //2010-10-08 18:40:53.585676 [INFO] mod_dialplan_xml.c:331 Processing 1002
<1002>->1200 in context default
 * //2010-10-08 18:41:23.842965 [INFO] helloworld.js:42 OK*
  //2010-10-08 18:41:35.962099 [NOTICE] switch_ivr.c:1480 Transfer
sofia/internal/1002 at z.com to XML[3000 at default]
  //2010-10-08 18:41:35.962099 [INFO] mod_dialplan_xml.c:331 Processing 1002
<1002>->3000 in context default
  //2010-10-08 18:41:43.134207 [NOTICE] sofia.c:528 Hangup sofia/internal/
z.com [CS_EXECUTE] [NORMAL_CLEARING]


  if (string=='OK') {

   //
   //
   //       <<<<<<<<<<<<<<<<<<------------------ It receive OK but it never
execute this two, and but it execute invalid macro
   //
   //
   //
   session.sayPhrase("valid", menuselection, "en");
   session.execute("transfer", '3000');
  } else {
   session.sayPhrase("invalid", menuselection, "en");
   return true;
}


/* FS wiki - example */
function promptfordigits_dtmf_callback(session, type, digits, arg)
{
   console_log("digit: " + digits.digit + "\n");
   promptfordigits_dtmf_digits += digits.digit;
   /* returning true does not interrupt the digit collection*/
   return(true);
}

/* FS wiki - example */
function promptfordigits(ivrsession, promptname, numdigits, timeout)
{
   var repeat = 0;

   console_log("saymenu: menu=[" + promptname + "] numdigits=[" + numdigits
+ "]\n");

   session.flushDigits();
   promptfordigits_dtmf_digits = "";

   while (ivrsession.ready() && promptfordigits_dtmf_digits.length <
numdigits && repeat < 3) {
       /* play phrase - if digit keyed while playing callback will catch
them.
       If less than numdigits collected we get the rest after the prompt.*/
       ivrsession.sayPhrase(promptname, numdigits, "en",
promptfordigits_dtmf_callback, "");

       console_log("Prompt done=[" + promptname + "] Collected " +
promptfordigits_dtmf_digits.length + " digits [" +
promptfordigits_dtmf_digits + "]\n");

       /* if caller still here and has not entered any selection yet (or
less than numdigits entered) - wait for the rest of the digits*/
       if (ivrsession.ready() && promptfordigits_dtmf_digits.length <
numdigits ) {
           promptfordigits_dtmf_digits += ivrsession.getDigits(numdigits -
promptfordigits_dtmf_digits.length, "", timeout);
           /* if still no selection or insufficient digits repeat menu */
           if (promptfordigits_dtmf_digits.length < numdigits) {
               promptfordigits_dtmf_digits = "";
               repeat++;
           }
       }
   }
   return(promptfordigits_dtmf_digits);
}



function retry()
{
  /* FS wiki - example */
  menuselection = promptfordigits(session, "DialByNumberMenu", 8, 6000);

  if ( session.ready() ) {

  var curl = new CURL();
  curl.run( "POST",
            "http://portal/conference/query",
            "om=" + menuselection + "&ty=__",
             my_callback,
             "my arg\n",
             "p:ier");

  }

}


/**
 * Let's answer our call
 */
session.answer();
retry();




Thanks & Regards
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From brokendash at gmail.com  Fri Oct  8 09:55:24 2010
From: brokendash at gmail.com (broken dash)
Date: Fri, 8 Oct 2010 11:55:24 -0500
Subject: [Freeswitch-users] FreeSWITCH parameters for voice quality
In-Reply-To: <AANLkTi=aVU7Ojv_xYQEDcvvrztq-FH20_mOHGk_2btLF@mail.gmail.com>
References: <1286450460556-5610915.post@n2.nabble.com>
	<AANLkTi=aVU7Ojv_xYQEDcvvrztq-FH20_mOHGk_2btLF@mail.gmail.com>
Message-ID: <AANLkTi=O8gB6SY0AyQCjsthkaL1Gr+52kCHV+PLetpTN@mail.gmail.com>

I've had some problems with the windows Xlite client during calls over
my wireless lan and setting transport type to tcp helped. The call
would start getting randomly choppy on me, etc.. but no analog static
noise tho... are you calling a POTS line or just another softphone?
Have you tried turning on VAD, or maybe passing -hp on FS startup?


Brian  :-)


On Thu, Oct 7, 2010 at 10:52 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> The defaults should be ideal.
>
> On Thu, Oct 7, 2010 at 6:21 AM, rex.alex <rajesh.npnr at yahoo.com> wrote:
>> Hello,
>>
>> I have FreeSWITCH on Cent OS 5.4 physical server and I am using X-Lite soft
>> phone on Windows client to make external calling through a registered
>> gateway. But, I am facing issues like voice breakage and static noise on
>> calls. I am monitoring the network connectivity but before that I want to
>> make sure that all FreeSWITCH parameters have been configured correctly.
>>
>> Please let me know as far as FreeSWITCH is concern what are the parameters
>> that I have to check for these kind of issues?
>>
>> Thanks,
>> Rex
>> ________________________________
>> View this message in context: FreeSWITCH parameters for voice quality
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Fri Oct  8 10:06:54 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 8 Oct 2010 10:06:54 -0700
Subject: [Freeswitch-users] Dead lua scripts
In-Reply-To: <F5C4BC8D-E276-4125-AAB3-655715875EFB@freeswitch.org>
References: <4CAE15F7.9090807@communicatefreely.net>
	<AANLkTinKznsWB6Zzf3cexnqHjesgQ_gsK++KkKD_h+U0@mail.gmail.com>
	<4CAE7EB0.2070600@communicatefreely.net>
	<F5C4BC8D-E276-4125-AAB3-655715875EFB@freeswitch.org>
Message-ID: <AANLkTikVadzzA5VP7UBksFvA12kZg4MwrwtAixwh_aL5@mail.gmail.com>

If you have the FS book this topic is discussed nicely in chapter 7.
-MC

On Fri, Oct 8, 2010 at 7:45 AM, Brian West <brian at freeswitch.org> wrote:

> Well anthony gave you a hint... wrap your whole loop in a
> while(session:ready) it will then end the script if the session hangs up.
>
> /b
>
> On Oct 7, 2010, at 9:15 PM, Tim St. Pierre wrote:
>
> > I had to completely restart freeswitch to make these calls go away.  Is
> there a less disruptive way
> > to terminate the scripts?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Oct  8 10:44:46 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 8 Oct 2010 10:44:46 -0700
Subject: [Freeswitch-users] Get on IRC day!!
Message-ID: <AANLkTimdGX0ganMLyd3uTD3Ux1spa5xx8mShuV9DEww9@mail.gmail.com>

Hey all, we need bodies in IRC! It's Friday, let's have a good group to
answer questions and help all the newbies.

-MC
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From Prometheus001 at gmx.net  Fri Oct  8 11:18:05 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 08 Oct 2010 20:18:05 +0200
Subject: [Freeswitch-users] german prompts - first release - how to get
 it to work
In-Reply-To: <4C9C6F0B.2060202@xpirio.com>
References: <4C9C6F0B.2060202@xpirio.com>
Message-ID: <4CAF605D.6080805@gmx.net>

I tried to install this, but I failed so far
    [ERR] switch_ivr_play_say.c:113 Can't find language de.
    [WARNING] switch_ivr_play_say.c:339 Macro [voicemail_hello]: '' did
not match any patterns


Step 1) I tried the follwowing (which worked)
(exchanged the sound files)
  I copied all geman language files to
  /usr/local/freeswitch/sounds/en/us/callie
and switched voice to "en"
I could hear all german annoucements without any problems, but of course
the phrase handling was not perfect


Step 2) Then I made the next step (which didn't work)
("de" configuration with english phrase management)
 I and switched voice to "de" and modified the geman language files
/usr/local/freeswitch/conf/lang/de/de.xml to
<include>
  <language name="de" sound-path="$${base_dir}/sounds/de/de/callie"
tts-engine="cepstral" tts-voice="katrin">
    <X-PRE-PROCESS cmd="include" data="demo/demo.xml"/>
    <!--voicemail_de_tts is purely implemented with tts, we need a files
based implementation too -->
    <!-- <X-PRE-PROCESS cmd="include" data="vm/tts.xml"/> -->
    <X-PRE-PROCESS cmd="include" data="vm/sounds.xml"/>  <!-- vm/tts.xml
if you want to use tts and have cepstral -->
    <X-PRE-PROCESS cmd="include" data="dir/sounds.xml"/>  <!--
dir/tts.xml if you want to use tts and have cepstral -->
  </language>
</include>
and copied vm/sounds.xml and dir/sounds.xml from the en directory
I expected to have the same behaviour and english phrasse handling with
german wavs as before. But this didn't work
Again
    [ERR] switch_ivr_play_say.c:113 Can't find language de.
    [WARNING] switch_ivr_play_say.c:339 Macro [voicemail_hello]: '' did
not match any patterns


Step 3) Then I made the next step (which didn't work)
(copy of en configuration with english phrase management and small
changes to de)
  I copied all language config files from
  /usr/local/freeswitch/conf/lang/en to /usr/local/freeswitch/conf/lang/de
and modified the de.xml as above
  <language name="de" sound-path="$${base_dir}/sounds/de/de/callie"
tts-engine="cepstral" tts-voice="katrin">
    <X-PRE-PROCESS cmd="include" data="demo/demo.xml"/>
    <!--voicemail_de_tts is purely implemented with tts, we need a files
based implementation too -->
    <!-- <X-PRE-PROCESS cmd="include" data="vm/tts.xml"/> -->
    <X-PRE-PROCESS cmd="include" data="vm/sounds.xml"/>  <!-- vm/tts.xml
if you want to use tts and have cepstral -->
    <X-PRE-PROCESS cmd="include" data="dir/sounds.xml"/>  <!--
dir/tts.xml if you want to use tts and have cepstral -->
  </language>
</include>
(just changed en/us to /de/de)
Again
    [ERR] switch_ivr_play_say.c:113 Can't find language de.
    [WARNING] switch_ivr_play_say.c:339 Macro [voicemail_hello]: '' did
not match any patterns

It seems that my freeswith doesn't accept the german language.
Mod_say_de is loaded successfully:
2010-10-08 20:12:21.929632 [DEBUG] switch_loadable_module.c:764 Write
lock interface 'de' to wait for existing references.
freeswitch at internal> 2010-10-08 20:12:21.929632 [NOTICE]
switch_loadable_module.c:772 Deleting Say interface 'de'
2010-10-08 20:12:21.933896 [CONSOLE] switch_loadable_module.c:1401
mod_say_de has no shutdown routine
2010-10-08 20:12:21.933896 [CONSOLE] switch_loadable_module.c:1418
mod_say_de unloaded.
2010-10-08 20:12:22.112108 [INFO] mod_pocketsphinx.c:482 PocketSphinx
Reloaded
2010-10-08 20:12:22.155122 [INFO] mod_enum.c:808 ENUM Reloaded
2010-10-08 20:12:22.177251 [INFO] switch_time.c:950 Timezone reloaded
530 definitions
2010-10-08 20:12:22.180670 [CONSOLE] switch_loadable_module.c:946
Successfully Loaded [mod_say_de]
2010-10-08 20:12:22.182284 [NOTICE] switch_loadable_module.c:399 Adding
Say interface 'de'

What can I do to fix this problem (GIT is from last week)

Best regards
Peter


Christian L?schenkohl schrieb:
> hello freeswitch community
>
> i can now announce the first release of the german prompt set.
> you can download it under http://freeswitch.xpirio.com/.
>
> this package is not useable at this time, because there are some xml files that
> need to be created or changed first. also the prompts are not 100% perfect and complete.
>
> i'll work on these topics and keep you updated.
>
> if somebody wants to help, please take a look at the prompts an send me your corrections (directly
> to my e-mail address, not to the list please).
> also look at the file de2.xls where you can find the prompts the need to be recorded or corrected.
> please take a look at the xml files under conf/lang, they also need to be created and/or modified too.
>
> br
>
>   


From curriegrad2004 at gmail.com  Fri Oct  8 11:19:12 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Fri, 8 Oct 2010 11:19:12 -0700
Subject: [Freeswitch-users] mod_dingaling segfault with libgcrypt and gnutls
	on Fedora 13
Message-ID: <AANLkTi=8tzOK4TMeCee570LDROpGgS0ej18VAu53QMVT@mail.gmail.com>

Hey all,

I've read and understand somewhere that with a bad copy of gnutls and
libgcrypt can cause mod_dingaling to segfault when loading or running.
However, I'd like to ask if there's anyone out there trying to use
mod_dingaling with Fedora 13's included libgcrypt and gnutls-devel
packages as I am experiencing segfaults when I try to load
mod_dingaling.


From tculjaga at gmail.com  Fri Oct  8 11:25:11 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 8 Oct 2010 20:25:11 +0200
Subject: [Freeswitch-users] [Freeswitch-dev] Get on IRC day!!
In-Reply-To: <AANLkTimdGX0ganMLyd3uTD3Ux1spa5xx8mShuV9DEww9@mail.gmail.com>
References: <AANLkTimdGX0ganMLyd3uTD3Ux1spa5xx8mShuV9DEww9@mail.gmail.com>
Message-ID: <AANLkTine8ef5z-R+Qsu5ftwewHfvP2X7KAimJ-mmZMDk@mail.gmail.com>

On Fri, Oct 8, 2010 at 7:44 PM, Michael Collins <msc at freeswitch.org> wrote:

> Hey all, we need bodies in IRC! It's Friday, let's have a good group to
> answer questions and help all the newbies.
>
> -MC
>
>
Firefox can't establish a connection to the server at irc.freenode.net.
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From tculjaga at gmail.com  Fri Oct  8 11:47:24 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 8 Oct 2010 20:47:24 +0200
Subject: [Freeswitch-users] german prompts - first release - how to get
 it to work
In-Reply-To: <4CAF605D.6080805@gmx.net>
References: <4C9C6F0B.2060202@xpirio.com>
	<4CAF605D.6080805@gmx.net>
Message-ID: <AANLkTi=f7RQ+nfOkWofZJmPk4g6BmQNwDV1jygvM7JAj@mail.gmail.com>

On Fri, Oct 8, 2010 at 8:18 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> I tried to install this, but I failed so far
>    [ERR] switch_ivr_play_say.c:113 Can't find language de.
>    [WARNING] switch_ivr_play_say.c:339 Macro [voicemail_hello]: '' did
> not match any patterns
>
>

well, you didn't load mod_say_de

i have EN, DE, FR, HR, IT, RU languages working fine



anyhow, to add a new language

edit freeswitch.xml and add your language within phrase section

e.g.:

  <section name="phrases" description="Speech Phrase Management">
    <macros>
      <X-PRE-PROCESS cmd="include" data="lang/de/*.xml"/>
      <X-PRE-PROCESS cmd="include" data="lang/en/*.xml"/>
      <X-PRE-PROCESS cmd="include" data="lang/fr/*.xml"/>
      <X-PRE-PROCESS cmd="include" data="lang/it/*.xml"/>
      <X-PRE-PROCESS cmd="include" data="lang/hr/*.xml"/>
      <X-PRE-PROCESS cmd="include" data="lang/ru/*.xml"/>
    </macros>
  </section>


go to:
/usr/local/freeswitch/conf/lang

and create a directory ... e.g. de in your case

change directory to de (cd de) and copy files from another language e.g. en.


cp -r ../en/* .

than rename en.xml into de.xml.

[tculjaga at cxss01 de]$ mv en.xml de.xml
edit the xml file in order to fit your needs.

here is my version:

[tculjaga at cxss01 de]$ cat de.xml
<include>
  <language name="de" sound-path="$${sounds_dir}/de/de/helge/"
tts-engine="cepstral" tts-voice="callie">
    <X-PRE-PROCESS cmd="include" data="demo/*.xml"/>
    <X-PRE-PROCESS cmd="include" data="vm/sounds.xml"/>
    <X-PRE-PROCESS cmd="include" data="dir/sounds.xml"/>
    <!-- <X-PRE-PROCESS cmd="include" data="ivr/ivr_funcs.xml"/> -->
  </language>
</include>

demo/*.xml
vm/sounds.xml
dir/sounds.xml
ivr/ivr_functions.xml

these are files containing macros you will later use in dialplan. You need
to have them consistent across all the languages.

when you done this, its time for your voice files.
go to:
/usr/local/freeswitch/sounds

copy the directory structure form another language e.g.  en, wipe the *.wav
files and copy your own files of course with the same naming!

build and load mod_say_de and thats it ...



hope it helps!
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From fraserredmond at gmail.com  Fri Oct  8 12:07:35 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 8 Oct 2010 20:07:35 +0100
Subject: [Freeswitch-users] Loss of first second of media from termination
	provider
Message-ID: <AANLkTime5SsVYNe8uaJ2FbbK25NRSyVeLF9Nk4ZicSU+@mail.gmail.com>

I've just set up with a new termination provider, and am finding that the
first second of media is being lost.

I get early media (ringing), then when the call is answered the first half a
second to a second is lost, then the rest of the call is fine.

Their engineers have run a sip trace and a full packet trace, and it looks
like everything is being delivered from the pstn to my server. They said
that none of their other customers have reported this problem before.

I've tested connecting to pstn's in several different countries, with the
same results. I've used 2-3 other termination providers (1 still in current
use) which have all been fine.

So it seems like it's a combination of Freeswitch with this provider.

This is running on Ubuntu on a Amazon AWS server, the version of freeswitch
is about 2 weeks old, I'll be trying on a new server with the latest version
tomorrow, in case that helps.

Any ideas?

Cheers,
Fraser
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From david.ponzone at ipeva.fr  Fri Oct  8 12:20:08 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 8 Oct 2010 21:20:08 +0200
Subject: [Freeswitch-users] Get on IRC day!!
In-Reply-To: <AANLkTimdGX0ganMLyd3uTD3Ux1spa5xx8mShuV9DEww9@mail.gmail.com>
References: <AANLkTimdGX0ganMLyd3uTD3Ux1spa5xx8mShuV9DEww9@mail.gmail.com>
Message-ID: <3DB3F973-A28E-45F5-9609-E5B707A38183@ipeva.fr>

Michael,

sorry, it's movie night with the kids :)

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 08/10/2010 ? 19:44, Michael Collins a ?crit :

> Hey all, we need bodies in IRC! It's Friday, let's have a good group  
> to answer questions and help all the newbies.
>
> -MC
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From xyangni at gmail.com  Fri Oct  8 12:23:18 2010
From: xyangni at gmail.com (xuyan yang)
Date: Sat, 9 Oct 2010 03:23:18 +0800
Subject: [Freeswitch-users] iLBC codec generating only noise.
Message-ID: <AANLkTinz4JEfyGnjU-4nneOFg=A0qXHjVw9xE=jp2qSo@mail.gmail.com>

Hi,

I am trying to use iLBC codec with both eyebeam and some iphone client. When
a ivr is called, the client can here system voice and make dtmf input. but
the voice recorded from client's microphone is only noise.
The call between 2 clients also have such problem.

Sometimes, FS may even got crashed with the following information:

2010-10-08 19:56:21.343750 [INFO] switch_rtp.c:2527 Auto Changing port from
10.20.132.244:18570 to 8
2.132.139.197:19536
alloc: asked for negative size -2147483648

in var.xml, I have added iLBC instead of iLBC at 30i, because  iLBC at 30i is
ignored during codec negotiation.
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h
,G722,speex at 8000h@20i,iLBC,PCMU,PCMA,GSM"/>

The git version last week is used in my test. Is there anything wrong with
my setup? Thanks.
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From Nabble at slickdeals.endjunk.com  Fri Oct  8 13:06:58 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 8 Oct 2010 13:06:58 -0700 (PDT)
Subject: [Freeswitch-users] Get on IRC day!!
In-Reply-To: <AANLkTimdGX0ganMLyd3uTD3Ux1spa5xx8mShuV9DEww9@mail.gmail.com>
References: <AANLkTimdGX0ganMLyd3uTD3Ux1spa5xx8mShuV9DEww9@mail.gmail.com>
Message-ID: <1286568418994-5616330.post@n2.nabble.com>



mercutioviz wrote:
> Hey all, we need bodies in IRC! It's Friday, let's have a good group to
> answer questions and help all the newbies.
The following opinions are based on mine and/or others whom I had spoken to
w.r.t FS.

1. FS needs to setup/provide a good forum, instead of the unfavorable
mailing list. I tried to bring in as many newbies and/or other PBX users as
I could (at least 4 newbies) and none of them has shown up.  I followed up
with them and got several answers. One of them is I had had my share with
mailing lists in the past and I don't like to deal with it again. I sure
don't mind if FS has a good forum to attend. One managed to lurk around and
told me that there are a lot of FS experts here trying to help, but she
wouldn't understand at all when reading the replies addressed to some posts.
She told me the replies/suggestions were brief and vague with no examples to
lead/guide any newbies to make them trying to understand. Honestly, I agree
with her. This leads me to the next item below.

2. When helping a newbie, try to explain in more detail (best with an
example) and don't be brief. For instance, when I posted 
http://freeswitch-users.2379917.n2.nabble.com/More-questions-migrating-from-Asterisk-to-FS-td5260522.html#a5260522
this  asking for some help, I know everyone, including Antonio, had
voluntarily tried to help. However, telling a newbie like me to use param is
really not going to help at all, let alone may completely discourage a
newbie who tries to learn FS. Some people, be it a newbie or an expert, will
learn better to understand through examples. For sure, we don't really want
to encourage any spoon fed activities here.

3. The Wiki needs to be revised with additional examples. There may be some
few good suggestions with examples replied to the posts that helped the
posters. Perhaps, one needs to make sure those good suggestions with
examples get added to the Wiki for others benefit. The 
http://wiki.freeswitch.org/wiki/Dialplan_XML Dialplan XML  and 
http://wiki.freeswitch.org/wiki/SIP_Provider_Examples SIP Provider Examples 
are two good examples to follow. This way, a newbie can learn easier,
instead of bombarding the list with the same questions from time to time.

The following is a bit off topic, but may be related.

I did briefly attend the past conference and noticed the speaker requested
the audiences to spread out and bring in additional FS users to attend the
next conference room. I believe there must be tons of VoIP users out there
who still don't know anything about FS and are frequent members of many VoIP
forums will be interested in FS as well as its conference calls. Some of
them may be pheasants in VoIP world and don't know much how to get into the
conference lines facility that they never heard of while others are merely
hesitant because they can't find a simple number to dial into the conference
room from an ATA device. For instance, I recently (as far back as 3 weeks
ago) told someone in a VoIP forum about FS conference calls and even
provided the  http://wiki.freeswitch.org/wiki/FS_weekly_2010_09_15 link  (at
the time). He replied back and told me that he doesn't know how to make his
VoIP device to support a SIP URI dial into the conference line. Perhaps,
this person may not sound like a good candidate at all, but at least this
goes to show FS conference call facility has at least a weak point. If FS
were to provide other means, i.e. SIPBroker, iNum, e164, etc., for any VoIP
users to dial into the conference calls, then I believe this will add more
audiences to the FS conference room.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Get-on-IRC-day-tp5615890p5616330.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Fri Oct  8 14:31:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 8 Oct 2010 14:31:53 -0700
Subject: [Freeswitch-users] Get on IRC day!!
In-Reply-To: <1286568418994-5616330.post@n2.nabble.com>
References: <AANLkTimdGX0ganMLyd3uTD3Ux1spa5xx8mShuV9DEww9@mail.gmail.com>
	<1286568418994-5616330.post@n2.nabble.com>
Message-ID: <AANLkTi=GSO2YDqjs9O=1ib=roP8nsbS6C9x6n8kQMkgz@mail.gmail.com>

On Fri, Oct 8, 2010 at 1:06 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> 1. FS needs to setup/provide a good forum, instead of the unfavorable
> mailing list.

No, it doesn't. We've heard this argument for three years. We've openly
stated that while *we*, personally (the core FS team) don't care for having
a forum, we are not against the community setting one up and maintaining it.
We had one person try two different things with a forum, one of which I
think is the Nabble thing. It died a slow, painful death from lack of
community support. Our community as a whole just doesn't want one. An
argument has been floated that having a forum will allow the community to
grow and thrive, and therefore *we* (the core FS team) should sponsor and
maintain it. Frankly, we don't buy it. We feel like having an AWESOME piece
of software with enthusiastic users is what makes the community grow and
thrive. If someone stays away from FreeSWITCH solely because they don't like
our combination of mailing list, IRC, and wiki then so be it. We respect
their decision.

I tried to bring in as many newbies and/or other PBX users as
> I could (at least 4 newbies) and none of them has shown up.  I followed up
> with them and got several answers. One of them is I had had my share with
> mailing lists in the past and I don't like to deal with it again. I sure
> don't mind if FS has a good forum to attend. One managed to lurk around and
> told me that there are a lot of FS experts here trying to help, but she
> wouldn't understand at all when reading the replies addressed to some
> posts.
>
She told me the replies/suggestions were brief and vague with no examples to
> lead/guide any newbies to make them trying to understand. Honestly, I agree
> with her. This leads me to the next item below.
>
> 2. When helping a newbie, try to explain in more detail (best with an
> example) and don't be brief. For instance, when I posted
>
> http://freeswitch-users.2379917.n2.nabble.com/More-questions-migrating-from-Asterisk-to-FS-td5260522.html#a5260522
> this  asking for some help, I know everyone, including Antonio, had
> voluntarily tried to help. However, telling a newbie like me to use param
> is
> really not going to help at all, let alone may completely discourage a
> newbie who tries to learn FS. Some people, be it a newbie or an expert,
> will
> learn better to understand through examples. For sure, we don't really want
> to encourage any spoon fed activities here.
>
I respect your honesty. I will encourage those answering newbie questions to
be more explicit wherever possible. Just understand that everyone on this
list and in IRC is an unpaid volunteer, giving of their time freely. In some
cases an expert has to make a choice: give a brief answer or give no answer.
Given the option, I think even a newbie would appreciate the quick answer as
opposed to silence on the wire.


>
> 3. The Wiki needs to be revised with additional examples.
>
Agreed. Volunteers welcome. :) Seriously, I need as much help as possible
with the wiki. I definitely have a core of people helping out. (You know who
you are, and I thank you profusely.) I would like to have more help. Keeping
the wiki updated is a big job. Anthony & company add features faster than we
can document them! In many cases, because Tony is such a great programmer,
he can add stuff in like an hour and it takes 2-3 hours for me to read the
source code, recompile, make changes to my dialplan or other configs, test
various scenarios, and then update the wiki with the new API, dp app, chan
var, SIP profile param, etc. All the while I have a day job and other
FreeSWITCH-related duties.


> There may be some
> few good suggestions with examples replied to the posts that helped the
> posters. Perhaps, one needs to make sure those good suggestions with
> examples get added to the Wiki for others benefit. The
> http://wiki.freeswitch.org/wiki/Dialplan_XML Dialplan XML  and
> http://wiki.freeswitch.org/wiki/SIP_Provider_Examples SIP Provider
> Examples
> are two good examples to follow. This way, a newbie can learn easier,
> instead of bombarding the list with the same questions from time to time.
>
We spent a ton of time on the FreeSWITCH book, which contains a lot of the
hand-holding newbies really need. It's relatively inexpensive and you can
get just the PDF if you like. (If you're destitute then you can probably
borrow the book from someone who has it.)


> <snip...>
> For instance, I recently (as far back as 3 weeks
> ago) told someone in a VoIP forum about FS conference calls and even
> provided the  http://wiki.freeswitch.org/wiki/FS_weekly_2010_09_15 link
>  (at
> the time). He replied back and told me that he doesn't know how to make his
> VoIP device to support a SIP URI dial into the conference line.

I'm not trying to be mean here, but honestly, if he can't dial any of our
ways of getting to the conference then he's *not ready for a VoIP
conference*!! We have many ways to connect:
PSTN: 1-919-386-990
SIP: sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
GTalk: gtalk:conf+888 at conference.freeswitch.org<gtalk%3Aconf%2B888 at conference.freeswitch.org>
Skype: call skype user "skypiax5"

That's five different ways of connecting. I'm sorry, but that's more than
enough ways to get people to call a *VoIP* conference about telecom
software. If people are incapable of connecting from one of these five ways
then they are not yet ready for FreeSWITCH.

I will keep working the wiki and making FreeSWITCH as easy as possible for
newbies. However, please keep in mind that those who don't have telecom,
networking, and/or VoIP experience are going to experience a much steeper
learning curve. That's just the nature of the beast. To ease their
transition maybe we could come up with a recommended reading list? "VoIP For
Dummies" kind of stuff maybe. If you guys have any suggestions I'll put the
on the wiki.

-MC
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From Prometheus001 at gmx.net  Fri Oct  8 14:42:55 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 08 Oct 2010 23:42:55 +0200
Subject: [Freeswitch-users] german prompts - first release - how to get
 it to work
In-Reply-To: <AANLkTi=f7RQ+nfOkWofZJmPk4g6BmQNwDV1jygvM7JAj@mail.gmail.com>
References: <4C9C6F0B.2060202@xpirio.com>	<4CAF605D.6080805@gmx.net>
	<AANLkTi=f7RQ+nfOkWofZJmPk4g6BmQNwDV1jygvM7JAj@mail.gmail.com>
Message-ID: <4CAF905F.4030407@gmx.net>

Thanks,

<X-PRE-PROCESS cmd="include" data="lang/de/*.xml"/>
did the trick. Now it works in German with TTS and voice files.

Thank you very much

Best regards
Peter

Tihomir Culjaga schrieb:
>
>
> On Fri, Oct 8, 2010 at 8:18 PM, Peter P GMX <Prometheus001 at gmx.net
> <mailto:Prometheus001 at gmx.net>> wrote:
>
>     I tried to install this, but I failed so far
>        [ERR] switch_ivr_play_say.c:113 Can't find language de.
>        [WARNING] switch_ivr_play_say.c:339 Macro [voicemail_hello]: '' did
>     not match any patterns
>
>
>
> well, you didn't load mod_say_de
>
> i have EN, DE, FR, HR, IT, RU languages working fine
>
>
>
> anyhow, to add a new language
>
> edit freeswitch.xml and add your language within phrase section
>
> e.g.:
>
>   <section name="phrases" description="Speech Phrase Management">
>     <macros>
>       <X-PRE-PROCESS cmd="include" data="lang/de/*.xml"/>
>       <X-PRE-PROCESS cmd="include" data="lang/en/*.xml"/>
>       <X-PRE-PROCESS cmd="include" data="lang/fr/*.xml"/>
>       <X-PRE-PROCESS cmd="include" data="lang/it/*.xml"/>
>       <X-PRE-PROCESS cmd="include" data="lang/hr/*.xml"/>
>       <X-PRE-PROCESS cmd="include" data="lang/ru/*.xml"/>
>     </macros>
>   </section>
>
>
> go to:
> /usr/local/freeswitch/conf/lang
>
> and create a directory ... e.g. de in your case
>
> change directory to de (cd de) and copy files from another language
> e.g. en.
>
> cp -r ../en/* .
>
> than rename en.xml into de.xml.
>
> [tculjaga at cxss01 de]$ mv en.xml de.xml
> edit the xml file in order to fit your needs.
>
> here is my version:
>
> [tculjaga at cxss01 de]$ cat de.xml
> <include>
>   <language name="de" sound-path="$${sounds_dir}/de/de/helge/"
> tts-engine="cepstral" tts-voice="callie">
>     <X-PRE-PROCESS cmd="include" data="demo/*.xml"/>
>     <X-PRE-PROCESS cmd="include" data="vm/sounds.xml"/>
>     <X-PRE-PROCESS cmd="include" data="dir/sounds.xml"/>
>     <!-- <X-PRE-PROCESS cmd="include" data="ivr/ivr_funcs.xml"/> -->
>   </language>
> </include>
>
> demo/*.xml
> vm/sounds.xml
> dir/sounds.xml
> ivr/ivr_functions.xml
>
> these are files containing macros you will later use in dialplan. You
> need to have them consistent across all the languages.
>
> when you done this, its time for your voice files.
> go to:
> /usr/local/freeswitch/sounds
>
> copy the directory structure form another language e.g.  en, wipe the
> *.wav files and copy your own files of course with the same naming!
>
> build and load mod_say_de and thats it ...
>
>
>
> hope it helps!
>
>
>
>
>
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From Prometheus001 at gmx.net  Fri Oct  8 15:01:17 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Sat, 09 Oct 2010 00:01:17 +0200
Subject: [Freeswitch-users] german prompts - first release - how to get
 it to work
In-Reply-To: <4CAF905F.4030407@gmx.net>
References: <4C9C6F0B.2060202@xpirio.com>	<4CAF605D.6080805@gmx.net>
	<AANLkTi=f7RQ+nfOkWofZJmPk4g6BmQNwDV1jygvM7JAj@mail.gmail.com>
	<4CAF905F.4030407@gmx.net>
Message-ID: <4CAF94AD.90508@gmx.net>

Wiki is also updated.



Peter P GMX schrieb:
> Thanks,
>
> <X-PRE-PROCESS cmd="include" data="lang/de/*.xml"/>
> did the trick. Now it works in German with TTS and voice files.
>
> Thank you very much
>
> Best regards
> Peter
>
> Tihomir Culjaga schrieb:
>   
>> On Fri, Oct 8, 2010 at 8:18 PM, Peter P GMX <Prometheus001 at gmx.net
>> <mailto:Prometheus001 at gmx.net>> wrote:
>>
>>     I tried to install this, but I failed so far
>>        [ERR] switch_ivr_play_say.c:113 Can't find language de.
>>        [WARNING] switch_ivr_play_say.c:339 Macro [voicemail_hello]: '' did
>>     not match any patterns
>>
>>
>>
>> well, you didn't load mod_say_de
>>
>> i have EN, DE, FR, HR, IT, RU languages working fine
>>
>>
>>
>> anyhow, to add a new language
>>
>> edit freeswitch.xml and add your language within phrase section
>>
>> e.g.:
>>
>>   <section name="phrases" description="Speech Phrase Management">
>>     <macros>
>>       <X-PRE-PROCESS cmd="include" data="lang/de/*.xml"/>
>>       <X-PRE-PROCESS cmd="include" data="lang/en/*.xml"/>
>>       <X-PRE-PROCESS cmd="include" data="lang/fr/*.xml"/>
>>       <X-PRE-PROCESS cmd="include" data="lang/it/*.xml"/>
>>       <X-PRE-PROCESS cmd="include" data="lang/hr/*.xml"/>
>>       <X-PRE-PROCESS cmd="include" data="lang/ru/*.xml"/>
>>     </macros>
>>   </section>
>>
>>
>> go to:
>> /usr/local/freeswitch/conf/lang
>>
>> and create a directory ... e.g. de in your case
>>
>> change directory to de (cd de) and copy files from another language
>> e.g. en.
>>
>> cp -r ../en/* .
>>
>> than rename en.xml into de.xml.
>>
>> [tculjaga at cxss01 de]$ mv en.xml de.xml
>> edit the xml file in order to fit your needs.
>>
>> here is my version:
>>
>> [tculjaga at cxss01 de]$ cat de.xml
>> <include>
>>   <language name="de" sound-path="$${sounds_dir}/de/de/helge/"
>> tts-engine="cepstral" tts-voice="callie">
>>     <X-PRE-PROCESS cmd="include" data="demo/*.xml"/>
>>     <X-PRE-PROCESS cmd="include" data="vm/sounds.xml"/>
>>     <X-PRE-PROCESS cmd="include" data="dir/sounds.xml"/>
>>     <!-- <X-PRE-PROCESS cmd="include" data="ivr/ivr_funcs.xml"/> -->
>>   </language>
>> </include>
>>
>> demo/*.xml
>> vm/sounds.xml
>> dir/sounds.xml
>> ivr/ivr_functions.xml
>>
>> these are files containing macros you will later use in dialplan. You
>> need to have them consistent across all the languages.
>>
>> when you done this, its time for your voice files.
>> go to:
>> /usr/local/freeswitch/sounds
>>
>> copy the directory structure form another language e.g.  en, wipe the
>> *.wav files and copy your own files of course with the same naming!
>>
>> build and load mod_say_de and thats it ...
>>
>>
>>
>> hope it helps!
>>
>>
>>
>>
>>
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>   
>>     
>
>   


From david.ponzone at ipeva.fr  Fri Oct  8 15:06:33 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sat, 9 Oct 2010 00:06:33 +0200
Subject: [Freeswitch-users] Get on IRC day!!
In-Reply-To: <AANLkTi=GSO2YDqjs9O=1ib=roP8nsbS6C9x6n8kQMkgz@mail.gmail.com>
References: <AANLkTimdGX0ganMLyd3uTD3Ux1spa5xx8mShuV9DEww9@mail.gmail.com>
	<1286568418994-5616330.post@n2.nabble.com>
	<AANLkTi=GSO2YDqjs9O=1ib=roP8nsbS6C9x6n8kQMkgz@mail.gmail.com>
Message-ID: <22087F1C-B0A5-4132-BDF7-446448E370BB@ipeva.fr>

I concur what Michael says.
We see quite some people on IRC, who (and I am just stating a fact)  
require a LOT of help, althought they didn't really do their homework,  
and their main objective is to connect 2 phones at home to a common  
SIP account.
So I would agree that in a such case, detailed explanations are  
required. But the issue is, most FreeSWITCHers won't have the time for  
it, and those newbies don't have money to spend for consultancy or a  
full VoIP course.

And for someone already "VOIP-enabled", FreeSWITCH's learning curve is  
quite fast, if you immerge yourself in it for at least 2 weeks, if you  
read the wiki and the book, and if you join the channel, not only to  
ask questions around, but also to read other discussions. I learn so  
many things just being there.

Also, a lot of people think that there is a team somewhere in charge  
of completing the wiki.
Well, no.
It's everybody's task to do it, if one feels he has an interesting  
info to share, or a nice example to give to newbies.
Also there are still quite some mistakes to correct (I corrected one  
the other day in the doublenat scenario example), and variables to  
document.
Even a newbie can document, once he has understood something that is  
not explained on the wiki.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 08/10/2010 ? 23:31, Michael Collins a ?crit :

>
>
> On Fri, Oct 8, 2010 at 1:06 PM, mazilo  
> <Nabble at slickdeals.endjunk.com> wrote:
>
> 1. FS needs to setup/provide a good forum, instead of the unfavorable
> mailing list.
> No, it doesn't. We've heard this argument for three years. We've  
> openly stated that while *we*, personally (the core FS team) don't  
> care for having a forum, we are not against the community setting  
> one up and maintaining it. We had one person try two different  
> things with a forum, one of which I think is the Nabble thing. It  
> died a slow, painful death from lack of community support. Our  
> community as a whole just doesn't want one. An argument has been  
> floated that having a forum will allow the community to grow and  
> thrive, and therefore *we* (the core FS team) should sponsor and  
> maintain it. Frankly, we don't buy it. We feel like having an  
> AWESOME piece of software with enthusiastic users is what makes the  
> community grow and thrive. If someone stays away from FreeSWITCH  
> solely because they don't like our combination of mailing list, IRC,  
> and wiki then so be it. We respect their decision.
>
> I tried to bring in as many newbies and/or other PBX users as
> I could (at least 4 newbies) and none of them has shown up.  I  
> followed up
> with them and got several answers. One of them is I had had my share  
> with
> mailing lists in the past and I don't like to deal with it again. I  
> sure
> don't mind if FS has a good forum to attend. One managed to lurk  
> around and
> told me that there are a lot of FS experts here trying to help, but  
> she
> wouldn't understand at all when reading the replies addressed to  
> some posts.
> She told me the replies/suggestions were brief and vague with no  
> examples to
> lead/guide any newbies to make them trying to understand. Honestly,  
> I agree
> with her. This leads me to the next item below.
>
> 2. When helping a newbie, try to explain in more detail (best with an
> example) and don't be brief. For instance, when I posted
> http://freeswitch-users.2379917.n2.nabble.com/More-questions-migrating-from-Asterisk-to-FS-td5260522.html#a5260522
> this  asking for some help, I know everyone, including Antonio, had
> voluntarily tried to help. However, telling a newbie like me to use  
> param is
> really not going to help at all, let alone may completely discourage a
> newbie who tries to learn FS. Some people, be it a newbie or an  
> expert, will
> learn better to understand through examples. For sure, we don't  
> really want
> to encourage any spoon fed activities here.
> I respect your honesty. I will encourage those answering newbie  
> questions to be more explicit wherever possible. Just understand  
> that everyone on this list and in IRC is an unpaid volunteer, giving  
> of their time freely. In some cases an expert has to make a choice:  
> give a brief answer or give no answer. Given the option, I think  
> even a newbie would appreciate the quick answer as opposed to  
> silence on the wire.
>
>
> 3. The Wiki needs to be revised with additional examples.
> Agreed. Volunteers welcome. :) Seriously, I need as much help as  
> possible with the wiki. I definitely have a core of people helping  
> out. (You know who you are, and I thank you profusely.) I would like  
> to have more help. Keeping the wiki updated is a big job. Anthony &  
> company add features faster than we can document them! In many  
> cases, because Tony is such a great programmer, he can add stuff in  
> like an hour and it takes 2-3 hours for me to read the source code,  
> recompile, make changes to my dialplan or other configs, test  
> various scenarios, and then update the wiki with the new API, dp  
> app, chan var, SIP profile param, etc. All the while I have a day  
> job and other FreeSWITCH-related duties.
>
> There may be some
> few good suggestions with examples replied to the posts that helped  
> the
> posters. Perhaps, one needs to make sure those good suggestions with
> examples get added to the Wiki for others benefit. The
> http://wiki.freeswitch.org/wiki/Dialplan_XML Dialplan XML  and
> http://wiki.freeswitch.org/wiki/SIP_Provider_Examples SIP Provider  
> Examples
> are two good examples to follow. This way, a newbie can learn easier,
> instead of bombarding the list with the same questions from time to  
> time.
> We spent a ton of time on the FreeSWITCH book, which contains a lot  
> of the hand-holding newbies really need. It's relatively inexpensive  
> and you can get just the PDF if you like. (If you're destitute then  
> you can probably borrow the book from someone who has it.)
>
> <snip...>
> For instance, I recently (as far back as 3 weeks
> ago) told someone in a VoIP forum about FS conference calls and even
> provided the  http://wiki.freeswitch.org/wiki/FS_weekly_2010_09_15  
> link  (at
> the time). He replied back and told me that he doesn't know how to  
> make his
> VoIP device to support a SIP URI dial into the conference line.
> I'm not trying to be mean here, but honestly, if he can't dial any  
> of our ways of getting to the conference then he's *not ready for a  
> VoIP conference*!! We have many ways to connect:
> PSTN: 1-919-386-990
> SIP: sip:888 at conference.freeswitch.org
> GTalk: gtalk:conf+888 at conference.freeswitch.org
> Skype: call skype user "skypiax5"
>
> That's five different ways of connecting. I'm sorry, but that's more  
> than enough ways to get people to call a *VoIP* conference about  
> telecom software. If people are incapable of connecting from one of  
> these five ways then they are not yet ready for FreeSWITCH.
>
> I will keep working the wiki and making FreeSWITCH as easy as  
> possible for newbies. However, please keep in mind that those who  
> don't have telecom, networking, and/or VoIP experience are going to  
> experience a much steeper learning curve. That's just the nature of  
> the beast. To ease their transition maybe we could come up with a  
> recommended reading list? "VoIP For Dummies" kind of stuff maybe. If  
> you guys have any suggestions I'll put the on the wiki.
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mario_fs at mgtech.com  Fri Oct  8 16:29:07 2010
From: mario_fs at mgtech.com (Mario)
Date: Fri, 08 Oct 2010 16:29:07 -0700
Subject: [Freeswitch-users] Get on IRC day!!
In-Reply-To: <22087F1C-B0A5-4132-BDF7-446448E370BB@ipeva.fr>
References: <AANLkTimdGX0ganMLyd3uTD3Ux1spa5xx8mShuV9DEww9@mail.gmail.com>	<1286568418994-5616330.post@n2.nabble.com>	<AANLkTi=GSO2YDqjs9O=1ib=roP8nsbS6C9x6n8kQMkgz@mail.gmail.com>
	<22087F1C-B0A5-4132-BDF7-446448E370BB@ipeva.fr>
Message-ID: <4CAFA943.7060301@mgtech.com>

I hope you don't mind comments and a thank you from a newbie. My
background is VOIP on Linksys SPA9000(I hate)/SPA962 phones in small
office for 3 years. I tried Asterisk and gave up, then FreeSwitch
earlier this year. I saw the book was coming so waited for that (I read
it 3 times). I am a mainframe computer programmer so not too dumb but
VOIP is all new to me. From a real newbies perspective:

1. The book was invaluable to get started. The wiki was invaluable to
keep going.

2. I started simple, trying to duplicate what I had: Two separate
systems one with answering and one with IVR that calls a cell phone.

3. I got it to work with the book and wiki and valuable help from the
mailing list that gave me info not in the book  or Wiki (that I could
find, such as bridge had to use :_:). In other words, I needed all three
resources to duplicate what I had.

4. I decided to add more functions since Freeswitch was so powerful, and
I was having fun with it. I have some pretty slick things (to me anyway)
implemented. I sure learned a lot I will share on the wiki.

5. But I have two problems, one a show stopper (prevents calls) since
September 21. Some people need to be hand-held all the time but I am not
one of them. Posting in the list is my last resort, usually after
several hours/days of searching. I have spent weeks on the two problems
(20 hours in the last 3 days). I posted on the list and got on IRC for
the first time today. My impression was that IRC was for the pros,
although to my surprise I was able to answer a newbies question! I
thought I would have figured it out by now but no. This scenario of a
newbie getting this far and having a show-stopper would be nice to
address. Could be a separate IRC for newbies?

As for paying/donations, I built FS on my OpenSuse system and plan to
put my goodies on the wiki. I also built this on osX where the final FS
will run (Mac mini). It was tough and someone else on the list had
problems on 10.6.4 and I provided everything I learned. I will also add
this to the wiki. Finally, my business has gone down a lot and don't
have much to spend but I was going to get the Hogans Heroes DVD
collection on the wish list or something like that (BTW, it's on HD on
UHD). You guys deserve a lot more but I am limited these days.

Anyway, thanks for a great product and I wish you luck and $$$ from it.
I will buy every book you publish and recommend FreeSwitch to anyone who
can use it.
Mario

On 10/08/10 15:06, David Ponzone wrote:
> I concur what Michael says.
> We see quite some people on IRC, who (and I am just stating a fact)
> require a LOT of help, althought they didn't really do their homework,
> and their main objective is to connect 2 phones at home to a common SIP
> account.
> So I would agree that in a such case, detailed explanations are
> required. But the issue is, most FreeSWITCHers won't have the time for
> it, and those newbies don't have money to spend for consultancy or a
> full VoIP course.
> 
> And for someone already "VOIP-enabled", FreeSWITCH's learning curve is
> quite fast, if you immerge yourself in it for at least 2 weeks, if you
> read the wiki and the book, and if you join the channel, not only to ask
> questions around, but also to read other discussions. I learn so many
> things just being there.
> 
> Also, a lot of people think that there is a team somewhere in charge of
> completing the wiki.
> Well, no.
> It's everybody's task to do it, if one feels he has an interesting info
> to share, or a nice example to give to newbies.
> Also there are still quite some mistakes to correct (I corrected one the
> other day in the doublenat scenario example), and variables to document.
> Even a newbie can document, once he has understood something that is not
> explained on the wiki.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com
> <BLOCKED::http://www.ipeva-studio.com/>
> 
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion non autoris?e est interdite. Tout message ?lectronique est
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous
> n'?tes pas destinataire de ce message, merci de le d?truire
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
> 
> 
> 
> Le 08/10/2010 ? 23:31, Michael Collins a ?crit :
> 
>>
>>
>> On Fri, Oct 8, 2010 at 1:06 PM, mazilo <Nabble at slickdeals.endjunk.com
>> <mailto:Nabble at slickdeals.endjunk.com>> wrote:
>>
>>
>>     1. FS needs to setup/provide a good forum, instead of the unfavorable
>>     mailing list. 
>>
>> No, it doesn't. We've heard this argument for three years. We've
>> openly stated that while *we*, personally (the core FS team) don't
>> care for having a forum, we are not against the community setting one
>> up and maintaining it. We had one person try two different things with
>> a forum, one of which I think is the Nabble thing. It died a slow,
>> painful death from lack of community support. Our community as a whole
>> just doesn't want one. An argument has been floated that having a
>> forum will allow the community to grow and thrive, and therefore *we*
>> (the core FS team) should sponsor and maintain it. Frankly, we don't
>> buy it. We feel like having an AWESOME piece of software with
>> enthusiastic users is what makes the community grow and thrive. If
>> someone stays away from FreeSWITCH solely because they don't like our
>> combination of mailing list, IRC, and wiki then so be it. We respect
>> their decision.
>>
>>     I tried to bring in as many newbies and/or other PBX users as
>>     I could (at least 4 newbies) and none of them has shown up.  I
>>     followed up
>>     with them and got several answers. One of them is I had had my
>>     share with
>>     mailing lists in the past and I don't like to deal with it again.
>>     I sure
>>     don't mind if FS has a good forum to attend. One managed to lurk
>>     around and
>>     told me that there are a lot of FS experts here trying to help,
>>     but she
>>     wouldn't understand at all when reading the replies addressed to
>>     some posts. 
>>
>>     She told me the replies/suggestions were brief and vague with no
>>     examples to
>>     lead/guide any newbies to make them trying to understand.
>>     Honestly, I agree
>>     with her. This leads me to the next item below.
>>
>>     2. When helping a newbie, try to explain in more detail (best with an
>>     example) and don't be brief. For instance, when I posted
>>     http://freeswitch-users.2379917.n2.nabble.com/More-questions-migrating-from-Asterisk-to-FS-td5260522.html#a5260522
>>     this  asking for some help, I know everyone, including Antonio, had
>>     voluntarily tried to help. However, telling a newbie like me to
>>     use param is
>>     really not going to help at all, let alone may completely discourage a
>>     newbie who tries to learn FS. Some people, be it a newbie or an
>>     expert, will
>>     learn better to understand through examples. For sure, we don't
>>     really want
>>     to encourage any spoon fed activities here.
>>
>> I respect your honesty. I will encourage those answering newbie
>> questions to be more explicit wherever possible. Just understand that
>> everyone on this list and in IRC is an unpaid volunteer, giving of
>> their time freely. In some cases an expert has to make a choice: give
>> a brief answer or give no answer. Given the option, I think even a
>> newbie would appreciate the quick answer as opposed to silence on the
>> wire.
>>  
>>
>>
>>     3. The Wiki needs to be revised with additional examples.
>>
>> Agreed. Volunteers welcome. :) Seriously, I need as much help as
>> possible with the wiki. I definitely have a core of people helping
>> out. (You know who you are, and I thank you profusely.) I would like
>> to have more help. Keeping the wiki updated is a big job. Anthony &
>> company add features faster than we can document them! In many cases,
>> because Tony is such a great programmer, he can add stuff in like an
>> hour and it takes 2-3 hours for me to read the source code, recompile,
>> make changes to my dialplan or other configs, test various scenarios,
>> and then update the wiki with the new API, dp app, chan var, SIP
>> profile param, etc. All the while I have a day job and other
>> FreeSWITCH-related duties.
>>  
>>
>>     There may be some
>>     few good suggestions with examples replied to the posts that
>>     helped the
>>     posters. Perhaps, one needs to make sure those good suggestions with
>>     examples get added to the Wiki for others benefit. The
>>     http://wiki.freeswitch.org/wiki/Dialplan_XML Dialplan XML  and
>>     http://wiki.freeswitch.org/wiki/SIP_Provider_Examples SIP Provider
>>     Examples
>>     are two good examples to follow. This way, a newbie can learn easier,
>>     instead of bombarding the list with the same questions from time
>>     to time.
>>
>> We spent a ton of time on the FreeSWITCH book, which contains a lot of
>> the hand-holding newbies really need. It's relatively inexpensive and
>> you can get just the PDF if you like. (If you're destitute then you
>> can probably borrow the book from someone who has it.)
>>  
>>
>>     <snip...>
>>     For instance, I recently (as far back as 3 weeks
>>     ago) told someone in a VoIP forum about FS conference calls and even
>>     provided the  http://wiki.freeswitch.org/wiki/FS_weekly_2010_09_15
>>     link  (at
>>     the time). He replied back and told me that he doesn't know how to
>>     make his
>>     VoIP device to support a SIP URI dial into the conference line. 
>>
>> I'm not trying to be mean here, but honestly, if he can't dial any of
>> our ways of getting to the conference then he's *not ready for a VoIP
>> conference*!! We have many ways to connect:
>> PSTN: 1-919-386-990
>> SIP: sip:888 at conference.freeswitch.org
>> <mailto:sip%3A888 at conference.freeswitch.org>
>> GTalk: gtalk:conf+888 at conference.freeswitch.org
>> <mailto:gtalk%3Aconf%2B888 at conference.freeswitch.org>
>> Skype: call skype user "skypiax5"
>>
>> That's five different ways of connecting. I'm sorry, but that's more
>> than enough ways to get people to call a *VoIP* conference about
>> telecom software. If people are incapable of connecting from one of
>> these five ways then they are not yet ready for FreeSWITCH.
>>
>> I will keep working the wiki and making FreeSWITCH as easy as possible
>> for newbies. However, please keep in mind that those who don't have
>> telecom, networking, and/or VoIP experience are going to experience a
>> much steeper learning curve. That's just the nature of the beast. To
>> ease their transition maybe we could come up with a recommended
>> reading list? "VoIP For Dummies" kind of stuff maybe. If you guys have
>> any suggestions I'll put the on the wiki.
>>
>> -MC
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From anthony.minessale at gmail.com  Fri Oct  8 20:34:47 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Oct 2010 22:34:47 -0500
Subject: [Freeswitch-users] iLBC codec generating only noise.
In-Reply-To: <AANLkTinz4JEfyGnjU-4nneOFg=A0qXHjVw9xE=jp2qSo@mail.gmail.com>
References: <AANLkTinz4JEfyGnjU-4nneOFg=A0qXHjVw9xE=jp2qSo@mail.gmail.com>
Message-ID: <AANLkTinvkxRaS+Wmso1vubuH3Pj4Mib96B32Bx+sqR7H@mail.gmail.com>

Try latest
On Oct 8, 2010 2:28 PM, "xuyan yang" <xyangni at gmail.com> wrote:
> Hi,
>
> I am trying to use iLBC codec with both eyebeam and some iphone client.
When
> a ivr is called, the client can here system voice and make dtmf input. but
> the voice recorded from client's microphone is only noise.
> The call between 2 clients also have such problem.
>
> Sometimes, FS may even got crashed with the following information:
>
> 2010-10-08 19:56:21.343750 [INFO] switch_rtp.c:2527 Auto Changing port
from
> 10.20.132.244:18570 to 8
> 2.132.139.197:19536
> alloc: asked for negative size -2147483648
>
> in var.xml, I have added iLBC instead of iLBC at 30i, because iLBC at 30i is
> ignored during codec negotiation.
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h
,G7221 at 16000h
> ,G722,speex at 8000h@20i,iLBC,PCMU,PCMA,GSM"/>
>
> The git version last week is used in my test. Is there anything wrong with
> my setup? Thanks.
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From Nabble at slickdeals.endjunk.com  Fri Oct  8 20:39:14 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 8 Oct 2010 20:39:14 -0700 (PDT)
Subject: [Freeswitch-users] Get on IRC day!!
In-Reply-To: <AANLkTi=GSO2YDqjs9O=1ib=roP8nsbS6C9x6n8kQMkgz@mail.gmail.com>
References: <AANLkTimdGX0ganMLyd3uTD3Ux1spa5xx8mShuV9DEww9@mail.gmail.com>
	<1286568418994-5616330.post@n2.nabble.com>
	<AANLkTi=GSO2YDqjs9O=1ib=roP8nsbS6C9x6n8kQMkgz@mail.gmail.com>
Message-ID: <1286595554559-5617237.post@n2.nabble.com>



mercutioviz wrote:
> If someone stays away from FreeSWITCH solely because they don't like
> our combination of mailing list, IRC, and wiki then so be it. We respect
> their decision.
> 
> <snip>
> 
> That's five different ways of connecting. I'm sorry, but that's more than
> enough ways to get people to call a *VoIP* conference about telecom
> software. If people are incapable of connecting from one of these five
> ways
> then they are not yet ready for FreeSWITCH.
Whether these FS wannabees are ready or not isn't for me to judge. But, a PR
with attitudes like above will likely to make wannabees and/or newcomers
uneasy and if not unwelcome, AFAIC. It sounded like as if FS doesn't need
you, but if you can make it to reach FS then you are welcome (because you
have passed the 1st hindrance), especially to attend FS conference room.


I respect your honesty. I will encourage those answering newbie questions to
> be more explicit wherever possible. Just understand that everyone on this
> list and in IRC is an unpaid volunteer, giving of their time freely. In
> some
> cases an expert has to make a choice: give a brief answer or give no
> answer.
> Given the option, I think even a newbie would appreciate the quick answer
> as
> opposed to silence on the wire.
I reckon we all do understand that our present here to help each other is on
a voluntarily basis. As w.r.t either give a brief answer or give no answer,
my opinion is silence is golden. As a newbie here, I may not be of any help.
But, there were several occasions that I had to retreat from trying to help
even though I had the answers. The reason was I simply didn't have enough
time to spare and I certainly wouldn't like to give brief answers that may
and/or may not lead to nowhere to resolve the issues. One thing we all need
to remember is the help one provides here isn't only for the current
readers, but also will be for future readers who need it, too. So, a more
elaborate answer will certainly make Perhaps, I shall not do that in the
future and just rush to lend my hands as my allotted time will allow.



We spent a ton of time on the FreeSWITCH book, which contains a lot of the
> hand-holding newbies really need. It's relatively inexpensive and you can
> get just the PDF if you like. (If you're destitute then you can probably
> borrow the book from someone who has it.)
I agree with you on this. The FS book is of invaluable to any FS users.
Those who are in the US and EU shouldn't have a problem to order the book
and/or e-book. But, those who live in some 3rd world countries may have
problems, i.e. their credit cards won't work (even to pay paypal), shipment
doesn't arrive, etc. In the long run, it will be a lot more headaches for
those who live in some 3rd world countries trying to order things from other
countries.



However, please keep in mind that those who don't have telecom,
> networking, and/or VoIP experience are going to experience a much steeper
> learning curve. That's just the nature of the beast.
I completely agree with you on this. Let's also not forget that, unlike
asterisk, FS is a huge software that can do more and can also be complicated
to newbies with a much steeper learning curve.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Get-on-IRC-day-tp5615890p5617237.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From hesser4900 at gmail.com  Fri Oct  8 11:32:56 2010
From: hesser4900 at gmail.com (Holger Esser)
Date: Fri, 8 Oct 2010 13:32:56 -0500
Subject: [Freeswitch-users] Setting origination_caller_id_number
Message-ID: <AANLkTimWD5go=yTpC5DkHx7b8Z+0PbJMLTG4WoNsMTW9@mail.gmail.com>

Hi,

Whenever I set the origination_caller_is_number in my dial request like this
for a local or external context,
perl http.pl originate
{origination_caller_id_number=5xxxxxxxxxsofia/internal/1 at 148.181.145.1485000
the caller id is set.
Whenever I use the gateway, it is no longer working and the username in the
gateway becomes the ani number in the SIP trace.
perl http.pl originate
{origination_caller_id_number=972xxxxxxx}sofia/gateway/teliax/4695xxxxxxx
5000

Any ideas?

Thx
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From chenzhanping at gmail.com  Fri Oct  8 18:12:18 2010
From: chenzhanping at gmail.com (=?GB2312?B?1bzGvbPC?=)
Date: Sat, 9 Oct 2010 09:12:18 +0800
Subject: [Freeswitch-users] Question about mod_nibblebill.
Message-ID: <AANLkTimgF20e6Qk4QrY3rrA-6_uh1tNC=SVNc7-eBs1A@mail.gmail.com>

Hello, I have a question about the mod_nibblebill module of freeswitch.


I can use mod_nibblebill to billing. But I found that his billing was not
what I want.


I want to achieve in this way: every 60 seconds for an interval of less than
60 seconds, 60 seconds to install basis, charging $ 0.1 for each billing
interval. For example: the user dials 45 seconds, charging $ 0.1. If user
dials 72 seconds, charging $ 0.2.


My current configuration in dialplain\default.xml is as follows:

<action application="bridge"
data="{enable_heartbeat_events=60,nibble_rate=0.1,nibble_account=1000}user/${dialed_extension}@${domain_name}"/>


May I ask, mod_nibblebill module can not achieve my desired function, if
realized, how can I configure?

Thank you.
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From Holger.Esser at Convergys.com  Fri Oct  8 13:32:46 2010
From: Holger.Esser at Convergys.com (Esser, Holger)
Date: Fri, 8 Oct 2010 15:32:46 -0500
Subject: [Freeswitch-users] origination_caller_is_number
Message-ID: <0FF607C13B7F7A42B5B80DA9EB204C9109D44BD2D8@srv-ex01-dal.intervoice.int>

Hi,

Whenever I set the origination_caller_is_number in my dial request like this for a local or external context,

perl http.pl<http://http.pl/> originate {origination_caller_id_number=5xxxxxxxxxsofia/internal/1 at 148.181.145.148<mailto:1 at 148.181.145.148> 5000 the caller id is set.

Whenever I use the gateway, it is no longer working and the username in the gateway becomes the ani number in the SIP trace.
perl http.pl<http://http.pl/> originate {origination_caller_id_number=972xxxxxxx}sofia/gateway/teliax/4695xxxxxxx 5000

Any ideas?

Thx


[cid:image001.gif at 01CB66FE.0ECB1E30]

Holger Esser
Staff Engineer, Continuing Engineering / RTM
Convergys Corporation

office 972-454-8167

www.convergys.com<http://www.convergys.com> | www.intervoice.com<http://www.intervoice.com/>




________________________________
This e-mail transmission may contain information that is proprietary, privileged and/or confidential and is intended exclusively for the person(s) to whom it is addressed. Any use, copying, retention or disclosure by any person other than the intended recipient or the intended recipient's designees is strictly prohibited. If you are the intended recipient, you must treat the information in confidence and in accordance with all laws related to the privacy and confidentiality of such information. If you are not the intended recipient or their designee, please notify the sender immediately by return e-mail and delete all copies of this email, including all attachments.

Intervoice, Inc. 17811 Waterview Parkway Dallas, TX 75252 USA

Intervoice Limited, Registered in England and Wales with number 2601740, 50 Park Road, Gatley, Cheshire, SK8 4HZ. VAT Number: 560421375

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From excelsio at gmx.com  Fri Oct  8 22:46:24 2010
From: excelsio at gmx.com (Michael Baye)
Date: Sat, 09 Oct 2010 07:46:24 +0200
Subject: [Freeswitch-users] MIKEY-Support again
Message-ID: <20101009060100.17320@gmx.com>

Hi,

I was looking for MIKEY support in freeswitch and found an old thread: http://lists.freeswitch.org/pipermail/freeswitch-users/2008-January/029829.html
Here, Brian wrote:
"Ed, Awesome. Looks like a Mikey lib will have to be created since the only libmikey out there is GPL. Unless someone wants to talk them into releasing it LGPL? I have talked to projects that are libraries in the past and asked if they would release LGPL and most have done so... It never hurts to ask." 

Well, looking at http://www.minisip.org/ it is said: 
"The source code is available as a number of libraries under the GNU Lesser General Public License (LGPL) and applications under the GNU General Public Licence (GPL)."

Looking at svn://svn.minisip.org/minisip/trunk/libmikey/libmikey.spec only GPL is mentioned. Has anyone asked them since then if they would release it as LGPL?

Why wouldn?t it possible to add an own branch with an GPL only libmikey extension within freeswitch?

Sincerely yours,
Michael
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From slim at thegreek.com  Fri Oct  8 14:07:58 2010
From: slim at thegreek.com (Jeroen C. van Gelderen)
Date: Fri, 8 Oct 2010 16:07:58 -0500
Subject: [Freeswitch-users] FreeTDM Partial Spans?
In-Reply-To: <AANLkTinAp=RSSLqPi16gejMoGBihg7YnnB2wj0kz7vJC@mail.gmail.com>
References: <8A7648CAD02A4BEBB8B4BF15A8733F37@mbnet.local><AANLkTinK+1i4s+tVqLMua6aNd_u+M9Fdx9_5pYxcnUrm@mail.gmail.com>
	<AANLkTinAp=RSSLqPi16gejMoGBihg7YnnB2wj0kz7vJC@mail.gmail.com>
Message-ID: <D6318F98C2AF4641B8AA896EF9B67588@mbnet.local>

Very nice. Thanks for the elaborate answer.

Cheers,
-Slim
--
Jeroen C. "Slim" van Gelderen

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moises
Silva
Sent: Saturday, October 02, 2010 10:35
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FreeTDM Partial Spans?

Btw, I hope you don't mind, I started a Mailing list FAQ section in
the FreeTDM wiki page and added your question there (and some more
information):

http://wiki.freeswitch.org/wiki/FreeTDM#Mailing_List_FAQ

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com

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



From thedjallen at gmail.com  Thu Oct  7 22:32:59 2010
From: thedjallen at gmail.com (David Allen)
Date: Fri, 08 Oct 2010 16:32:59 +1100
Subject: [Freeswitch-users] Routing DIDs to Dynamically Registered SIP User
Message-ID: <4CAEAD0B.8020704@gmail.com>

  Hi,

I'm trying to send multiple Direct Indial Numbers down to a dynamically 
registered SIP User. I need to ensure that both the To and Target URI 
contain the direct Indial number. I'm able to modify the SIP TO Header 
of a call that is sent to them like below:

<action application="bridge" 
data="sofia/external/56778977%${domain}^61390009000"/>

which sends the request as:

    
------------------------------------------------------------------------

    INVITE sip:56778977 at 192.168.22.2:5061 SIP/2.0

    Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j

    Max-Forwards: 69

    From: "0390001000" <sip:0390001000 at 192.168.2.200>;tag=K4HHaZ9v1H07Q

    To: <sip:61390009000 at 192.168.2.200>

  But in order to maintain compatability with a number of PBX's/VoIP 
devices on the market, I need to be able to send the invite to the 
dynamically registered SIP user, however have it set the Target URI and 
To as the same contact number like below:

    
------------------------------------------------------------------------

    INVITE sip:61390009000 at 192.168.22.2:5061 SIP/2.0

    Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j

    Max-Forwards: 69

    From: "0390001000" <sip:0390001000 at 192.168.2.200>;tag=K4HHaZ9v1H07Q

    To: <sip:61390009000 at 192.168.22.2:5061>

Is there a way to do this setting via variables? I can't seem to find 
any details for it.

Thanks

David


From thedjallen at gmail.com  Thu Oct  7 22:44:36 2010
From: thedjallen at gmail.com (David Allen)
Date: Fri, 08 Oct 2010 16:44:36 +1100
Subject: [Freeswitch-users] Routing DIDs to Dynamically Registered SIP User
Message-ID: <4CAEAFC4.3040504@gmail.com>

  Hi,

I'm trying to send multiple Direct Indial Numbers down to a dynamically 
registered SIP User. I need to ensure that both the To and Target URI 
contain the direct Indial number. I'm able to modify the SIP TO Header 
of a call that is sent to them like below:

<action application="bridge" 
data="sofia/external/56778977%${domain}^61390009000"/>

which sends the request as:

    ------------------------------------------------------------------------

    INVITE sip:56778977 at 192.168.22.2:5061 SIP/2.0

    Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j

    Max-Forwards: 69

    From: "0390001000" <sip:0390001000 at 192.168.2.200>;tag=K4HHaZ9v1H07Q

    To: <sip:61390009000 at 192.168.2.200>

  But in order to maintain compatability with a number of PBX's/VoIP 
devices on the market, I need to be able to send the invite to the 
dynamically registered SIP user, however have it set the Target URI and 
To as the same contact number like below:

    ------------------------------------------------------------------------

    INVITE sip:61390009000 at 192.168.22.2:5061 SIP/2.0

    Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j

    Max-Forwards: 69

    From: "0390001000" <sip:0390001000 at 192.168.2.200>;tag=K4HHaZ9v1H07Q

    To: <sip:61390009000 at 192.168.22.2:5061>

Is there a way to do this setting via variables? I can't seem to find 
any details for it.

Thanks

David


From jody.rudolph at gmail.com  Sat Oct  9 00:13:58 2010
From: jody.rudolph at gmail.com (Jody Rudolph)
Date: Sat, 9 Oct 2010 03:13:58 -0400
Subject: [Freeswitch-users] Sharing storage between servers
Message-ID: <AANLkTinMs5D-0ip7pj6TLqUvrrt9pbF152s+eNdYi2yQ@mail.gmail.com>

I am curious as to just how far you can take sharing disk storage for the
purpose of clustering. Is anyone doing this with the voicemail storage
directories? Is it possible to share the SQLite internal databases to avoid
resorting to ODBC? I realize that isn't likely, but I have access to some
high performance SAN hardware and want to take the most advantage possible.


Thanks,
Jody Rudolph
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From yehavi.bourvine at gmail.com  Sat Oct  9 00:49:51 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sat, 9 Oct 2010 09:49:51 +0200
Subject: [Freeswitch-users] Sharing storage between servers
In-Reply-To: <AANLkTinMs5D-0ip7pj6TLqUvrrt9pbF152s+eNdYi2yQ@mail.gmail.com>
References: <AANLkTinMs5D-0ip7pj6TLqUvrrt9pbF152s+eNdYi2yQ@mail.gmail.com>
Message-ID: <AANLkTi=8kH5-c8pL1K_KRsNBo2gEO4y+24iwCk=AkJ3D@mail.gmail.com>

Hi,

  We use a NAS server to share the voicemail between two servers (one is FS,
the other is WEB interface we wrote to handle  voicemail via WEB).
  For the database: we use MySQL with replication.

                          Regards, __Yehavi:

 2010/10/9 Jody Rudolph <jody.rudolph at gmail.com>

> I am curious as to just how far you can take sharing disk storage for the
> purpose of clustering. Is anyone doing this with the voicemail storage
> directories? Is it possible to share the SQLite internal databases to avoid
> resorting to ODBC? I realize that isn't likely, but I have access to some
> high performance SAN hardware and want to take the most advantage possible.
>
>
> Thanks,
> Jody Rudolph
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From christian.loeschenkohl at xpirio.com  Sat Oct  9 01:07:58 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Sat, 09 Oct 2010 10:07:58 +0200
Subject: [Freeswitch-users] event_socket - sendevent PRESENCE_IN
Message-ID: <4CB022DE.2020708@xpirio.com>

hello

has this event changed lately?
if i send this event with actual fs versions (latest git) the following notify sip body is send to our snom phones

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="2" state="partial" entity="sip:xxx at domain.com">
</dialog-info>

this function is used to switch lights on an off (phone has an active subscription for xxx at domain.com)

-----------------

in the versions used bevor something like this was send

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="" state="full" entity="sip:xxx at domain.com">
<dialog id="" direction="recipient">
<state>confirmed</state>
</dialog>
</dialog-info>

this works as phones get the state of the led correctly

-----------------

the normal notify (a calls b and c is subscribed with state of a or b) works as expected (sip body of notify packet
is complete).

br

-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From xyangni at gmail.com  Sat Oct  9 05:43:29 2010
From: xyangni at gmail.com (xuyan yang)
Date: Sat, 9 Oct 2010 20:43:29 +0800
Subject: [Freeswitch-users] iLBC codec generating only noise.
In-Reply-To: <AANLkTinvkxRaS+Wmso1vubuH3Pj4Mib96B32Bx+sqR7H@mail.gmail.com>
References: <AANLkTinz4JEfyGnjU-4nneOFg=A0qXHjVw9xE=jp2qSo@mail.gmail.com>
	<AANLkTinvkxRaS+Wmso1vubuH3Pj4Mib96B32Bx+sqR7H@mail.gmail.com>
Message-ID: <AANLkTimmMM8Ua9AP9BwXU0yDfP9FReK2e9w-iMRiMVFu@mail.gmail.com>

Hi Anthony,

Tried the latest, it works when both lag A and B are using iLBC. But there
are still some problems:
1, when lag A use PCMU and lag B use iLBC or the reverse, the PCMU side hear
only noise even if it is muted.

2, when added "iLBC at 30i" to codec list, it is ignored. traced the debug,
iLBC is not compared as a choice. So adding only "iLBC" should be used as
mitigation.




On Sat, Oct 9, 2010 at 11:34 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Try latest
> On Oct 8, 2010 2:28 PM, "xuyan yang" <xyangni at gmail.com> wrote:
> > Hi,
> >
> > I am trying to use iLBC codec with both eyebeam and some iphone client.
> When
> > a ivr is called, the client can here system voice and make dtmf input.
> but
> > the voice recorded from client's microphone is only noise.
> > The call between 2 clients also have such problem.
> >
> > Sometimes, FS may even got crashed with the following information:
> >
> > 2010-10-08 19:56:21.343750 [INFO] switch_rtp.c:2527 Auto Changing port
> from
> > 10.20.132.244:18570 to 8
> > 2.132.139.197:19536
> > alloc: asked for negative size -2147483648
> >
> > in var.xml, I have added iLBC instead of iLBC at 30i, because iLBC at 30i is
> > ignored during codec negotiation.
> > <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h
> ,G7221 at 16000h
> > ,G722,speex at 8000h@20i,iLBC,PCMU,PCMA,GSM"/>
> >
> > The git version last week is used in my test. Is there anything wrong
> with
> > my setup? Thanks.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From neilp at cs.stanford.edu  Sat Oct  9 06:34:58 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Sat, 9 Oct 2010 19:04:58 +0530
Subject: [Freeswitch-users] outbound calling error: multiple PRIs with
 Sangoma + FS
In-Reply-To: <AANLkTin01O6WPqXfXN+aiaPcOd+Ww1HiFwzcbUEeMz2Z@mail.gmail.com>
References: <AANLkTimn7P64Exf9JTwdLZB8Vam8wBLuvrLS5MY+xAwT@mail.gmail.com>
	<AANLkTin01O6WPqXfXN+aiaPcOd+Ww1HiFwzcbUEeMz2Z@mail.gmail.com>
Message-ID: <AANLkTim-UWMzVNkdnTyZnESkKTou=VqN0Y3_UU=B80oX@mail.gmail.com>

Hi Moises,

Thanks for the heads-up about freetdm. I have installed the freetdm module
on a freshly updated FS instance, and have also updated my wanpipe drivers
to the latest. Here is my freetdm.conf:

[span wanpipe wp1]
trunk_type => e1
group=>grp1
b-channel => 1:1-15
b-channel => 1:17-31
d-channel => 1:16

[span wanpipe wp2]
trunk_type => e1
group=>grp2
b-channel => 2:1-15
b-channel => 2:17-31
d-channel => 2:16

Here's my freetdm.conf.xml:

<configuration name="freetdm.conf" description="FreeTDM Configuration">
  <settings>
    <param name="debug" value="0"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
    <!--<param name="enable-analog-option" value="call-swap"/>-->
    <!--<param name="enable-analog-option" value="3-way"/>-->
  </settings>
   <pri_spans>
     <span name="wp1">
       <!-- Log Levels: none, alert, crit, err, warning, notice, info, debug
-->
       <param name="q921loglevel" value="alert"/>
       <param name="q931loglevel" value="alert"/>
       <param name="mode" value="user"/>
       <param name="dialect" value="5ess"/>
       <param name="dialplan" value="XML"/>
       <param name="context" value="default"/>
     </span>
     <span name="wp2">
       <param name="q921loglevel" value="alert"/>
       <param name="q931loglevel" value="alert"/>
       <param name="mode" value="user"/>
       <param name="dialect" value="5ess"/>
       <param name="dialplan" value="XML"/>
       <param name="context" value="default"/>
     </span>
   </pri_spans>
  <!-- one entry here per openzap span -->
  <analog_spans>
    <span id="1">
      <!--<param name="hold-music" value="$${moh_uri}"/>-->
      <!--<param name="enable-analog-option" value="call-swap"/>-->
      <!--<param name="enable-analog-option" value="3-way"/>-->
      <param name="tonegroup" value="us"/>
      <param name="digit-timeout" value="2000"/>
      <param name="max-digits" value="11"/>
      <param name="dialplan" value="XML"/>
      <param name="context" value="default"/>
      <param name="enable-callerid" value="true"/>
      <!-- regex to stop dialing when it matches -->
      <!--<param name="dial-regex" value="5555"/>-->
      <!-- regex to stop dialing when it does not match -->
      <!--<param name="fail-dial-regex" value="^5"/>-->
    </span>
  </analog_spans>
</configuration>

And pasted below is the errors I am getting when I fire up FS. Note
especially the bolded parts:

2010-10-09 18:28:58.364527 [INFO] ftdm_io.c:4288 Loading IO from
/usr/local/freeswitch/mod/ftmod_wanpipe.so [wanpipe]
2010-10-09 18:28:58.364647 [INFO] ftdm_io.c:604 Auto-loaded I/O module
'wanpipe'
2010-10-09 18:28:58.364845 [INFO] ftmod_wanpipe.c:363 [s1c1][1:1] configured
wanpipe device s1c1 as FreeTDM channel 1:1 fd:43 DTMF: hardware
2010-10-09 18:28:58.364921 [INFO] ftmod_wanpipe.c:363 [s1c2][1:2] configured
wanpipe device s1c2 as FreeTDM channel 1:2 fd:46 DTMF: hardware
2010-10-09 18:28:58.364991 [INFO] ftmod_wanpipe.c:363 [s1c3][1:3] configured
wanpipe device s1c3 as FreeTDM channel 1:3 fd:49 DTMF: hardware
2010-10-09 18:28:58.365055 [INFO] ftmod_wanpipe.c:363 [s1c4][1:4] configured
wanpipe device s1c4 as FreeTDM channel 1:4 fd:52 DTMF: hardware
2010-10-09 18:28:58.365123 [INFO] ftmod_wanpipe.c:363 [s1c5][1:5] configured
wanpipe device s1c5 as FreeTDM channel 1:5 fd:55 DTMF: hardware
2010-10-09 18:28:58.365198 [INFO] ftmod_wanpipe.c:363 [s1c6][1:6] configured
wanpipe device s1c6 as FreeTDM channel 1:6 fd:58 DTMF: hardware
2010-10-09 18:28:58.365268 [INFO] ftmod_wanpipe.c:363 [s1c7][1:7] configured
wanpipe device s1c7 as FreeTDM channel 1:7 fd:61 DTMF: hardware
2010-10-09 18:28:58.365336 [INFO] ftmod_wanpipe.c:363 [s1c8][1:8] configured
wanpipe device s1c8 as FreeTDM channel 1:8 fd:64 DTMF: hardware
2010-10-09 18:28:58.365400 [INFO] ftmod_wanpipe.c:363 [s1c9][1:9] configured
wanpipe device s1c9 as FreeTDM channel 1:9 fd:67 DTMF: hardware
2010-10-09 18:28:58.365463 [INFO] ftmod_wanpipe.c:363 [s1c10][1:10]
configured wanpipe device s1c10 as FreeTDM channel 1:10 fd:70 DTMF: hardware
2010-10-09 18:28:58.365620 [INFO] ftmod_wanpipe.c:363 [s1c11][1:11]
configured wanpipe device s1c11 as FreeTDM channel 1:11 fd:41 DTMF: hardware
2010-10-09 18:28:58.365685 [INFO] ftmod_wanpipe.c:363 [s1c12][1:12]
configured wanpipe device s1c12 as FreeTDM channel 1:12 fd:75 DTMF: hardware
2010-10-09 18:28:58.365791 [INFO] ftmod_wanpipe.c:363 [s1c13][1:13]
configured wanpipe device s1c13 as FreeTDM channel 1:13 fd:78 DTMF: hardware
2010-10-09 18:28:58.365861 [INFO] ftmod_wanpipe.c:363 [s1c14][1:14]
configured wanpipe device s1c14 as FreeTDM channel 1:14 fd:81 DTMF: hardware
2010-10-09 18:28:58.365926 [INFO] ftmod_wanpipe.c:363 [s1c15][1:15]
configured wanpipe device s1c15 as FreeTDM channel 1:15 fd:84 DTMF: hardware
2010-10-09 18:28:58.366117 [INFO] ftmod_wanpipe.c:363 [s1c16][1:17]
configured wanpipe device s1c17 as FreeTDM channel 1:16 fd:87 DTMF: hardware
2010-10-09 18:28:58.366184 [INFO] ftmod_wanpipe.c:363 [s1c17][1:18]
configured wanpipe device s1c18 as FreeTDM channel 1:17 fd:90 DTMF: hardware
2010-10-09 18:28:58.366250 [INFO] ftmod_wanpipe.c:363 [s1c18][1:19]
configured wanpipe device s1c19 as FreeTDM channel 1:18 fd:93 DTMF: hardware
2010-10-09 18:28:58.366325 [INFO] ftmod_wanpipe.c:363 [s1c19][1:20]
configured wanpipe device s1c20 as FreeTDM channel 1:19 fd:96 DTMF: hardware
2010-10-09 18:28:58.366441 [INFO] ftmod_wanpipe.c:363 [s1c20][1:21]
configured wanpipe device s1c21 as FreeTDM channel 1:20 fd:99 DTMF: hardware
2010-10-09 18:28:58.366505 [INFO] ftmod_wanpipe.c:363 [s1c21][1:22]
configured wanpipe device s1c22 as FreeTDM channel 1:21 fd:102 DTMF:
hardware
2010-10-09 18:28:58.366571 [INFO] ftmod_wanpipe.c:363 [s1c22][1:23]
configured wanpipe device s1c23 as FreeTDM channel 1:22 fd:105 DTMF:
hardware
2010-10-09 18:28:58.366645 [INFO] ftmod_wanpipe.c:363 [s1c23][1:24]
configured wanpipe device s1c24 as FreeTDM channel 1:23 fd:108 DTMF:
hardware
2010-10-09 18:28:58.366707 [INFO] ftmod_wanpipe.c:363 [s1c24][1:25]
configured wanpipe device s1c25 as FreeTDM channel 1:24 fd:111 DTMF:
hardware
2010-10-09 18:28:58.366775 [INFO] ftmod_wanpipe.c:363 [s1c25][1:26]
configured wanpipe device s1c26 as FreeTDM channel 1:25 fd:114 DTMF:
hardware
2010-10-09 18:28:58.366898 [INFO] ftmod_wanpipe.c:363 [s1c26][1:27]
configured wanpipe device s1c27 as FreeTDM channel 1:26 fd:40 DTMF: hardware
2010-10-09 18:28:58.366962 [INFO] ftmod_wanpipe.c:363 [s1c27][1:28]
configured wanpipe device s1c28 as FreeTDM channel 1:27 fd:119 DTMF:
hardware
2010-10-09 18:28:58.367027 [INFO] ftmod_wanpipe.c:363 [s1c28][1:29]
configured wanpipe device s1c29 as FreeTDM channel 1:28 fd:122 DTMF:
hardware
2010-10-09 18:28:58.367091 [INFO] ftmod_wanpipe.c:363 [s1c29][1:30]
configured wanpipe device s1c30 as FreeTDM channel 1:29 fd:125 DTMF:
hardware
2010-10-09 18:28:58.367168 [INFO] ftmod_wanpipe.c:363 [s1c30][1:31]
configured wanpipe device s1c31 as FreeTDM channel 1:30 fd:128 DTMF:
hardware
*2010-10-09 18:28:58.367265 [ERR] ftmod_wanpipe.c:238 Failed to open wanpipe
device span 1 channel 16
2010-10-09 18:28:58.367305 [ERR] ftdm_io.c:3949 1:Failed to configure
span2010-10-09 18:28:58.367470 [INFO] *ftmod_wanpipe.c:363 [s2c1][2:1]
configured wanpipe device s2c1 as FreeTDM channel 2:1 fd:131 DTMF: hardware
2010-10-09 18:28:58.367556 [INFO] ftmod_wanpipe.c:363 [s2c2][2:2] configured
wanpipe device s2c2 as FreeTDM channel 2:2 fd:134 DTMF: hardware
2010-10-09 18:28:58.367639 [INFO] ftmod_wanpipe.c:363 [s2c3][2:3] configured
wanpipe device s2c3 as FreeTDM channel 2:3 fd:137 DTMF: hardware
2010-10-09 18:28:58.367708 [INFO] ftmod_wanpipe.c:363 [s2c4][2:4] configured
wanpipe device s2c4 as FreeTDM channel 2:4 fd:140 DTMF: hardware
2010-10-09 18:28:58.367778 [INFO] ftmod_wanpipe.c:363 [s2c5][2:5] configured
wanpipe device s2c5 as FreeTDM channel 2:5 fd:143 DTMF: hardware
2010-10-09 18:28:58.367850 [INFO] ftmod_wanpipe.c:363 [s2c6][2:6] configured
wanpipe device s2c6 as FreeTDM channel 2:6 fd:146 DTMF: hardware
2010-10-09 18:28:58.367917 [INFO] ftmod_wanpipe.c:363 [s2c7][2:7] configured
wanpipe device s2c7 as FreeTDM channel 2:7 fd:149 DTMF: hardware
2010-10-09 18:28:58.367994 [INFO] ftmod_wanpipe.c:363 [s2c8][2:8] configured
wanpipe device s2c8 as FreeTDM channel 2:8 fd:152 DTMF: hardware
2010-10-09 18:28:58.368059 [INFO] ftmod_wanpipe.c:363 [s2c9][2:9] configured
wanpipe device s2c9 as FreeTDM channel 2:9 fd:155 DTMF: hardware
2010-10-09 18:28:58.368128 [INFO] ftmod_wanpipe.c:363 [s2c10][2:10]
configured wanpipe device s2c10 as FreeTDM channel 2:10 fd:158 DTMF:
hardware
2010-10-09 18:28:58.368201 [INFO] ftmod_wanpipe.c:363 [s2c11][2:11]
configured wanpipe device s2c11 as FreeTDM channel 2:11 fd:161 DTMF:
hardware
2010-10-09 18:28:58.368269 [INFO] ftmod_wanpipe.c:363 [s2c12][2:12]
configured wanpipe device s2c12 as FreeTDM channel 2:12 fd:164 DTMF:
hardware
2010-10-09 18:28:58.368345 [INFO] ftmod_wanpipe.c:363 [s2c13][2:13]
configured wanpipe device s2c13 as FreeTDM channel 2:13 fd:167 DTMF:
hardware
2010-10-09 18:28:58.368411 [INFO] ftmod_wanpipe.c:363 [s2c14][2:14]
configured wanpipe device s2c14 as FreeTDM channel 2:14 fd:170 DTMF:
hardware
2010-10-09 18:28:58.368478 [INFO] ftmod_wanpipe.c:363 [s2c15][2:15]
configured wanpipe device s2c15 as FreeTDM channel 2:15 fd:173 DTMF:
hardware
2010-10-09 18:28:58.368589 [INFO] ftmod_wanpipe.c:363 [s2c16][2:17]
configured wanpipe device s2c17 as FreeTDM channel 2:16 fd:176 DTMF:
hardware
2010-10-09 18:28:58.368677 [INFO] ftmod_wanpipe.c:363 [s2c17][2:18]
configured wanpipe device s2c18 as FreeTDM channel 2:17 fd:179 DTMF:
hardware
2010-10-09 18:28:58.368744 [INFO] ftmod_wanpipe.c:363 [s2c18][2:19]
configured wanpipe device s2c19 as FreeTDM channel 2:18 fd:182 DTMF:
hardware
2010-10-09 18:28:58.368818 [INFO] ftmod_wanpipe.c:363 [s2c19][2:20]
configured wanpipe device s2c20 as FreeTDM channel 2:19 fd:185 DTMF:
hardware
2010-10-09 18:28:58.368884 [INFO] ftmod_wanpipe.c:363 [s2c20][2:21]
configured wanpipe device s2c21 as FreeTDM channel 2:20 fd:188 DTMF:
hardware
2010-10-09 18:28:58.368951 [INFO] ftmod_wanpipe.c:363 [s2c21][2:22]
configured wanpipe device s2c22 as FreeTDM channel 2:21 fd:191 DTMF:
hardware
2010-10-09 18:28:58.369018 [INFO] ftmod_wanpipe.c:363 [s2c22][2:23]
configured wanpipe device s2c23 as FreeTDM channel 2:22 fd:194 DTMF:
hardware
2010-10-09 18:28:58.369084 [INFO] ftmod_wanpipe.c:363 [s2c23][2:24]
configured wanpipe device s2c24 as FreeTDM channel 2:23 fd:197 DTMF:
hardware
2010-10-09 18:28:58.369150 [INFO] ftmod_wanpipe.c:363 [s2c24][2:25]
configured wanpipe device s2c25 as FreeTDM channel 2:24 fd:200 DTMF:
hardware
2010-10-09 18:28:58.369213 [INFO] ftmod_wanpipe.c:363 [s2c25][2:26]
configured wanpipe device s2c26 as FreeTDM channel 2:25 fd:203 DTMF:
hardware
2010-10-09 18:28:58.369278 [INFO] ftmod_wanpipe.c:363 [s2c26][2:27]
configured wanpipe device s2c27 as FreeTDM channel 2:26 fd:206 DTMF:
hardware
2010-10-09 18:28:58.369341 [INFO] ftmod_wanpipe.c:363 [s2c27][2:28]
configured wanpipe device s2c28 as FreeTDM channel 2:27 fd:209 DTMF:
hardware
2010-10-09 18:28:58.369406 [INFO] ftmod_wanpipe.c:363 [s2c28][2:29]
configured wanpipe device s2c29 as FreeTDM channel 2:28 fd:212 DTMF:
hardware
2010-10-09 18:28:58.369476 [INFO] ftmod_wanpipe.c:363 [s2c29][2:30]
configured wanpipe device s2c30 as FreeTDM channel 2:29 fd:215 DTMF:
hardware
2010-10-09 18:28:58.369548 [INFO] ftmod_wanpipe.c:363 [s2c30][2:31]
configured wanpipe device s2c31 as FreeTDM channel 2:30 fd:218 DTMF:
hardware
*2010-10-09 18:28:58.369640 [ERR] ftmod_wanpipe.c:238 Failed to open wanpipe
device span 2 channel 16
2010-10-09 18:28:58.369687 [ERR] ftdm_io.c:3949 2:Failed to configure
span2*010-10-09
18:28:58.369718 [INFO] ftdm_io.c:4213 Configured 60 channel(s)
2010-10-09 18:28:58.452329 [INFO] ftdm_io.c:4288 Loading IO from
/usr/local/freeswitch/mod/ftmod_sangoma_isdn.so [sangoma_isdn]
2010-10-09 18:28:58.452369 [INFO] ftmod_sangoma_isdn.c:851 Loading
ftmod_sangoma_isdn...
2010-10-09 18:28:58.452688 [INFO] ftmod_sangoma_isdn_stack_rcv.c:877
sng_isdn->

===========================================
=       Sangoma FreeTDM ISDN Library      =
=          Version:1.0.0               =
===========================================

2010-10-09 18:28:58.480173 [INFO] ftmod_sangoma_isdn_stack_rcv.c:874
sng_isdn->Successfully registered Stack Manager
2010-10-09 18:28:58.480421 [INFO] ftmod_sangoma_isdn_stack_rcv.c:874
sng_isdn->Successfully started a task for Stack Manager
2010-10-09 18:28:58.480476 [INFO] ftmod_sangoma_isdn_stack_rcv.c:874
sng_isdn->Successfully attached Stack Manager to it's task
2010-10-09 18:28:58.480529 [INFO] ftmod_sangoma_isdn_stack_rcv.c:874
sng_isdn->Successfully registered Call Control
2010-10-09 18:28:58.480611 [INFO] ftmod_sangoma_isdn_stack_rcv.c:874
sng_isdn->Successfully started a task for Call Control
2010-10-09 18:28:58.480657 [INFO] ftmod_sangoma_isdn_stack_rcv.c:874
sng_isdn->Successfully attached Call Control to it's task
2010-10-09 18:28:58.486424 [INFO] ftmod_sangoma_isdn_stack_rcv.c:874
sng_isdn->Successfully registered Q.930/Q.931
2010-10-09 18:28:58.486535 [INFO] ftmod_sangoma_isdn_stack_rcv.c:874
sng_isdn->Successfully started a task for Q.930/Q.931
2010-10-09 18:28:58.486607 [INFO] ftmod_sangoma_isdn_stack_rcv.c:874
sng_isdn->Successfully attached Q.930/Q.931 to it's task
2010-10-09 18:28:58.486658 [INFO] ftmod_sangoma_isdn_stack_rcv.c:874
sng_isdn->Successfully registered LAPD
2010-10-09 18:28:58.486720 [INFO] ftmod_sangoma_isdn_stack_rcv.c:874
sng_isdn->Successfully started a task for LAPD
2010-10-09 18:28:58.486772 [INFO] ftmod_sangoma_isdn_stack_rcv.c:874
sng_isdn->Successfully attached LAPD to it's task
2010-10-09 18:28:58.486823 [INFO] ftmod_sangoma_isdn_stack_rcv.c:874
sng_isdn->Successfully registered Physical Layer
2010-10-09 18:28:58.486881 [INFO] ftmod_sangoma_isdn_stack_rcv.c:874
sng_isdn->Successfully started a task for Physical Layer
2010-10-09 18:28:58.486930 [INFO] ftmod_sangoma_isdn_stack_rcv.c:874
sng_isdn->Successfully attached Physical Layer to it's task
2010-10-09 18:28:58.486986 [INFO] ftdm_io.c:4300 Loading SIG from
/usr/local/freeswitch/mod/ftmod_sangoma_isdn.so
2010-10-09 18:28:58.487035 [INFO] ftdm_io.c:4523 auto-loaded 'sangoma_isdn'
2010-10-09 18:28:58.487076 [INFO] ftmod_sangoma_isdn.c:791 Configuring
ftmod_sangoma_isdn span = wp1
2010-10-09 18:28:58.487311 [WARNING] ftmod_sangoma_isdn_cfg.c:262 Ignoring
unknown parameter q921loglevel
2010-10-09 18:28:58.487369 [WARNING] ftmod_sangoma_isdn_cfg.c:262 Ignoring
unknown parameter q931loglevel
2010-10-09 18:28:58.487425 [WARNING] ftmod_sangoma_isdn_cfg.c:262 Ignoring
unknown parameter mode
2010-10-09 18:28:58.487480 [WARNING] ftmod_sangoma_isdn_cfg.c:262 Ignoring
unknown parameter dialect
*2010-10-09 18:28:58.487517 [ERR] ftmod_sangoma_isdn_cfg.c:267 wp1:
switchtype not specified2010-10-09 18:28:58.487537 [ERR]
ftmod_sangoma_isdn.c:807 Failed to parse configuration*
2010-10-09 18:28:58.487578 [ERR] mod_freetdm.c:2478 Error configuring
Sangoma ISDN FreeTDM span 1
2010-10-09 18:28:58.487628 [INFO] ftmod_sangoma_isdn.c:791 Configuring
ftmod_sangoma_isdn span = wp2
2010-10-09 18:28:58.487869 [WARNING] ftmod_sangoma_isdn_cfg.c:262 Ignoring
unknown parameter q921loglevel
2010-10-09 18:28:58.487927 [WARNING] ftmod_sangoma_isdn_cfg.c:262 Ignoring
unknown parameter q931loglevel
2010-10-09 18:28:58.487984 [WARNING] ftmod_sangoma_isdn_cfg.c:262 Ignoring
unknown parameter mode
2010-10-09 18:28:58.488039 [WARNING] ftmod_sangoma_isdn_cfg.c:262 Ignoring
unknown parameter dialect
*2010-10-09 18:28:58.488076 [ERR] ftmod_sangoma_isdn_cfg.c:267 wp2:
switchtype not specified2010-10-09 18:28:58.488096 [ERR]
ftmod_sangoma_isdn.c:807 Failed to parse configuration
2010-10-09 18:28:58.488135 [ERR] mod_freetdm.c:2478 Error configuring
Sangoma ISDN FreeTDM span 2*

Your help is greatly appreciated,
Neil

On Fri, Oct 8, 2010 at 1:21 AM, Moises Silva <moises.silva at gmail.com> wrote:

> On Thu, Oct 7, 2010 at 6:52 AM, Neil Patel <neilp at cs.stanford.edu> wrote:
> > ...and here is the output from FS CLI. Note that the PRI line connected
> to
> > port 2 (g2) works great; it is g1 that is producing the
> NO_ROUTE_DESTINATION
> > error:
> >
> > freeswitch at otalo> originate openzap/smg_prid/a/958655XXXX at g1 &echo
> > 2010-10-07 18:08:25.273814 [WARNING] ozmod_sangoma_boost.c:344 TX EVENT:
> > CALL_START:(80) [w1g1] CSid=5 Seq=41 Cn=[N/A] Cd=[958655XXXX] Ci=[N/A]
> > Rdnis=[]
> > 2010-10-07 18:08:25.306941 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT
> > (N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=5 Seq=34
> > 2010-10-07 18:08:25.307953 [NOTICE] switch_channel.c:675 New Channel
> > OpenZAP/1:1/958655XXXX at g1 [c4305297-3a6e-4bd8-a25c-71e96da6b77d]
> > 2010-10-07 18:08:25.411216 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT
> > (N): CALL_STOPPED:(85) [w1g1] Rc=3 CSid=5 Seq=35
> > 2010-10-07 18:08:25.411216 [NOTICE] mod_openzap.c:1935 Hangup
> > OpenZAP/1:1/958655XXXX at g1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
> >
> > -ERR NO_ROUTE_DESTINATION
> >
> > 2010-10-07 18:08:25.412218 [NOTICE] switch_core_session.c:1188 Session 16
> > (OpenZAP/1:1/958655XXXX at g1) Ended
> > 2010-10-07 18:08:25.412218 [NOTICE] switch_core_session.c:1190 Close
> Channel
> > OpenZAP/1:1/958655XXXX at g1 [CS_DESTROY]
> > 2010-10-07 18:08:25.441373 [WARNING] sangoma_boost_client.c:221 TX EVENT
> > (N): CALL_STOPPED_ACK:(86) [w1g1] Rc=0 CSid=0 Seq=42
> > freeswitch at otalo> originate openzap/smg_prid/a/958655XXXX at g2 &echo
> > 2010-10-07 18:08:30.199197 [WARNING] ozmod_sangoma_boost.c:344 TX EVENT:
> > CALL_START:(80) [w1g1] CSid=6 Seq=43 Cn=[N/A] Cd=[958655XXXX] Ci=[N/A]
> > Rdnis=[]
> > 2010-10-07 18:08:30.263078 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT
> > (N): CALL_START_ACK:(81) [w2g1] Rc=0 CSid=6 Seq=36
> > 2010-10-07 18:08:30.264081 [NOTICE] switch_channel.c:675 New Channel
> > OpenZAP/1:31/958655XXXX at g2 [fe1dad03-1086-4d56-bc55-62563e170a78]
> > 2010-10-07 18:08:30.493273 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT:
> > CALL PROGRESS:(50) [w2g1] CSid=6 Seq=37 Cn=[N/A] Cd=[N/A] Ci=[N/A]
> Rdnis=[]
> > 2010-10-07 18:08:30.493273 [NOTICE] mod_openzap.c:1962 Pre-Answer
> > OpenZAP/1:31/958655XXXX at g2!
> >
> > +OK fe1dad03-1086-4d56-bc55-62563e170a78
> >
> > I don't notice much going on in /var/log/sangoma_mgd.log or
> > /var/log/messages, but then again I'm not sure what to look for. Any
> > guidance would be greatly appreciated!
> > Neil
> >
>
> Hi Neil,
>
> Any reason you are using deprecated software? FreeTDM replaced openzap
> and libsng_isdn replaced boost.
>
> See:
>
> http://wiki.freeswitch.org/wiki/FreeTDM
> http://wiki.sangoma.com/FreeTDM-Sangoma-ISDN-Library
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
> L3R 9R6 Canada
> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gerrit308 at gmail.com  Sat Oct  9 07:22:18 2010
From: gerrit308 at gmail.com (humbr)
Date: Sat, 9 Oct 2010 07:22:18 -0700 (PDT)
Subject: [Freeswitch-users] Get on IRC day!!
In-Reply-To: <AANLkTi=GSO2YDqjs9O=1ib=roP8nsbS6C9x6n8kQMkgz@mail.gmail.com>
References: <AANLkTimdGX0ganMLyd3uTD3Ux1spa5xx8mShuV9DEww9@mail.gmail.com>
	<1286568418994-5616330.post@n2.nabble.com>
	<AANLkTi=GSO2YDqjs9O=1ib=roP8nsbS6C9x6n8kQMkgz@mail.gmail.com>
Message-ID: <1286634138140-5618146.post@n2.nabble.com>


The Nabble thing is working fine for me, in fact it is why I am still able to
get answers. I won't deal with the flood of emails from a pure mailing list.
Nabble is to me a decent bridge between those wanting a Forum and the
reality of there not being one. IRC, well it depends who happens to be on
when you need help. Sometimes you win, sometimes you don't :-) Don't give,
try again.

As a long time computer user/coder/designer/hw fixer (wrote my first program
for an NCR315 in 1965 while in high school!) I am disturbed over the last 41
years to see the sheer laziness of people. With the www that has become much
worse. Everyone wants an instant solution to a complex problem, preferably
with no effort on their part. 

My reco is to provide an answer with an example or a link when possible. And
if you want to search for similar FS issues to your own, get on Nabble.

I bought the eBook. Love it. Thank you.

(FS on Dockstar with OpenWRT, awesome!!!!)
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Get-on-IRC-day-tp5615890p5618146.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From babak.freeswitch at gmail.com  Sat Oct  9 08:12:44 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sat, 9 Oct 2010 18:42:44 +0330
Subject: [Freeswitch-users] fs_cli
Message-ID: <AANLkTi=L4VArPuBSDV5hgGtJYr3Kbkoh1XjGTtPwM0BE@mail.gmail.com>

Hi
is it possible to execute and get result of an api like this
fs_cli.exe -H 192.168.11.30 -x "managed MY API" now I'm getting
-ERR no reply
but it works fine for show calls
thanx
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From mnhassan at usa.net  Sat Oct  9 08:30:09 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Sat, 9 Oct 2010 21:30:09 +0600
Subject: [Freeswitch-users] [Freeswitch-dev] Get on IRC day!!
In-Reply-To: <1286634138140-5618146.post@n2.nabble.com>
References: <AANLkTimdGX0ganMLyd3uTD3Ux1spa5xx8mShuV9DEww9@mail.gmail.com>
	<1286568418994-5616330.post@n2.nabble.com>
	<AANLkTi=GSO2YDqjs9O=1ib=roP8nsbS6C9x6n8kQMkgz@mail.gmail.com>
	<1286634138140-5618146.post@n2.nabble.com>
Message-ID: <AANLkTi=6JtgAi4NVJU8=j3o=3iaeKXEjKNNcvrMC1P4n@mail.gmail.com>

What's the problem with mailing lists? I think they are great!

Regards
HASSAN


On 2010-10-09, humbr <gerrit308 at gmail.com> wrote:
>
> The Nabble thing is working fine for me, in fact it is why I am still able
> to
> get answers. I won't deal with the flood of emails from a pure mailing list.
> Nabble is to me a decent bridge between those wanting a Forum and the
> reality of there not being one. IRC, well it depends who happens to be on
> when you need help. Sometimes you win, sometimes you don't :-) Don't give,
> try again.
>
> As a long time computer user/coder/designer/hw fixer (wrote my first program
> for an NCR315 in 1965 while in high school!) I am disturbed over the last 41
> years to see the sheer laziness of people. With the www that has become much
> worse. Everyone wants an instant solution to a complex problem, preferably
> with no effort on their part.
>
> My reco is to provide an answer with an example or a link when possible. And
> if you want to search for similar FS issues to your own, get on Nabble.
>
> I bought the eBook. Love it. Thank you.
>
> (FS on Dockstar with OpenWRT, awesome!!!!)
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Get-on-IRC-day-tp5615890p5618146.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device


From mnhassan at usa.net  Sat Oct  9 08:32:00 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Sat, 9 Oct 2010 21:32:00 +0600
Subject: [Freeswitch-users] fs_cli
In-Reply-To: <AANLkTi=L4VArPuBSDV5hgGtJYr3Kbkoh1XjGTtPwM0BE@mail.gmail.com>
References: <AANLkTi=L4VArPuBSDV5hgGtJYr3Kbkoh1XjGTtPwM0BE@mail.gmail.com>
Message-ID: <AANLkTinbHtywdVdPWEMNQRY+41WkmP-0+ZY8cpbp+7=a@mail.gmail.com>

Can you run that command from inside an fs_cli session?

Regards
HASSAN


On 2010-10-09, babak yakhchali <babak.freeswitch at gmail.com> wrote:
> Hi
> is it possible to execute and get result of an api like this
> fs_cli.exe -H 192.168.11.30 -x "managed MY API" now I'm getting
> -ERR no reply
> but it works fine for show calls
> thanx
>

-- 
Sent from my mobile device


From oseslija at gmail.com  Sat Oct  9 10:13:14 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Sat, 9 Oct 2010 19:13:14 +0200
Subject: [Freeswitch-users] FS Components
In-Reply-To: <COL109-W232E0055DD6B7F7E508364F4500@phx.gbl>
References: <COL109-W232E0055DD6B7F7E508364F4500@phx.gbl>
Message-ID: <AANLkTinHpaRkqu0iWokLiTsSTOGHbLFG=vxOoMGzMCAG@mail.gmail.com>

Hi,

read at http://wiki.freeswitch.org/wiki/Specsheet.


On Fri, Oct 8, 2010 at 10:24 AM, Abid Saleem <abid_freeswitch at live.com>wrote:

>  Hi,
>
> Could someone through some light on what are the functions FS can provide
> out of the below.
>
> SIP Server/Proxy Server
> Location/Presence Server
> Redirect Server
>
> Thanks.
>
> Regards
> -----------
> Abid Saleem
> Sr. Product Manager
> Terminus Technologies
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From max.clark at gmail.com  Sat Oct  9 10:15:44 2010
From: max.clark at gmail.com (Max Clark)
Date: Sat, 9 Oct 2010 10:15:44 -0700
Subject: [Freeswitch-users] Sharing storage between servers
In-Reply-To: <AANLkTinMs5D-0ip7pj6TLqUvrrt9pbF152s+eNdYi2yQ@mail.gmail.com>
References: <AANLkTinMs5D-0ip7pj6TLqUvrrt9pbF152s+eNdYi2yQ@mail.gmail.com>
Message-ID: <AANLkTimqC3u6RhcK10GCcxT49h=37z91Rb6Fb+WTMYjE@mail.gmail.com>

Jody,

Keep in mind that if you want to share disk (direct attached (DAS) or
SAN via Fiber Channel or iSCSI) you'll need to have a cluster
filesystem running on your hosts. NAS is a much simpler way to go if
you don't have experience with file system clustering.

-Max

On Sat, Oct 9, 2010 at 12:13 AM, Jody Rudolph <jody.rudolph at gmail.com> wrote:
> I am curious as to just how far you can take sharing disk storage for the
> purpose of clustering. Is anyone doing this with the voicemail storage
> directories? Is it possible to share the SQLite internal databases to avoid
> resorting to ODBC? I realize that isn't likely, but I have access to some
> high performance SAN hardware and want to take the most advantage possible.
>
>
> Thanks,
> Jody Rudolph
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Sat Oct  9 11:04:35 2010
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 9 Oct 2010 11:04:35 -0700
Subject: [Freeswitch-users] fs_cli
In-Reply-To: <AANLkTi=L4VArPuBSDV5hgGtJYr3Kbkoh1XjGTtPwM0BE@mail.gmail.com>
References: <AANLkTi=L4VArPuBSDV5hgGtJYr3Kbkoh1XjGTtPwM0BE@mail.gmail.com>
Message-ID: <AANLkTi=HeFmsJ3BVTdBkc9YHAK7Apgcivj-cX0degiDm@mail.gmail.com>

Yes, this kind of thing is completely legal. The -ERR is coming from the
managed called. Like Hassan said, make sure that you can run fs_cli.exe and
make the exact same call manually.

-MC

On Sat, Oct 9, 2010 at 8:12 AM, babak yakhchali
<babak.freeswitch at gmail.com>wrote:

> Hi
> is it possible to execute and get result of an api like this
> fs_cli.exe -H 192.168.11.30 -x "managed MY API" now I'm getting
> -ERR no reply
> but it works fine for show calls
> thanx
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Sat Oct  9 13:54:07 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 9 Oct 2010 15:54:07 -0500
Subject: [Freeswitch-users] Get on IRC day!!
In-Reply-To: <1286634138140-5618146.post@n2.nabble.com>
References: <AANLkTimdGX0ganMLyd3uTD3Ux1spa5xx8mShuV9DEww9@mail.gmail.com>
	<1286568418994-5616330.post@n2.nabble.com>
	<AANLkTi=GSO2YDqjs9O=1ib=roP8nsbS6C9x6n8kQMkgz@mail.gmail.com>
	<1286634138140-5618146.post@n2.nabble.com>
Message-ID: <AANLkTikY-GFgBCgUY67QgK-pTacz+0mo18QuQuLXpz7i@mail.gmail.com>

The irony is with every new graduating class of FS users we repeat this
discussion about how much the mailing list sucks VIA the mailing list!! =p

We are always open to suggestions but as always only suggestions that come
with action since we are already overloaded with what we are currently doing
on a daily basis.

Keep up the discussion but keep it positive......
On Oct 9, 2010 9:27 AM, "humbr" <gerrit308 at gmail.com> wrote:
>
> The Nabble thing is working fine for me, in fact it is why I am still able
to
> get answers. I won't deal with the flood of emails from a pure mailing
list.
> Nabble is to me a decent bridge between those wanting a Forum and the
> reality of there not being one. IRC, well it depends who happens to be on
> when you need help. Sometimes you win, sometimes you don't :-) Don't give,
> try again.
>
> As a long time computer user/coder/designer/hw fixer (wrote my first
program
> for an NCR315 in 1965 while in high school!) I am disturbed over the last
41
> years to see the sheer laziness of people. With the www that has become
much
> worse. Everyone wants an instant solution to a complex problem, preferably
> with no effort on their part.
>
> My reco is to provide an answer with an example or a link when possible.
And
> if you want to search for similar FS issues to your own, get on Nabble.
>
> I bought the eBook. Love it. Thank you.
>
> (FS on Dockstar with OpenWRT, awesome!!!!)
> --
> View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/Get-on-IRC-day-tp5615890p5618146.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From gservat at gmail.com  Sat Oct  9 16:39:38 2010
From: gservat at gmail.com (Gonzalo Servat)
Date: Sun, 10 Oct 2010 10:39:38 +1100
Subject: [Freeswitch-users] Get on IRC day!!
In-Reply-To: <AANLkTikY-GFgBCgUY67QgK-pTacz+0mo18QuQuLXpz7i@mail.gmail.com>
References: <AANLkTimdGX0ganMLyd3uTD3Ux1spa5xx8mShuV9DEww9@mail.gmail.com>
	<1286568418994-5616330.post@n2.nabble.com>
	<AANLkTi=GSO2YDqjs9O=1ib=roP8nsbS6C9x6n8kQMkgz@mail.gmail.com>
	<1286634138140-5618146.post@n2.nabble.com>
	<AANLkTikY-GFgBCgUY67QgK-pTacz+0mo18QuQuLXpz7i@mail.gmail.com>
Message-ID: <AANLkTim9d8G6k7tr4QR_NcD8LbYHyMUmrrDM22QTUEhA@mail.gmail.com>

Just like Michael said, if someone wants to set-up a forum and maintain, go
for it. An attempt has been made and it failed miserably, yet people
continue to ask for forums.

On Sun, Oct 10, 2010 at 7:54 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> The irony is with every new graduating class of FS users we repeat this
> discussion about how much the mailing list sucks VIA the mailing list!! =p
>
> We are always open to suggestions but as always only suggestions that come
> with action since we are already overloaded with what we are currently doing
> on a daily basis.
>
> Keep up the discussion but keep it positive......
> On Oct 9, 2010 9:27 AM, "humbr" <gerrit308 at gmail.com> wrote:
> >
> > The Nabble thing is working fine for me, in fact it is why I am still
> able to
> > get answers. I won't deal with the flood of emails from a pure mailing
> list.
> > Nabble is to me a decent bridge between those wanting a Forum and the
> > reality of there not being one. IRC, well it depends who happens to be on
> > when you need help. Sometimes you win, sometimes you don't :-) Don't
> give,
> > try again.
> >
> > As a long time computer user/coder/designer/hw fixer (wrote my first
> program
> > for an NCR315 in 1965 while in high school!) I am disturbed over the last
> 41
> > years to see the sheer laziness of people. With the www that has become
> much
> > worse. Everyone wants an instant solution to a complex problem,
> preferably
> > with no effort on their part.
> >
> > My reco is to provide an answer with an example or a link when possible.
> And
> > if you want to search for similar FS issues to your own, get on Nabble.
> >
> > I bought the eBook. Love it. Thank you.
> >
> > (FS on Dockstar with OpenWRT, awesome!!!!)
> > --
> > View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Get-on-IRC-day-tp5615890p5618146.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peter.schrock at gmail.com  Sat Oct  9 18:37:31 2010
From: peter.schrock at gmail.com (Peter Schrock)
Date: Sat, 9 Oct 2010 18:37:31 -0700
Subject: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57E0E6928E@cooper>
References: <AANLkTingeYfPT_b6dG3bKQOVjxYsmBTKhDuLw_3BKGac@mail.gmail.com>
	<A499E7B3-9727-4822-B2AB-8FDDCA3213FA@avgs.ca>
	<AANLkTinOUhDgzCB2rTL20_RzGq3=uS7zOKE064DOR0v6@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57E0E68E2E@cooper>
	<AANLkTinHyjVKiidD4d2X3d3hKDWDcRpokhVG8pU3dcUt@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57E0E6928E@cooper>
Message-ID: <5929949752088577909@unknownmsgid>

You know, I am asking myself the same question. "Is it really that hard?" Of
course, I wouldn't be asking unless I was having issues now, would I. As
annoyed as you sound in this email, so am I that, as you have indirectly
stated, this basic function should just work. However it is not and that is
why I am asking for help.

Thanks for the answered question on how to shutdown in the background.

PeterS

On Oct 8, 2010, at 12:22 AM, Peter Olsson <peter.olsson at visionutveckling.se>
wrote:

 I?m pretty sure that fs_cli is used by everyone who uses FS, since we want
the FreeSWITCH process itself to run in background.



I?m not really getting the problem here. Install FS with default config,
start FS in background, then run fs_cli to connect to the running instance ?
is it really that hard?



If running in background, you can stop with ?freeswitch ?stop? or ?fs_cli ?x
shutdown?



/Peter



*Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *F?r *Peter Schrock
*Skickat:* den 7 oktober 2010 21:41
*Till:* FreeSWITCH Users Help
*?mne:* Re: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4



John Covici,



To make sure I understand correctly, do you me the copy found in:

/usr/local/freeswitch/libs/esl

or somewhere else?

Also, is there a wiki that discusses how to make those parameters agreeable?



Peter Olsson,

Thanks for clarifying that. I guess my question now would be (cause I am
trying to use the book as much as possible to help me understand FS) is why
would the book be insistent on using fs_cli and how many people actually use
it. Is there a wiki discussing how to manage the mod_event_socket so that
fs_cli will function the way it should instead popping up errors like it has
been.



Also, in general, if one were to set up FS in background mode, how do you
stop it from being in background mode?



PeterS

On Wed, Oct 6, 2010 at 11:14 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

fs_cli is a tool to connect to a running instance of FS, on the same
computer, or another computer. It communicates with FS using the
mod_event_socket loaded within FS. The server socket in FS listens for
requests on port 8021 by default, this can be changed in
event_socket.conf.xml



/Peter



*Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *F?r *Peter Schrock
*Skickat:* den 7 oktober 2010 07:55
*Till:* FreeSWITCH Users Help
*?mne:* Re: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4



In answer to your first question, yes. I guess my next question would be, if
I don't need fs_cli, then what is it for? I guess I am assuming to access FS
when it is running in background. What would be the difference between
running it in the foreground vs background?

Also, I ran FS without -nc and it showed that mod_event_socket loaded
successfully. So why won't it with fs_cli?



PeterS

On Wed, Oct 6, 2010 at 5:29 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

Hi,



Did you start FreeSWITCH before? fs_cli is only used to connect to the
console of a running FreeSWITCH instance.



If you did, try starting it without the -nc switch, it should stay in the
foreground and tell you whats wrong, if you have a console you can also
check that mod_event_socket is properly loaded (load mod_event_socket)



Mathieu Rene

Avant-Garde Solutions Inc

Office: + 1 (514) 664-1044 x100

Cell: +1 (514) 664-1044 x200

mrene at avgs.ca









On 2010-10-06, at 8:23 PM, Peter Schrock wrote:



 Okay, so I managed to get FS working the other day and I even managed to
test a call and test my voicemail. All seemed to be working smoothly until,
because of the rain here, my power went out and I had to reboot my computer.
I logged in through the terminal, set up FS in background went to fs_cli and
I get this error message:

[ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
Error]

I managed to figure out that in the file "fs_cli.c" on line 1206 you find
the code for displaying this error message. The problem is that I don't know
why this error message is occurring. Does anyone have any helpful hints as
to what I should look at to resolve this problem?

I even tried going to the git tree and make current, but that gave me
problems that forced me to turn off mod_spandsp and mod_skyopen in the
modules.conf, which I had running earlier. Any thoughts?



PeterS



PS

I am not sure if this is of any help, but in addition to the error line
above, it also posted this info:

Usage: fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x
command] [profile]



  -?,-h --help                    Usage Information

  -H, --host=hostname             Host to connect

  -P, --port=port                 Port to connect (1 - 65535)

  -u, --user=user at domain          user at domain

  -p, --password=password         Password

  -x, --execute=command           Execute Command and Exit

  -l, --loglevel=command          Log Level

  -q, --quiet                     Disable logging

  -r, --retry                     Retry connection on failure

  -R, --reconnect                 Reconnect if disconnected

  -d, --debug=level               Debug Level (0 - 7)



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




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




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



!DSPAM:4cae242c32931240012030!

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From peter.schrock at gmail.com  Sat Oct  9 18:39:54 2010
From: peter.schrock at gmail.com (Peter Schrock)
Date: Sat, 9 Oct 2010 18:39:54 -0700
Subject: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4
In-Reply-To: <AANLkTi=YZsrcSWbrnUrZ6+P_fnXpQVAROCNc+9okF0X9@mail.gmail.com>
References: <AANLkTingeYfPT_b6dG3bKQOVjxYsmBTKhDuLw_3BKGac@mail.gmail.com>
	<AANLkTi=YZsrcSWbrnUrZ6+P_fnXpQVAROCNc+9okF0X9@mail.gmail.com>
Message-ID: <-4967171847893240536@unknownmsgid>

Thanks Steve, I will check on this and see if this works.

PeterS

On Oct 8, 2010, at 5:10 AM, Steven Ayre <steveayre at gmail.com> wrote:

> 1. Check FS is running
> 2. Check the port you're connecting to matches the one in event_socket.conf.xml
> 3. Check that event_socket.conf.xml binding to the same IP you're
> connecting to (e.g. 0.0.0.0 if you're connecting from a remote
> machine, since the default 127.0.0.1 won't work then)
> 4. Use netstat on the FS server to verify FreeSWITCH is actually
> listening on the port you're trying to connect to (I've had a syntax
> error in the config file make the module fail to load in the past
> which left FS running but with no ESL socket). (If the module fails to
> load there'll also be an error in the log file).
> 5. Check a firewall isn't blocking access to the port
>
> -Steve
>
>
>
> On 7 October 2010 01:23, Peter Schrock <peter.schrock at gmail.com> wrote:
>> Okay, so I managed to get FS working the other day and I even managed to
>> test a call and test my voicemail. All seemed to be working smoothly until,
>> because of the rain here, my power went out and I had to reboot my computer.
>> I logged in through the terminal, set up FS in background went to fs_cli and
>> I get this error message:
>> [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
>> Error]
>> I managed to figure out that in the file "fs_cli.c" on line 1206 you find
>> the code for displaying this error message. The problem is that I don't know
>> why this error message is occurring. Does anyone have any helpful hints as
>> to what I should look at to resolve this problem?
>> I even tried going to the git tree and make current, but that gave me
>> problems that forced me to turn off mod_spandsp and mod_skyopen in the
>> modules.conf, which I had running earlier. Any thoughts?
>> PeterS
>> PS
>> I am not sure if this is of any help, but in addition to the error line
>> above, it also posted this info:
>> Usage: fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x
>> command] [profile]
>>   -?,-h --help                    Usage Information
>>   -H, --host=hostname             Host to connect
>>   -P, --port=port                 Port to connect (1 - 65535)
>>   -u, --user=user at domain          user at domain
>>   -p, --password=password         Password
>>   -x, --execute=command           Execute Command and Exit
>>   -l, --loglevel=command          Log Level
>>   -q, --quiet                     Disable logging
>>   -r, --retry                     Retry connection on failure
>>   -R, --reconnect                 Reconnect if disconnected
>>   -d, --debug=level               Debug Level (0 - 7)
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From slim at thegreek.com  Sat Oct  9 21:24:01 2010
From: slim at thegreek.com (Jeroen C. van Gelderen)
Date: Sat, 9 Oct 2010 23:24:01 -0500
Subject: [Freeswitch-users] tone_detect vs. ignore_early_media (per leg?)
Message-ID: <429823D0454A4EB4AE179DF13803F9B9@mbnet.local>

Hi Guys,

I have a legacy Panasonic PBX which does not support CPC/Disconnect
Supervision. Calls from this PBX are sent to Freeswitch by way of
DAHDI/FreeTDM FXO ports (on a Xorcom Astribank). I use tone_detect to deal
with far-end hang-ups. This works fine:

<!-- Detect busy tone from Panasonic -->
<action application="tone_detect" data="busy 350,440 r 0 hangup 4"/>

Some calls from the Panasonic PBX are put in a FIFO and from there they are
sent to a SIP-to-GSM bridge (GoIP) to be terminated on one of 3 GSM phones
in a round-robin fashion.

The GSM bridge requires me to use "ignore_early_media=true" otherwise the
caller will receive messages like "the number you are calling does not
answer". When I set "ignore_early_media=true" the FIFO correctly keeps
hunting for a GSM phone that is actually answered and will ignore phones
that are busy, no-answer or turned off. This too works fine.

The problem occurs when the two are combined as follows:

Panasonic PBX --FXO--> Freeswitch --SIP--> GSM-bridge

If I enable ignore_early_media then tone_detect doesn't work UNTIL one of
the GSMs is answered. This is a problem when none of the GSMs are answered
and the Panasonic PBX hangs up. The FXO port will stay "up" and the FIFO
will keep hunting until it times out.
 
If I don't use "ignore_early_media=true" then tone_detect works fine but
then every telco message gets mistaken for an answered call and the hunting
stops early.

I tried changing this example line from my fifo.conf:

<member timeout="30" simo="1" lag="5">
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7701 at 192.1
68.3.11:5060
</member>

to 

<member timeout="30" simo="1" lag="5">
{member_wait=nowait}[ignore_early_media=true]sofia/internal/1??????7701 at 192.
168.3.11:5060
</member>

in a vain attempt to ignore early media on the SIP leg only. This doesn't
seem to do anything however.

Can anyone clue me in on what I'm missing? I've snipped the relevant
configuration bits below. I have the feeling I'm missing something obvious.

Cheers,
-Slim

----8<----8<----8<----8<----8<----

<configuration name="fifo.conf" description="FIFO Configuration">
  <settings>
    <param name="delete-all-outbound-member-on-startup" value="false"/>
  </settings>
  <fifos>
    <fifo name="cool_fifo@$${domain}" importance="0">

      <member timeout="30" simo="1" lag="5">
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7701 at 192.1
68.3.11:5060</member>

      <member timeout="30" simo="1" lag="5">
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7693 at 192.1
68.3.11:5060</member>

      <member timeout="30" simo="1" lag="5">
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7698 at 192.1
68.3.11:5060</member>

    </fifo>
  </fifos>
</configuration>

----8<----8<----8<----8<----8<----

<include>
  <context name="pbx_test">

    <extension name="test1">
      <condition field="${freetdm_span_name}" expression="^span_fxo_test$">
        <action application="answer"/>

        <!-- Detect busy tone from Panasonic -->
        <action application="tone_detect" data="busy 350,440 r 0 hangup 4"/>

        <action application="playback"
data="ivr/ivr-hold_connect_call.wav"/>

        <action application="fifo" data="cool_fifo@$${domain} in"/>

      </condition>
    </extension>
  </context>
</include>

----8<----8<----8<----8<----8<----

Cheers,
-Slim
--
Jeroen C. "Slim" van Gelderen




From babak.freeswitch at gmail.com  Sat Oct  9 22:18:49 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sun, 10 Oct 2010 08:48:49 +0330
Subject: [Freeswitch-users] fs_cli
In-Reply-To: <AANLkTi=HeFmsJ3BVTdBkc9YHAK7Apgcivj-cX0degiDm@mail.gmail.com>
References: <AANLkTi=L4VArPuBSDV5hgGtJYr3Kbkoh1XjGTtPwM0BE@mail.gmail.com>
	<AANLkTi=HeFmsJ3BVTdBkc9YHAK7Apgcivj-cX0degiDm@mail.gmail.com>
Message-ID: <AANLkTikRX3QcFfHhAA8+FHSddok0D_bDWDiqKQDCZwky@mail.gmail.com>

I'm writing output of my apis to console. on freeswitch cli it works and
shows the output but on fs_cli it is not
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From mnhassan at usa.net  Sat Oct  9 23:10:05 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Sun, 10 Oct 2010 12:10:05 +0600
Subject: [Freeswitch-users] fs_cli
In-Reply-To: <AANLkTikRX3QcFfHhAA8+FHSddok0D_bDWDiqKQDCZwky@mail.gmail.com>
References: <AANLkTi=L4VArPuBSDV5hgGtJYr3Kbkoh1XjGTtPwM0BE@mail.gmail.com>
	<AANLkTi=HeFmsJ3BVTdBkc9YHAK7Apgcivj-cX0degiDm@mail.gmail.com>
	<AANLkTikRX3QcFfHhAA8+FHSddok0D_bDWDiqKQDCZwky@mail.gmail.com>
Message-ID: <AANLkTinp4GdOyPZUcF1FoHB=dB2YbxEfi1JLrvvM2L23@mail.gmail.com>

If a command stop from within the console session, then it would run
from the command line as well, and should yield the same output.

If it is not, then you need to run it with full debug, put the output
into pastebin and send the reference here.

Regards
HASSAN


On 2010-10-10, babak yakhchali <babak.freeswitch at gmail.com> wrote:
> I'm writing output of my apis to console. on freeswitch cli it works and
> shows the output but on fs_cli it is not
>

-- 
Sent from my mobile device


From oseslija at gmail.com  Sat Oct  9 23:39:14 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Sun, 10 Oct 2010 08:39:14 +0200
Subject: [Freeswitch-users] tone_detect vs. ignore_early_media (per leg?)
In-Reply-To: <429823D0454A4EB4AE179DF13803F9B9@mbnet.local>
References: <429823D0454A4EB4AE179DF13803F9B9@mbnet.local>
Message-ID: <AANLkTi=jBe001uH8uRRCX9hpEeJuH_SwJr8eU86LT==w@mail.gmail.com>

I don

On Oct 10, 2010 6:30 AM, "Jeroen C. van Gelderen" <slim at thegreek.com> wrote:

Hi Guys,

I have a legacy Panasonic PBX which does not support CPC/Disconnect
Supervision. Calls from this PBX are sent to Freeswitch by way of
DAHDI/FreeTDM FXO ports (on a Xorcom Astribank). I use tone_detect to deal
with far-end hang-ups. This works fine:

<!-- Detect busy tone from Panasonic -->
<action application="tone_detect" data="busy 350,440 r 0 hangup 4"/>

Some calls from the Panasonic PBX are put in a FIFO and from there they are
sent to a SIP-to-GSM bridge (GoIP) to be terminated on one of 3 GSM phones
in a round-robin fashion.

The GSM bridge requires me to use "ignore_early_media=true" otherwise the
caller will receive messages like "the number you are calling does not
answer". When I set "ignore_early_media=true" the FIFO correctly keeps
hunting for a GSM phone that is actually answered and will ignore phones
that are busy, no-answer or turned off. This too works fine.

The problem occurs when the two are combined as follows:

Panasonic PBX --FXO--> Freeswitch --SIP--> GSM-bridge

If I enable ignore_early_media then tone_detect doesn't work UNTIL one of
the GSMs is answered. This is a problem when none of the GSMs are answered
and the Panasonic PBX hangs up. The FXO port will stay "up" and the FIFO
will keep hunting until it times out.

If I don't use "ignore_early_media=true" then tone_detect works fine but
then every telco message gets mistaken for an answered call and the hunting
stops early.

I tried changing this example line from my fifo.conf:

<member timeout="30" simo="1" lag="5">
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7701 at 192.1
68.3.11:5060
</member>

to

<member timeout="30" simo="1" lag="5">
{member_wait=nowait}[ignore_early_media=true]sofia/internal/1??????7701 at 192.
168.3.11:5060
</member>

in a vain attempt to ignore early media on the SIP leg only. This doesn't
seem to do anything however.

Can anyone clue me in on what I'm missing? I've snipped the relevant
configuration bits below. I have the feeling I'm missing something obvious.

Cheers,
-Slim

----8<----8<----8<----8<----8<----

<configuration name="fifo.conf" description="FIFO Configuration">
 <settings>
   <param name="delete-all-outbound-member-on-startup" value="false"/>
 </settings>
 <fifos>
   <fifo name="cool_fifo@$${domain}" importance="0">

     <member timeout="30" simo="1" lag="5">
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7701 at 192.1
68.3.11:5060</member>

     <member timeout="30" simo="1" lag="5">
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7693 at 192.1
68.3.11:5060</member>

     <member timeout="30" simo="1" lag="5">
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7698 at 192.1
68.3.11:5060</member>

   </fifo>
 </fifos>
</configuration>

----8<----8<----8<----8<----8<----

<include>
 <context name="pbx_test">

   <extension name="test1">
     <condition field="${freetdm_span_name}" expression="^span_fxo_test$">
       <action application="answer"/>

       <!-- Detect busy tone from Panasonic -->
       <action application="tone_detect" data="busy 350,440 r 0 hangup 4"/>

       <action application="playback"
data="ivr/ivr-hold_connect_call.wav"/>

       <action application="fifo" data="cool_fifo@$${domain} in"/>

     </condition>
   </extension>
 </context>
</include>

----8<----8<----8<----8<----8<----

Cheers,
-Slim
--
Jeroen C. "Slim" van Gelderen



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From jeff at jefflenk.com  Sun Oct 10 07:33:19 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sun, 10 Oct 2010 07:33:19 -0700 (PDT)
Subject: [Freeswitch-users] fs_cli
In-Reply-To: <AANLkTinp4GdOyPZUcF1FoHB=dB2YbxEfi1JLrvvM2L23@mail.gmail.com>
References: <AANLkTi=L4VArPuBSDV5hgGtJYr3Kbkoh1XjGTtPwM0BE@mail.gmail.com>
	<AANLkTi=HeFmsJ3BVTdBkc9YHAK7Apgcivj-cX0degiDm@mail.gmail.com>
	<AANLkTikRX3QcFfHhAA8+FHSddok0D_bDWDiqKQDCZwky@mail.gmail.com>
	<AANLkTinp4GdOyPZUcF1FoHB=dB2YbxEfi1JLrvvM2L23@mail.gmail.com>
Message-ID: <1286721199996-5620475.post@n2.nabble.com>


Here are the results from fs_cli for the sample Api call from demo.csx in the
mod_managed project folder. This was tested on the same machine as the fs
process. Compare your results to this and let us know.

fs_cli.exe -x "managed ApiDemo test"
ApiDemo executed with args 'test' and event type API.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/fs-cli-tp5618264p5620475.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mario_fs at mgtech.com  Sun Oct 10 09:37:16 2010
From: mario_fs at mgtech.com (Mario)
Date: Sun, 10 Oct 2010 09:37:16 -0700
Subject: [Freeswitch-users] Where does "Expires:" in a sip trace come from?
Message-ID: <4CB1EBBC.2000503@mgtech.com>

This newbie has 60+ hours into this problem and I really needs some
expert explanation for 2 items in a "sofia profile external siptrace on":

1. The section that starts with "REGISTER", the line that reads
"Expires:". Does the value always reflect what FS has set or is it
reflecting the ITSP? I thought it was the expiry-seconds from the
gateway definition.

2. Same question for the "SIP/2.0 200 Ok" section, line "Contact:", at
the end it reads "expires:".

Thanks.
Mario


From chat2jesse at gmail.com  Sun Oct 10 10:55:30 2010
From: chat2jesse at gmail.com (jesse)
Date: Sun, 10 Oct 2010 10:55:30 -0700
Subject: [Freeswitch-users] Where does "Expires:" in a sip trace come
	from?
In-Reply-To: <4CB1EBBC.2000503@mgtech.com>
References: <4CB1EBBC.2000503@mgtech.com>
Message-ID: <AANLkTi=LKuh2NM22T10wvgkMO3Y3c51oT3dzpznA93N_@mail.gmail.com>

Your questions are related with sip domain knowledge. Not FS specific. Read
RFC3261.

Jesse
On Oct 10, 2010 9:42 AM, "Mario" <mario_fs at mgtech.com> wrote:
> This newbie has 60+ hours into this problem and I really needs some
> expert explanation for 2 items in a "sofia profile external siptrace on":
>
> 1. The section that starts with "REGISTER", the line that reads
> "Expires:". Does the value always reflect what FS has set or is it
> reflecting the ITSP? I thought it was the expiry-seconds from the
> gateway definition.
>
> 2. Same question for the "SIP/2.0 200 Ok" section, line "Contact:", at
> the end it reads "expires:".
>
> Thanks.
> Mario
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From daniel.neubert at solomo.de  Sun Oct 10 11:53:20 2010
From: daniel.neubert at solomo.de (Daniel Neubert)
Date: Sun, 10 Oct 2010 20:53:20 +0200
Subject: [Freeswitch-users] Question about mod_nibblebill.
In-Reply-To: <AANLkTimgF20e6Qk4QrY3rrA-6_uh1tNC=SVNc7-eBs1A@mail.gmail.com>
References: <AANLkTimgF20e6Qk4QrY3rrA-6_uh1tNC=SVNc7-eBs1A@mail.gmail.com>
Message-ID: <4CB20BA0.6020506@solomo.de>

 Unfortunately I think that this is a scenario that is not covered by
mod_nibblebill. The module is not capable of charging customers this way.

We've solved this by using the nibble_rate as you do to realize a
prepaid billing setup. After the final xml CDR is posted to our billing
system via curl, we calculate the final price that will be billed to the
customer and refund the amount that has been charged by mod_nibblebill.
This works pretty good - but there is a catch: Customers can
theoretically run below an amount of 0,00 - so we had to configure a
nobal_amt to prevent this.

Viele Gr??e / Regards
Daniel Neubert


Am 09.10.2010 03:12, schrieb ???:
>
> Hello, I have a question about the mod_nibblebill module of freeswitch.
>
>
> I can use mod_nibblebill to billing. But I found that his billing was
> not what I want.
>
>
> I want to achieve in this way: every 60 seconds for an interval of
> less than 60 seconds, 60 seconds to install basis, charging $ 0.1 for
> each billing interval. For example: the user dials 45 seconds,
> charging $ 0.1. If user dials 72 seconds, charging $ 0.2.
>
>
> My current configuration in dialplain\default.xml is as follows:
>
> <action application="bridge"
> data="{enable_heartbeat_events=60,nibble_rate=0.1,nibble_account=1000}user/${dialed_extension}@${domain_name}"/>
>
>
> May I ask, mod_nibblebill module can not achieve my desired function,
> if realized, how can I configure?
>
> Thank you.
>
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From mario_fs at mgtech.com  Sun Oct 10 13:57:32 2010
From: mario_fs at mgtech.com (Mario)
Date: Sun, 10 Oct 2010 13:57:32 -0700
Subject: [Freeswitch-users] Where does "Expires:" in a sip trace come
 from?
In-Reply-To: <AANLkTi=LKuh2NM22T10wvgkMO3Y3c51oT3dzpznA93N_@mail.gmail.com>
References: <4CB1EBBC.2000503@mgtech.com>
	<AANLkTi=LKuh2NM22T10wvgkMO3Y3c51oT3dzpznA93N_@mail.gmail.com>
Message-ID: <4CB228BC.9090305@mgtech.com>

Thank you! I read it but not being a phone/voip expert like yourself it
did not answer my questions, at least that I understood. At least now I
realize this is the wrong place to be and why there was no response to
this problem for a week or so. I'm all done with FS config and very
happy with it but this bugger won't let me use FS for more than an 60-90
minutes. I wish I knew this earlier.. Will look for other online
(sofia?) resources. Thanks again. Mario

On 10/10/10 10:55, jesse wrote:
> Your questions are related with sip domain knowledge. Not FS specific.
> Read RFC3261.
> 
> Jesse
> 
> On Oct 10, 2010 9:42 AM, "Mario" <mario_fs at mgtech.com
> <mailto:mario_fs at mgtech.com>> wrote:
>> This newbie has 60+ hours into this problem and I really needs some
>> expert explanation for 2 items in a "sofia profile external siptrace on":
>>
>> 1. The section that starts with "REGISTER", the line that reads
>> "Expires:". Does the value always reflect what FS has set or is it
>> reflecting the ITSP? I thought it was the expiry-seconds from the
>> gateway definition.
>>
>> 2. Same question for the "SIP/2.0 200 Ok" section, line "Contact:", at
>> the end it reads "expires:".
>>
>> Thanks.
>> Mario
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Sun Oct 10 17:00:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 10 Oct 2010 17:00:50 -0700
Subject: [Freeswitch-users] tone_detect vs. ignore_early_media (per leg?)
In-Reply-To: <429823D0454A4EB4AE179DF13803F9B9@mbnet.local>
References: <429823D0454A4EB4AE179DF13803F9B9@mbnet.local>
Message-ID: <AANLkTim1B_oxbCAXNPTHNoZxh6vUfbUiWsyHzxG=xwif@mail.gmail.com>

Check out monitor_early_media_fail:
http://wiki.freeswitch.org/wiki/Channel_Variables#monitor_early_media_fail

It is a hybrid of ignoring early media and listening to early media for
various conditions. In your case you'll need to play around with it. In your
case you need to figure out which early media scenarios count as a "fail"
and will cause processing to move on as if the call really did fail.

Roll up your sleeves, you have some work to do. :)

-MC

On Sat, Oct 9, 2010 at 9:24 PM, Jeroen C. van Gelderen <slim at thegreek.com>wrote:

> Hi Guys,
>
> I have a legacy Panasonic PBX which does not support CPC/Disconnect
> Supervision. Calls from this PBX are sent to Freeswitch by way of
> DAHDI/FreeTDM FXO ports (on a Xorcom Astribank). I use tone_detect to deal
> with far-end hang-ups. This works fine:
>
> <!-- Detect busy tone from Panasonic -->
> <action application="tone_detect" data="busy 350,440 r 0 hangup 4"/>
>
> Some calls from the Panasonic PBX are put in a FIFO and from there they are
> sent to a SIP-to-GSM bridge (GoIP) to be terminated on one of 3 GSM phones
> in a round-robin fashion.
>
> The GSM bridge requires me to use "ignore_early_media=true" otherwise the
> caller will receive messages like "the number you are calling does not
> answer". When I set "ignore_early_media=true" the FIFO correctly keeps
> hunting for a GSM phone that is actually answered and will ignore phones
> that are busy, no-answer or turned off. This too works fine.
>
> The problem occurs when the two are combined as follows:
>
> Panasonic PBX --FXO--> Freeswitch --SIP--> GSM-bridge
>
> If I enable ignore_early_media then tone_detect doesn't work UNTIL one of
> the GSMs is answered. This is a problem when none of the GSMs are answered
> and the Panasonic PBX hangs up. The FXO port will stay "up" and the FIFO
> will keep hunting until it times out.
>
> If I don't use "ignore_early_media=true" then tone_detect works fine but
> then every telco message gets mistaken for an answered call and the hunting
> stops early.
>
> I tried changing this example line from my fifo.conf:
>
> <member timeout="30" simo="1" lag="5">
>
> {member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7701 at 192.1
> 68.3.11:5060
> </member>
>
> to
>
> <member timeout="30" simo="1" lag="5">
>
> {member_wait=nowait}[ignore_early_media=true]sofia/internal/1??????7701 at 192.
> 168.3.11:5060
> </member>
>
> in a vain attempt to ignore early media on the SIP leg only. This doesn't
> seem to do anything however.
>
> Can anyone clue me in on what I'm missing? I've snipped the relevant
> configuration bits below. I have the feeling I'm missing something obvious.
>
> Cheers,
> -Slim
>
> ----8<----8<----8<----8<----8<----
>
> <configuration name="fifo.conf" description="FIFO Configuration">
>  <settings>
>    <param name="delete-all-outbound-member-on-startup" value="false"/>
>  </settings>
>  <fifos>
>    <fifo name="cool_fifo@$${domain}" importance="0">
>
>      <member timeout="30" simo="1" lag="5">
>
> {member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7701 at 192.1
> 68.3.11:5060</member>
>
>      <member timeout="30" simo="1" lag="5">
>
> {member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7693 at 192.1
> 68.3.11:5060</member>
>
>      <member timeout="30" simo="1" lag="5">
>
> {member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7698 at 192.1
> 68.3.11:5060</member>
>
>    </fifo>
>  </fifos>
> </configuration>
>
> ----8<----8<----8<----8<----8<----
>
> <include>
>  <context name="pbx_test">
>
>    <extension name="test1">
>      <condition field="${freetdm_span_name}" expression="^span_fxo_test$">
>        <action application="answer"/>
>
>        <!-- Detect busy tone from Panasonic -->
>        <action application="tone_detect" data="busy 350,440 r 0 hangup 4"/>
>
>        <action application="playback"
> data="ivr/ivr-hold_connect_call.wav"/>
>
>        <action application="fifo" data="cool_fifo@$${domain} in"/>
>
>      </condition>
>    </extension>
>  </context>
> </include>
>
> ----8<----8<----8<----8<----8<----
>
> Cheers,
> -Slim
> --
> Jeroen C. "Slim" van Gelderen
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From slim at thegreek.com  Mon Oct 11 01:12:34 2010
From: slim at thegreek.com (Jeroen C. van Gelderen)
Date: Mon, 11 Oct 2010 03:12:34 -0500
Subject: [Freeswitch-users] tone_detect vs. ignore_early_media (per leg?)
In-Reply-To: <AANLkTim1B_oxbCAXNPTHNoZxh6vUfbUiWsyHzxG=xwif@mail.gmail.com>
References: <429823D0454A4EB4AE179DF13803F9B9@mbnet.local>
	<AANLkTim1B_oxbCAXNPTHNoZxh6vUfbUiWsyHzxG=xwif@mail.gmail.com>
Message-ID: <DAFF983A4D6A43A186D8662FE5876375@mbnet.local>

Thank you for the quick response.

 

It looks like monitor_early_media_fail should do what I need (thanks for the
suggestion!) but I can't seem to make it work. See below for my uneducated
best guess why.

 

In my case the failure conditions from the GSM side are handled out-of-band
by SIP. That leaves only one failure condition I need to listen for in early
media (the Panasonic far-end hang up on FXO) which can successfully be
detected with:

 

  <action application="tone_detect" data="busy-pan 350,440 r 0 hangup>

 

This results in the following log entries:

 

[DEBUG] switch_ivr_async.c:2072 Adding tone spec 350,440 index 0 hits 1

[DEBUG] switch_core_media_bug.c:360 Attaching BUG to FreeTDM/4:1/

 

Translating tone_detect to monitor_early_media_fail I tried the following:

 

  <action application="set" data="result=${fifo_member(add fifo_helpdesk
{fifo_member_wait=nowait,ignore_early_media=true,monitor_early_media_fail=bu
sy-pan:1:350+440}sofia/internal/${caller_id_number}@192.168.3.11:5060)}"/>

 

This results in the following relevant log entries:

 

[DEBUG] switch_channel.c:2444 (sofia/internal/1876??????@192.168.3.11:5060)
Callstate Change RINGING -> EARLY

[DEBUG] switch_ivr_async.c:2072 Adding tone spec 350,440 index 0 hits 1

[DEBUG] switch_core_media_bug.c:360 Attaching BUG to
sofia/internal/1??????7693 at 192.168.3.11:5060

 

So Freeswitch is listening for the right tones (tone_spec present and
identical to the one used in tone_detect approach) but it isn't detecting
them. The obvious difference is in the BUG attachment.

 

Is it possible that BUG isn't listening to the right (A) leg in the case of
monitor_early_media_fail? Or is this too easy? :-)

Cheers,
-Slim
--
Jeroen C. "Slim" van Gelderen



  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Sunday, October 10, 2010 19:01
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] tone_detect vs. ignore_early_media (per
leg?)

 

Check out monitor_early_media_fail:
http://wiki.freeswitch.org/wiki/Channel_Variables#monitor_early_media_fail

It is a hybrid of ignoring early media and listening to early media for
various conditions. In your case you'll need to play around with it. In your
case you need to figure out which early media scenarios count as a "fail"
and will cause processing to move on as if the call really did fail. 

Roll up your sleeves, you have some work to do. :)

-MC

On Sat, Oct 9, 2010 at 9:24 PM, Jeroen C. van Gelderen <slim at thegreek.com>
wrote:

Hi Guys,

I have a legacy Panasonic PBX which does not support CPC/Disconnect
Supervision. Calls from this PBX are sent to Freeswitch by way of
DAHDI/FreeTDM FXO ports (on a Xorcom Astribank). I use tone_detect to deal
with far-end hang-ups. This works fine:

<!-- Detect busy tone from Panasonic -->
<action application="tone_detect" data="busy 350,440 r 0 hangup 4"/>

Some calls from the Panasonic PBX are put in a FIFO and from there they are
sent to a SIP-to-GSM bridge (GoIP) to be terminated on one of 3 GSM phones
in a round-robin fashion.

The GSM bridge requires me to use "ignore_early_media=true" otherwise the
caller will receive messages like "the number you are calling does not
answer". When I set "ignore_early_media=true" the FIFO correctly keeps
hunting for a GSM phone that is actually answered and will ignore phones
that are busy, no-answer or turned off. This too works fine.

The problem occurs when the two are combined as follows:

Panasonic PBX --FXO--> Freeswitch --SIP--> GSM-bridge

If I enable ignore_early_media then tone_detect doesn't work UNTIL one of
the GSMs is answered. This is a problem when none of the GSMs are answered
and the Panasonic PBX hangs up. The FXO port will stay "up" and the FIFO
will keep hunting until it times out.

If I don't use "ignore_early_media=true" then tone_detect works fine but
then every telco message gets mistaken for an answered call and the hunting
stops early.

I tried changing this example line from my fifo.conf:

<member timeout="30" simo="1" lag="5">
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7701 at 192.1
68.3.11:5060
</member>

to

<member timeout="30" simo="1" lag="5">
{member_wait=nowait}[ignore_early_media=true]sofia/internal/1??????7701 at 192.
168.3.11:5060
</member>

in a vain attempt to ignore early media on the SIP leg only. This doesn't
seem to do anything however.

Can anyone clue me in on what I'm missing? I've snipped the relevant
configuration bits below. I have the feeling I'm missing something obvious.

Cheers,
-Slim

----8<----8<----8<----8<----8<----

<configuration name="fifo.conf" description="FIFO Configuration">
 <settings>
   <param name="delete-all-outbound-member-on-startup" value="false"/>
 </settings>
 <fifos>
   <fifo name="cool_fifo@$${domain}" importance="0">

     <member timeout="30" simo="1" lag="5">
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7701 at 192.1
68.3.11:5060</member>

     <member timeout="30" simo="1" lag="5">
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7693 at 192.1
68.3.11:5060</member>

     <member timeout="30" simo="1" lag="5">
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7698 at 192.1
68.3.11:5060</member>

   </fifo>
 </fifos>
</configuration>

----8<----8<----8<----8<----8<----

<include>
 <context name="pbx_test">

   <extension name="test1">
     <condition field="${freetdm_span_name}" expression="^span_fxo_test$">
       <action application="answer"/>

       <!-- Detect busy tone from Panasonic -->
       <action application="tone_detect" data="busy 350,440 r 0 hangup 4"/>

       <action application="playback"
data="ivr/ivr-hold_connect_call.wav"/>

       <action application="fifo" data="cool_fifo@$${domain} in"/>

     </condition>
   </extension>
 </context>
</include>

----8<----8<----8<----8<----8<----

Cheers,
-Slim
--
Jeroen C. "Slim" van Gelderen



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

 

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From slim at thegreek.com  Mon Oct 11 03:30:52 2010
From: slim at thegreek.com (Jeroen C. van Gelderen)
Date: Mon, 11 Oct 2010 05:30:52 -0500
Subject: [Freeswitch-users] tone_detect vs. ignore_early_media AND
	mod_fifo
In-Reply-To: <DAFF983A4D6A43A186D8662FE5876375@mbnet.local>
References: <429823D0454A4EB4AE179DF13803F9B9@mbnet.local><AANLkTim1B_oxbCAXNPTHNoZxh6vUfbUiWsyHzxG=xwif@mail.gmail.com>
	<DAFF983A4D6A43A186D8662FE5876375@mbnet.local>
Message-ID: <515E6743DC69484A8E3514E807B24353@mbnet.local>

 

Hmm. the plot thickens.

 

When I bridge my FXO port to the SIP-GSM gateway directly (i.e. without
using mod_fifo) I don't seem to need monitor_early_media_fail. Using
tone_detect on the A leg works fine when "ignore_early_media=true" is used
on the B leg:

 

The following dialplan excerpt WORKS (i.e. FXO hang-ups are detected at all
stages by tone_detect):

 

    <extension name="pan_125">

      <condition field="${freetdm_span_name}"
expression="^span_fxo_helpdesk$">

        <!-- Detect busy tone from Panasonic -->

        <action application="tone_detect" data="busy-pan 350,440 r 0 hangup
undef 2"/>

 

        <action application="answer"/>

 

        <action application="set" data="continue_on_fail=true"/>

        <action application="set" data="instant_ringback=true"/>

        <action application="set" data="ringback=%(1000,1500,440.0,480.0)"/>

 

        <action application="bridge"
data="{ignore_early_media=true}sofia/internal/1??????76??@192.168.3.11:5060"
/>

        <action application="bridge"
data="{ignore_early_media=true}sofia/internal/1??????77??@192.168.3.11:5060"
/>

      </condition>

    </extension>

 

The problem seems to occur only when mod_fifo is added to the mix:

 

    <extension name="test2">

      <condition field="${freetdm_span_name}" expression="^span_fxo_test$">

        <!-- Detect busy tone from Panasonic -->

        <action application="tone_detect" data="busy-pan 350,440 r 0 hangup
undef 2"/>

 

        <action application="answer"/>

 

        <action application="playback" data="thankyou.wav"/>

        <action application="set" data="fifo_music=$${hold_music}"/>

        <action application="fifo" data="fifo_helpdesk in"/>

      </condition>

    </extension>

 

In all cases: 

-          tone_detect works BEFORE the call is handed to mod_fifo (i.e.
during playback)

-          tone_detect works AFTER the call is established by mod_fifo and
audio is being exchanged between A and B leg.

 

IF the A leg (with tone_detect enabled) is handed to mod_fifo AND there are
no agents in the fifo THEN tone_detect does not work. Mod_fifo simply plays
MOH to the A leg perpetually.

 

IF the A leg (with tone_detect enabled) is handed to mod_fifo AND mod_fifo
places an outbound call with "ignore_early_media=true" THEN tone_detect does
not work on the A leg during the early media phase on leg B.

 

IF the A leg (with tone_detect enabled) is handed to mod_fifo AND mod_fifo
places an outbound call with "ignore_early_media=false" THEN tone_detect
does work. I guess this is because audio is being exchanged between A and B
legs.

 

Any idea what would cause the tone_detect to be "suspended" when mod_fifo is
in the mix?

Cheers,
-Slim
--
Jeroen C. "Slim" van Gelderen



  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeroen
C. van Gelderen
Sent: Monday, October 11, 2010 03:13
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] tone_detect vs. ignore_early_media (per
leg?)

 

Thank you for the quick response.

 

It looks like monitor_early_media_fail should do what I need (thanks for the
suggestion!) but I can't seem to make it work. See below for my uneducated
best guess why.

 

In my case the failure conditions from the GSM side are handled out-of-band
by SIP. That leaves only one failure condition I need to listen for in early
media (the Panasonic far-end hang up on FXO) which can successfully be
detected with:

 

  <action application="tone_detect" data="busy-pan 350,440 r 0 hangup>

 

This results in the following log entries:

 

[DEBUG] switch_ivr_async.c:2072 Adding tone spec 350,440 index 0 hits 1

[DEBUG] switch_core_media_bug.c:360 Attaching BUG to FreeTDM/4:1/

 

Translating tone_detect to monitor_early_media_fail I tried the following:

 

  <action application="set" data="result=${fifo_member(add fifo_helpdesk
{fifo_member_wait=nowait,ignore_early_media=true,monitor_early_media_fail=bu
sy-pan:1:350+440}sofia/internal/${caller_id_number}@192.168.3.11:5060)}"/>

 

This results in the following relevant log entries:

 

[DEBUG] switch_channel.c:2444 (sofia/internal/1876??????@192.168.3.11:5060)
Callstate Change RINGING -> EARLY

[DEBUG] switch_ivr_async.c:2072 Adding tone spec 350,440 index 0 hits 1

[DEBUG] switch_core_media_bug.c:360 Attaching BUG to
sofia/internal/1??????7693 at 192.168.3.11:5060

 

So Freeswitch is listening for the right tones (tone_spec present and
identical to the one used in tone_detect approach) but it isn't detecting
them. The obvious difference is in the BUG attachment.

 

Is it possible that BUG isn't listening to the right (A) leg in the case of
monitor_early_media_fail? Or is this too easy? :-)

Cheers,
-Slim
--
Jeroen C. "Slim" van Gelderen

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Sunday, October 10, 2010 19:01
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] tone_detect vs. ignore_early_media (per
leg?)

 

Check out monitor_early_media_fail:
http://wiki.freeswitch.org/wiki/Channel_Variables#monitor_early_media_fail

It is a hybrid of ignoring early media and listening to early media for
various conditions. In your case you'll need to play around with it. In your
case you need to figure out which early media scenarios count as a "fail"
and will cause processing to move on as if the call really did fail. 

Roll up your sleeves, you have some work to do. :)

-MC

On Sat, Oct 9, 2010 at 9:24 PM, Jeroen C. van Gelderen <slim at thegreek.com>
wrote:

Hi Guys,

I have a legacy Panasonic PBX which does not support CPC/Disconnect
Supervision. Calls from this PBX are sent to Freeswitch by way of
DAHDI/FreeTDM FXO ports (on a Xorcom Astribank). I use tone_detect to deal
with far-end hang-ups. This works fine:

<!-- Detect busy tone from Panasonic -->
<action application="tone_detect" data="busy 350,440 r 0 hangup 4"/>

Some calls from the Panasonic PBX are put in a FIFO and from there they are
sent to a SIP-to-GSM bridge (GoIP) to be terminated on one of 3 GSM phones
in a round-robin fashion.

The GSM bridge requires me to use "ignore_early_media=true" otherwise the
caller will receive messages like "the number you are calling does not
answer". When I set "ignore_early_media=true" the FIFO correctly keeps
hunting for a GSM phone that is actually answered and will ignore phones
that are busy, no-answer or turned off. This too works fine.

The problem occurs when the two are combined as follows:

Panasonic PBX --FXO--> Freeswitch --SIP--> GSM-bridge

If I enable ignore_early_media then tone_detect doesn't work UNTIL one of
the GSMs is answered. This is a problem when none of the GSMs are answered
and the Panasonic PBX hangs up. The FXO port will stay "up" and the FIFO
will keep hunting until it times out.

If I don't use "ignore_early_media=true" then tone_detect works fine but
then every telco message gets mistaken for an answered call and the hunting
stops early.

I tried changing this example line from my fifo.conf:

<member timeout="30" simo="1" lag="5">
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7701 at 192.1
68.3.11:5060
</member>

to

<member timeout="30" simo="1" lag="5">
{member_wait=nowait}[ignore_early_media=true]sofia/internal/1??????7701 at 192.
168.3.11:5060
</member>

in a vain attempt to ignore early media on the SIP leg only. This doesn't
seem to do anything however.

Can anyone clue me in on what I'm missing? I've snipped the relevant
configuration bits below. I have the feeling I'm missing something obvious.

Cheers,
-Slim

----8<----8<----8<----8<----8<----

<configuration name="fifo.conf" description="FIFO Configuration">
 <settings>
   <param name="delete-all-outbound-member-on-startup" value="false"/>
 </settings>
 <fifos>
   <fifo name="cool_fifo@$${domain}" importance="0">

     <member timeout="30" simo="1" lag="5">
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7701 at 192.1
68.3.11:5060</member>

     <member timeout="30" simo="1" lag="5">
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7693 at 192.1
68.3.11:5060</member>

     <member timeout="30" simo="1" lag="5">
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7698 at 192.1
68.3.11:5060</member>

   </fifo>
 </fifos>
</configuration>

----8<----8<----8<----8<----8<----

<include>
 <context name="pbx_test">

   <extension name="test1">
     <condition field="${freetdm_span_name}" expression="^span_fxo_test$">
       <action application="answer"/>

       <!-- Detect busy tone from Panasonic -->
       <action application="tone_detect" data="busy 350,440 r 0 hangup 4"/>

       <action application="playback"
data="ivr/ivr-hold_connect_call.wav"/>

       <action application="fifo" data="cool_fifo@$${domain} in"/>

     </condition>
   </extension>
 </context>
</include>

----8<----8<----8<----8<----8<----

Cheers,
-Slim
--
Jeroen C. "Slim" van Gelderen



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

 

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From slim at thegreek.com  Mon Oct 11 04:51:15 2010
From: slim at thegreek.com (Jeroen C. van Gelderen)
Date: Mon, 11 Oct 2010 06:51:15 -0500
Subject: [Freeswitch-users] tone_detect vs. ignore_early_media
	ANDmod_fifo
In-Reply-To: <515E6743DC69484A8E3514E807B24353@mbnet.local>
References: <429823D0454A4EB4AE179DF13803F9B9@mbnet.local><AANLkTim1B_oxbCAXNPTHNoZxh6vUfbUiWsyHzxG=xwif@mail.gmail.com><DAFF983A4D6A43A186D8662FE5876375@mbnet.local>
	<515E6743DC69484A8E3514E807B24353@mbnet.local>
Message-ID: <D33D5C7C9BAE42CA96354582D25F8B21@mbnet.local>

 

I have an inkling that the following commit made between 1.0.2 and 1.0.3
might have something to do with this:

 

   * mod_fifo: pause media bugs while not in a bridge (r:11466,11490)

 

http://lists.freeswitch.org/pipermail/freeswitch-svn/2009-January/009566.htm
l

http://lists.freeswitch.org/pipermail/freeswitch-svn/2009-January/009589.htm
l

 

Can anyone comment on the how and why? Is there any way to reconcile this
with use of tone_detect?

Cheers,
-Slim
--
Jeroen C. "Slim" van Gelderen
Olympic Sports Data Services
Email: slim at thegreek.com
Phone: +1 876 953 6182 x128

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeroen
C. van Gelderen
Sent: Monday, October 11, 2010 05:31
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] tone_detect vs. ignore_early_media
ANDmod_fifo

 

 

Hmm. the plot thickens.

 

When I bridge my FXO port to the SIP-GSM gateway directly (i.e. without
using mod_fifo) I don't seem to need monitor_early_media_fail. Using
tone_detect on the A leg works fine when "ignore_early_media=true" is used
on the B leg:

 

The following dialplan excerpt WORKS (i.e. FXO hang-ups are detected at all
stages by tone_detect):

 

    <extension name="pan_125">

      <condition field="${freetdm_span_name}"
expression="^span_fxo_helpdesk$">

        <!-- Detect busy tone from Panasonic -->

        <action application="tone_detect" data="busy-pan 350,440 r 0 hangup
undef 2"/>

 

        <action application="answer"/>

 

        <action application="set" data="continue_on_fail=true"/>

        <action application="set" data="instant_ringback=true"/>

        <action application="set" data="ringback=%(1000,1500,440.0,480.0)"/>

 

        <action application="bridge"
data="{ignore_early_media=true}sofia/internal/1??????76??@192.168.3.11:5060"
/>

        <action application="bridge"
data="{ignore_early_media=true}sofia/internal/1??????77??@192.168.3.11:5060"
/>

      </condition>

    </extension>

 

The problem seems to occur only when mod_fifo is added to the mix:

 

    <extension name="test2">

      <condition field="${freetdm_span_name}" expression="^span_fxo_test$">

        <!-- Detect busy tone from Panasonic -->

        <action application="tone_detect" data="busy-pan 350,440 r 0 hangup
undef 2"/>

 

        <action application="answer"/>

 

        <action application="playback" data="thankyou.wav"/>

        <action application="set" data="fifo_music=$${hold_music}"/>

        <action application="fifo" data="fifo_helpdesk in"/>

      </condition>

    </extension>

 

In all cases: 

-          tone_detect works BEFORE the call is handed to mod_fifo (i.e.
during playback)

-          tone_detect works AFTER the call is established by mod_fifo and
audio is being exchanged between A and B leg.

 

IF the A leg (with tone_detect enabled) is handed to mod_fifo AND there are
no agents in the fifo THEN tone_detect does not work. Mod_fifo simply plays
MOH to the A leg perpetually.

 

IF the A leg (with tone_detect enabled) is handed to mod_fifo AND mod_fifo
places an outbound call with "ignore_early_media=true" THEN tone_detect does
not work on the A leg during the early media phase on leg B.

 

IF the A leg (with tone_detect enabled) is handed to mod_fifo AND mod_fifo
places an outbound call with "ignore_early_media=false" THEN tone_detect
does work. I guess this is because audio is being exchanged between A and B
legs.

 

Any idea what would cause the tone_detect to be "suspended" when mod_fifo is
in the mix?

Cheers,
-Slim
--
Jeroen C. "Slim" van Gelderen

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeroen
C. van Gelderen
Sent: Monday, October 11, 2010 03:13
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] tone_detect vs. ignore_early_media (per
leg?)

 

Thank you for the quick response.

 

It looks like monitor_early_media_fail should do what I need (thanks for the
suggestion!) but I can't seem to make it work. See below for my uneducated
best guess why.

 

In my case the failure conditions from the GSM side are handled out-of-band
by SIP. That leaves only one failure condition I need to listen for in early
media (the Panasonic far-end hang up on FXO) which can successfully be
detected with:

 

  <action application="tone_detect" data="busy-pan 350,440 r 0 hangup>

 

This results in the following log entries:

 

[DEBUG] switch_ivr_async.c:2072 Adding tone spec 350,440 index 0 hits 1

[DEBUG] switch_core_media_bug.c:360 Attaching BUG to FreeTDM/4:1/

 

Translating tone_detect to monitor_early_media_fail I tried the following:

 

  <action application="set" data="result=${fifo_member(add fifo_helpdesk
{fifo_member_wait=nowait,ignore_early_media=true,monitor_early_media_fail=bu
sy-pan:1:350+440}sofia/internal/${caller_id_number}@192.168.3.11:5060)}"/>

 

This results in the following relevant log entries:

 

[DEBUG] switch_channel.c:2444 (sofia/internal/1876??????@192.168.3.11:5060)
Callstate Change RINGING -> EARLY

[DEBUG] switch_ivr_async.c:2072 Adding tone spec 350,440 index 0 hits 1

[DEBUG] switch_core_media_bug.c:360 Attaching BUG to
sofia/internal/1??????7693 at 192.168.3.11:5060

 

So Freeswitch is listening for the right tones (tone_spec present and
identical to the one used in tone_detect approach) but it isn't detecting
them. The obvious difference is in the BUG attachment.

 

Is it possible that BUG isn't listening to the right (A) leg in the case of
monitor_early_media_fail? Or is this too easy? :-)

Cheers,
-Slim
--
Jeroen C. "Slim" van Gelderen

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Sunday, October 10, 2010 19:01
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] tone_detect vs. ignore_early_media (per
leg?)

 

Check out monitor_early_media_fail:
http://wiki.freeswitch.org/wiki/Channel_Variables#monitor_early_media_fail

It is a hybrid of ignoring early media and listening to early media for
various conditions. In your case you'll need to play around with it. In your
case you need to figure out which early media scenarios count as a "fail"
and will cause processing to move on as if the call really did fail. 

Roll up your sleeves, you have some work to do. :)

-MC

On Sat, Oct 9, 2010 at 9:24 PM, Jeroen C. van Gelderen <slim at thegreek.com>
wrote:

Hi Guys,

I have a legacy Panasonic PBX which does not support CPC/Disconnect
Supervision. Calls from this PBX are sent to Freeswitch by way of
DAHDI/FreeTDM FXO ports (on a Xorcom Astribank). I use tone_detect to deal
with far-end hang-ups. This works fine:

<!-- Detect busy tone from Panasonic -->
<action application="tone_detect" data="busy 350,440 r 0 hangup 4"/>

Some calls from the Panasonic PBX are put in a FIFO and from there they are
sent to a SIP-to-GSM bridge (GoIP) to be terminated on one of 3 GSM phones
in a round-robin fashion.

The GSM bridge requires me to use "ignore_early_media=true" otherwise the
caller will receive messages like "the number you are calling does not
answer". When I set "ignore_early_media=true" the FIFO correctly keeps
hunting for a GSM phone that is actually answered and will ignore phones
that are busy, no-answer or turned off. This too works fine.

The problem occurs when the two are combined as follows:

Panasonic PBX --FXO--> Freeswitch --SIP--> GSM-bridge

If I enable ignore_early_media then tone_detect doesn't work UNTIL one of
the GSMs is answered. This is a problem when none of the GSMs are answered
and the Panasonic PBX hangs up. The FXO port will stay "up" and the FIFO
will keep hunting until it times out.

If I don't use "ignore_early_media=true" then tone_detect works fine but
then every telco message gets mistaken for an answered call and the hunting
stops early.

I tried changing this example line from my fifo.conf:

<member timeout="30" simo="1" lag="5">
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7701 at 192.1
68.3.11:5060
</member>

to

<member timeout="30" simo="1" lag="5">
{member_wait=nowait}[ignore_early_media=true]sofia/internal/1??????7701 at 192.
168.3.11:5060
</member>

in a vain attempt to ignore early media on the SIP leg only. This doesn't
seem to do anything however.

Can anyone clue me in on what I'm missing? I've snipped the relevant
configuration bits below. I have the feeling I'm missing something obvious.

Cheers,
-Slim

----8<----8<----8<----8<----8<----

<configuration name="fifo.conf" description="FIFO Configuration">
 <settings>
   <param name="delete-all-outbound-member-on-startup" value="false"/>
 </settings>
 <fifos>
   <fifo name="cool_fifo@$${domain}" importance="0">

     <member timeout="30" simo="1" lag="5">
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7701 at 192.1
68.3.11:5060</member>

     <member timeout="30" simo="1" lag="5">
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7693 at 192.1
68.3.11:5060</member>

     <member timeout="30" simo="1" lag="5">
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7698 at 192.1
68.3.11:5060</member>

   </fifo>
 </fifos>
</configuration>

----8<----8<----8<----8<----8<----

<include>
 <context name="pbx_test">

   <extension name="test1">
     <condition field="${freetdm_span_name}" expression="^span_fxo_test$">
       <action application="answer"/>

       <!-- Detect busy tone from Panasonic -->
       <action application="tone_detect" data="busy 350,440 r 0 hangup 4"/>

       <action application="playback"
data="ivr/ivr-hold_connect_call.wav"/>

       <action application="fifo" data="cool_fifo@$${domain} in"/>

     </condition>
   </extension>
 </context>
</include>

----8<----8<----8<----8<----8<----

Cheers,
-Slim
--
Jeroen C. "Slim" van Gelderen



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

 

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From riedinger at sns.eu  Mon Oct 11 05:05:19 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Mon, 11 Oct 2010 14:05:19 +0200
Subject: [Freeswitch-users] Problematic Behaviour of FS regarding ptime
 negotiation
In-Reply-To: <AANLkTim21cO0ua2e6-4cx0rX7MQ8E0eawM4OSOW2z6wH@mail.gmail.com>
References: <4CAEF500.1050202@sns.eu>	<E18C5FFA-F9BD-43F0-9697-1BCC7A7176C3@ipeva.fr>	<4CAF02ED.60508@sns.eu>	<D364FEBB-0B8F-4CD0-AA95-D7BE30E3C24A@ipeva.fr>	<AANLkTin9A+m6koQu5YnEVJs6bFBUzd1_vX_XhhFKf+Wz@mail.gmail.com>	<4CAF32DC.3040005@sns.eu>
	<AANLkTim21cO0ua2e6-4cx0rX7MQ8E0eawM4OSOW2z6wH@mail.gmail.com>
Message-ID: <4CB2FD7F.9060802@sns.eu>

  I could solve the problem by setting "rtp-autofix-timing=false", which 
disabled the (too) smart behaviour of FreeSwitch. I'm happy with this 
solution, but maybe it would helpful in general to enable the setting of 
this variable channel specific. Thus it would be possible to use 
standard ports even for interconnects to problematic routes/GWs.

BR
    Jan

Am 08.10.2010 17:14, schrieb Anthony Minessale:
> I already stated my position and supplied you with a solution.
> I am on to other issues now.
>
>
> On Fri, Oct 8, 2010 at 10:03 AM, Jan Riedinger<riedinger at sns.eu>  wrote:
>> Hi Anthony,
>>
>> you are writing: "Wanting to send 60 and not actually specifying it..."
>>
>> According my interpretation of the RFC it's just not possible for gateway to
>> specify the preferred ptime for SENDING. The ptime specifies the preferred
>> frame size for RECEIVING.
>>
>> In the RFC 3264 is written:
>>
>> If the ptime attribute is present for a stream, it indicates the
>> desired packetization interval that the offerer would like to
>> RECEIVE.
>>
>> Indications that the RFC is to be interpreted in this way can be found under
>>
>> http://www.cisco.com/en/US/docs/ios/12_3/sip/configuration/guide/chapter8.html#wp1064009
>>
>> If you study the examples of this web site for asymmetric SDP you will find,
>> that the ptime, which was requested by GW A is used from GW B for sending
>> packets, whereas GW A itself uses the frame size for sending, which was
>> requested by GW B!
>>
>> BR
>>      Jan
>>
>>
>>
>>
>>
>>
>> Am 08.10.2010 16:18, schrieb Anthony Minessale:
>>
>> Everything you described is how we behave.
>> We will not be changing it.
>>
>> Not specifying the ptime is a giant performance hit because we cannot
>> initilize the timers.
>> Wanting to send 60 and not actually specifying it could also probably
>> explained away in some deep interpretation of the RFC but it's not
>> typical and it's plain foolish.
>>
>> The only advice I can give you is to create a dedicated sip profile
>> for this call path and configure the codec negotiation to scrooge and
>> define G729 at 60i in your codec config.
>>
>> This will make FS use 60ms g729 regardless of what it sees in the sdp,
>> this is not optimal for anyone but your one case which is why I say to
>> put it in a specific profile.
>>
>>
>> On Fri, Oct 8, 2010 at 6:57 AM, David Ponzone<david.ponzone at ipeva.fr>
>> wrote:
>>
>> I am out of my league, I would need to dig into the RFCs so I prefer to wait
>> for comments from the people who wrote the code, because I am sure they have
>> an opinion on how the RFCs should be read or why they did what they did.
>> For the autofix-timing thing, I don't think there is a way to change that
>> per gateway or with a channel variable.
>> So yes, you have to use another IP or same IP with non-standard port.
>> But you just need 2: one profile with it, one without.
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>> Le 08/10/2010 ? 13:39, Jan Riedinger a ?crit :
>>
>>
>> Am 08.10.2010 13:06, schrieb David Ponzone:
>>
>> Jan,
>> to answer to 2 others questions you ask:
>> Why FS tries to enforce 20 ?
>> Well, the default is 20ms for most codecs, except perhaps G723, and no
>> explicit ptime means 20ms for most codecs.
>> So your carrier is sending no ptime, meaning they want 20.
>> FS agrees and send back 20 (and explictly, because smart people always do
>> things explictly and avoid relying on default values/behaviours).
>>
>> I think this isn't correct. If you work with a codec list in a Cisco and set
>> any byte / frame size values for the codecs of the codec list, the Cisco
>> doesn't specify any ptime in the initial INVITE message, even if for all
>> codecs of the codec list the same frame size is specified. Thus it's risky
>> at this point, if FS assums that the caller wants to use 20 ms
>>
>>
>> So, yes, the message displayed by FS is correct at some point:
>> FS asked for 20ms, and your carrier is sending 60ms.
>>
>> But the usage of 60 ms nevertheless, is ok according the RFC.
>>
>> Now, I see your point: perhaps the phrase is not very clear.
>> I think the issue is (and Anthony or Brian will correct me on this if
>> required) that FS tries to negotiate the same ptime on both directions,
>> because what the RFC says about asymmetrical ptimes is scary, AFAIK. I heard
>> people reporting major issues trying to do this.
>>
>> I configured for a long time a payload of 40 or 60 bytes on my Ciscos,
>> because of the disadvantageous TCP/IP header overhead, if you go with 20
>> bytes. I asked my business partners to do it in the same way. However, often
>> the didn't change their standard config and continued to use 20 bytes. I had
>> trouble by this asymmetry only once out of more than 200 configured
>> interconnects.
>>
>> Ok the RFC allows it, but as usual, it was probably badly implemented by
>> most vendors, and anyway, there is no real benefit.
>>
>> So FS tries to stay simple.
>> I think that's what FS means by "We were told": the other party asked us for
>> 20ms, and as we like to keep things simple, we also asked for 20ms, and they
>> send back 60ms, those p....bast.... :)
>>
>> As you see in the trace graph attached to my previous e-mail, the re-INVITE
>> of FFS results in an "internal server error" at the terminating GW. Of
>> course this shouldn't be the case and doesn't comply with the RFC, but this
>> problem is caused by the efforts of FS to fix a problem, which doesn' exist
>> - at least according the RFC.
>>
>> Basically, I think what you are asking is a new parameter that would
>> instruct FS to stop trying to re-packetize and accept asymmetrical ptimes.
>> About the message, you can get rid of it with rtp-autofix-timing=false, but
>> use it at your own risk.
>>
>> Is it possible to use rtp-autofix-timing just for a specific carrier? If I
>> specify it in the default profile, it is used for all carriers.
>> Maybe/Probably I'm wrong, but according my current knowledge I have to use
>> another non standard IP port, if I want to use another profile just for this
>> specific carrier.
>>
>> BR
>>      Jan
>>
>> David Ponzone   Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client  IP eva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>> Le 08/10/2010 ? 12:40, Jan Riedinger a ?crit :
>>
>> I'm terminating various destination by various carriers. After migrating one
>> customer to Freeswitch, we observed problems for the termination of a
>> specific route for a specific carrier. I tried to examine the problem in
>> detail and I think it's related to problems regarding the ptime negotiation.
>> I think Freeswitch doesn't breach any RFC, but I'm not sure, if the
>> behaviour is optimal.
>>
>> The SDP of the Caller INVITE-Message at time 1160,056 in the attached trace
>> doesn't include any ptime setting. Nevertheless Freeswitch includes a
>> ptime=20 media attribute in the forwarded INVITE message at time 1160,065.
>> The ringing SDP sent by the callee at time 1161,948 again doesn't include
>> any ptime setting. Nevertheless, Freeswitch includes in the Session Progress
>> SDP (at time 1164,240) a ptime=20 media atrribute. Why try Freeswitch to
>> force the usage of ptime=20 for the communication?
>>
>> The OK SDP at time 1164,240 again doesn't contain a ptime media attribute.
>> Nevertheless, the Freeswitch add a ptime=20 media attribute forwarded to the
>> caller at 1164,256.
>>
>> It seems that the callee is sending in the following with a frame size of 60
>> bytes - it never claimed to use ptime=20 and according the RFC 3264 it
>> SHOULD send with ptime=20 because of the received INVITE message
>> specification, but it DON'T HAVE to send with ptime=20.
>>
>> At next Freeswitch tries to fix "the issue". In the logfile I found:
>>
>> e686b430-5d2d-488b-8b58-0fca1965eea7 2010-10-07 15:20:25.673206 [WARNING]
>> mod_sofia.c:1033 We were told to use ptime 20 but what they meant to say was
>> 60
>> This issue has so far been identified to happen on the following broken
>> platforms/devices:
>> Linksys/Sipura aka Cisco
>> ShoreTel
>> Sonus/L3
>> We will try to fix it but some of the devices on this list are so broken,
>> who knows what will happen..
>>
>> This log message isn't correct. The callee never specified anything about
>> the usage of a specific ptime. Furthermore, according RFC 3264  the ptime
>> doesn't specify the frame size, which will be used to send packages by the
>> side, which specify it in the SDP. In the RFC 3264 is written:
>>
>> If the ptime attribute is present for a stream, it indicates the
>> desired packetization interval that the offerer would like to
>> receive .
>>
>> ...
>>
>> There is now requirement that the packetization interval be the same in
>> each direction for a particular stream.
>>
>> IMHO that means, that it isn't possible in principle that a device is lying
>> about it's ptime usage, because it only specify by the media attribute the
>> packetization it likes to receive and doesn't specify the packetization it
>> will use itself.
>>
>> For fixing "the problem" Freeswitch sends a re-INVITE message at 1164,777.
>> This message includes in the message header "X-Broken-PTIME: Adv=20;
>> Sent=60", and ptime = 60 media attribute.
>> The callee fails to process this re-INVITE and drops the call.
>>
>> I made the trace after I set the newly introduced parameter
>> "passthru_ptime_mismatch=true" (it's documented in the Wiki since
>> yesterday). Does it make sense, that Freeswitch tries to fix any ptime
>> setting if this variable is set to true?
>>
>> If someone wants to examine this issue more detailed, I can provide the
>> Wireshark-cap file of the call and the debug output of Freeswitch.
>>
>>
>> Thank you in advance
>>             Jan
>>
>>
>>
>>
>>
>> --
>> Jan Riedinger Phone : +49-30-39 73 19 66
>> Dipl.-Inf. | Managing Director Fax : +49-30-39 73 19 64
>> E-Mail: riedinger at sns.eu
>> SNS Consult GmbH ICQ : 163-237-041
>> S?dwestkorso 49a MSN : jan at sns-consult.de
>> 14197 Berlin GERMANY Skype : Jan Riedinger
>>
>> AG Charlottenburg - HRB 71973
>>
>> <Ptime Problem Call
>> Trace.tif>_______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> --
>> Jan Riedinger Phone : +49-30-39 73 19 66
>> Dipl.-Inf. | Managing Director Fax : +49-30-39 73 19 64
>> E-Mail: riedinger at sns.eu
>> SNS Consult GmbH ICQ : 163-237-041
>> S?dwestkorso 49a MSN : jan at sns-consult.de
>> 14197 Berlin GERMANY Skype : Jan Riedinger
>>
>> AG Charlottenburg - HRB 71973
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>>
>>
>> --
>> Jan Riedinger Phone : +49-30-39 73 19 66
>> Dipl.-Inf. | Managing Director Fax : +49-30-39 73 19 64
>> E-Mail: riedinger at sns.eu
>> SNS Consult GmbH ICQ : 163-237-041
>> S?dwestkorso 49a MSN : jan at sns-consult.de
>> 14197 Berlin GERMANY Skype : Jan Riedinger
>>
>> AG Charlottenburg - HRB 71973
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>

-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                         E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973



From chenzhanping at gmail.com  Mon Oct 11 05:57:16 2010
From: chenzhanping at gmail.com (=?GB2312?B?1bzGvbPC?=)
Date: Mon, 11 Oct 2010 20:57:16 +0800
Subject: [Freeswitch-users] Question about mod_nibblebill.
In-Reply-To: <4CB20BA0.6020506@solomo.de>
References: <AANLkTimgF20e6Qk4QrY3rrA-6_uh1tNC=SVNc7-eBs1A@mail.gmail.com>
	<4CB20BA0.6020506@solomo.de>
Message-ID: <AANLkTimrCrOrW9we31-EvHU2qaDc3w65e-T54RZQQYo6@mail.gmail.com>

Thank u very much.

Now,I have another question about mod_nibblebill Module:

I found that charging is completed, b-leg has variables
variable_nibble_total_billed,
 but a-leg without this variable.

If the called side hangs up, the desired b-leg variable
variable_nibble_total_billed is impossible.

May I ask, how can I get the variable variable_nibble_total_billed after
b-leg hangup.


Thanks.


? 2010?10?11? ??2:53?Daniel Neubert <daniel.neubert at solomo.de>???

> Unfortunately I think that this is a scenario that is not covered by
> mod_nibblebill. The module is not capable of charging customers this way.
>
> We've solved this by using the nibble_rate as you do to realize a prepaid
> billing setup. After the final xml CDR is posted to our billing system via
> curl, we calculate the final price that will be billed to the customer and
> refund the amount that has been charged by mod_nibblebill. This works pretty
> good - but there is a catch: Customers can theoretically run below an amount
> of 0,00 - so we had to configure a nobal_amt to prevent this.
>
> Viele Gr??e / Regards
> Daniel Neubert
>
>
> Am 09.10.2010 03:12, schrieb ???:
>
> Hello, I have a question about the mod_nibblebill module of freeswitch.
>
>
> I can use mod_nibblebill to billing. But I found that his billing was not
> what I want.
>
>
> I want to achieve in this way: every 60 seconds for an interval of less
> than 60 seconds, 60 seconds to install basis, charging $ 0.1 for each
> billing interval. For example: the user dials 45 seconds, charging $ 0.1. If
> user dials 72 seconds, charging $ 0.2.
>
>
> My current configuration in dialplain\default.xml is as follows:
>
> <action application="bridge"
> data="{enable_heartbeat_events=60,nibble_rate=0.1,nibble_account=1000}user/${dialed_extension}@${domain_name}"/>
>
>
> May I ask, mod_nibblebill module can not achieve my desired function, if
> realized, how can I configure?
>
> Thank you.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ken at ukgb.net  Mon Oct 11 07:05:19 2010
From: ken at ukgb.net (Ken Gillett)
Date: Mon, 11 Oct 2010 15:05:19 +0100
Subject: [Freeswitch-users] SIP behaviour
Message-ID: <873888F9-9FF2-4B58-936D-3A8E2BCA17A6@ukgb.net>

I'm trying to get to the bottom of some strange SIP behaviour and hoped someone here might have an idea what's going on.

2 different customers have accounts with the same VOIP provider and they have several SIP accounts each. One of the customers (me) can easily have multiple devices registered to the same SIP account so they all ring on incoming calls, i.e. exactly is as it should be. But the other customer has a problem with multiple registrations. A single device can register ok and keeps working, but once a second device is simultaneously registered to the same SIP account, the first one may be dropped after a couple of minutes. If a third device is registered, then one of the others will definitely be dropped, within no more than 10 mins. While a device is registered, calls work perfectly.

Initially I suspected that one of the SIP client devices was causing this behaviour, but that has now been de-registered and is no longer relevant, but the problem is still there. I am now suspecting the NAT router being used to connect this local network onto the Internet, but how possible is this? Can a NAT router be having an effect like this, making multiple registrations impossible? I must say I cannot see how, but it is one thing that is different between the 2 setups. What else could cause registrations to fail like this? Can anyone suggest any possible solutions to this?

I have spoken to the VOIP provider, but they are of the opinion that multiple registrations don't work and are not supported. However, it is part of the SIP spec to support this functionality and it obviously DOES work as I don't have problems with up to 4 devices registered to the same account and they all work all of the time. However, it does mean they are reluctant to put any effort into troubleshooting it.

Anyone got any ideas about why multiple registrations are failing in one instance, but not another?




Ken G i l l e t t

_/_/_/_/_/_/_/_/





From awais-nazeer at hotmail.com  Mon Oct 11 01:32:16 2010
From: awais-nazeer at hotmail.com (awais nazir)
Date: Mon, 11 Oct 2010 14:32:16 +0600
Subject: [Freeswitch-users] G723 bitrate 5.3 kbps
Message-ID: <COL110-W61AD88FA95894D4F32FFDC88530@phx.gbl>


 

Hi, 

 

>
>
>
>       


The default bitrate in freeswitch for g723 is 6.3 kbps and we need to use 5.3 kbps. Please advise?

freeswitch-users at lists.freeswitch.org

 

 

<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<,
 		 	   		  
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From edpimentl at gmail.com  Mon Oct 11 07:11:03 2010
From: edpimentl at gmail.com (EdPimentl)
Date: Mon, 11 Oct 2010 10:11:03 -0400
Subject: [Freeswitch-users] Increase in Voip attacks
Message-ID: <AANLkTin2ors5xJnSVZ_M1U1kMOi=TiCO05J60Gec7EUr@mail.gmail.com>

http://it.slashdot.org/story/10/10/10/2313255/In-Australia-Rising-VoIP-Attacks-Mean-Huge-Bills-For-Victims?from=rss
http://isc.sans.edu/diary.html?storyid=9193

-E
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From brett at woollum.com  Sat Oct  9 01:10:49 2010
From: brett at woollum.com (Brett Woollum)
Date: Sat, 9 Oct 2010 01:10:49 -0700 (PDT)
Subject: [Freeswitch-users] Sharing storage between servers
In-Reply-To: <AANLkTi=8kH5-c8pL1K_KRsNBo2gEO4y+24iwCk=AkJ3D@mail.gmail.com>
Message-ID: <760758090.3041286611849227.JavaMail.root@mail>

Is there a way to store voicemail to a MySQL cluster like Asterisk does? This is one of the issues I have been trying to find an answer for, but I haven't been able to. It would be great if there were a LUA/Javascript command that would stream audio to the call directly from the database, or the other way around. This would be useful for voicemail recordings, IVR prompts, etc, when running several clustered FS servers. I have a MySQL Cluster spread across each of my FS servers that should be able to handle the additional load. Using MySQL cluster as an on-demand audio storage system would make access to the audio files fault-tolerant and distributed (even to other servers such as a web server for web voicemail playback). 

Any ideas on how this could be achieved? 


Brett Woollum 
Brett at Woollum.com 


----- Original Message ----- 
From: "Yehavi Bourvine" <yehavi.bourvine at gmail.com> 
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org> 
Sent: Saturday, October 9, 2010 12:49:51 AM GMT -08:00 US/Canada Pacific 
Subject: Re: [Freeswitch-users] Sharing storage between servers 



Hi, 

We use a NAS server to share the voicemail between two servers (one is FS, the other is WEB interface we wrote to handle voicemail via WEB). 
For the database: we use MySQL with replication. 

Regards, __Yehavi: 



2010/10/9 Jody Rudolph < jody.rudolph at gmail.com > 


I am curious as to just how far you can take sharing disk storage for the purpose of clustering. Is anyone doing this with the voicemail storage directories? Is it possible to share the SQLite internal databases to avoid resorting to ODBC? I realize that isn't likely, but I have access to some high performance SAN hardware and want to take the most advantage possible. 


Thanks, 
Jody Rudolph 

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




_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 
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From Russell.Mosemann at cune.org  Sat Oct  9 07:01:36 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sat, 9 Oct 2010 09:01:36 -0500
Subject: [Freeswitch-users] outbound calling error: multiple PRIs with
	Sangoma + FS
In-Reply-To: <AANLkTim-UWMzVNkdnTyZnESkKTou=VqN0Y3_UU=B80oX@mail.gmail.com>
References: <AANLkTimn7P64Exf9JTwdLZB8Vam8wBLuvrLS5MY+xAwT@mail.gmail.com><AANLkTin01O6WPqXfXN+aiaPcOd+Ww1HiFwzcbUEeMz2Z@mail.gmail.com>
	<AANLkTim-UWMzVNkdnTyZnESkKTou=VqN0Y3_UU=B80oX@mail.gmail.com>
Message-ID: <625156AD55324D3DB824F758A190C370@cune.pri>

Neil Patel typed:
>        <param name="q921loglevel" value="alert"/>
>        <param name="q931loglevel" value="alert"/>
>        <param name="mode" value="user"/>
>        <param name="dialect" value="5ess"/>
...
> 2010-10-09 18:28:58.487311 [WARNING] ftmod_sangoma_isdn_cfg.c:262 Ignoring
> unknown parameter q921loglevel
> 2010-10-09 18:28:58.487369 [WARNING] ftmod_sangoma_isdn_cfg.c:262 Ignoring
> unknown parameter q931loglevel
> 2010-10-09 18:28:58.487425 [WARNING] ftmod_sangoma_isdn_cfg.c:262 Ignoring
> unknown parameter mode
> 2010-10-09 18:28:58.487480 [WARNING] ftmod_sangoma_isdn_cfg.c:262 Ignoring
> unknown parameter dialect

http://wiki.sangoma.com/wanpipe-api-freetdm
http://wiki.sangoma.com/Freeswitch-FreeTDM-Sangoma-ISDN-Library


mose



From anthony.minessale at gmail.com  Mon Oct 11 07:59:29 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Oct 2010 09:59:29 -0500
Subject: [Freeswitch-users] iLBC codec generating only noise.
In-Reply-To: <AANLkTimmMM8Ua9AP9BwXU0yDfP9FReK2e9w-iMRiMVFu@mail.gmail.com>
References: <AANLkTinz4JEfyGnjU-4nneOFg=A0qXHjVw9xE=jp2qSo@mail.gmail.com>
	<AANLkTinvkxRaS+Wmso1vubuH3Pj4Mib96B32Bx+sqR7H@mail.gmail.com>
	<AANLkTimmMM8Ua9AP9BwXU0yDfP9FReK2e9w-iMRiMVFu@mail.gmail.com>
Message-ID: <AANLkTimiSPgCzs31VjEp0x8g1qDe-kzppsDM7uig0YGU@mail.gmail.com>

you can't mix ptimes in your codec list, they will be filtered out.
I can not reproduce any problems on latest GIT, have you done a complete build?

On Sat, Oct 9, 2010 at 7:43 AM, xuyan yang <xyangni at gmail.com> wrote:
> Hi?Anthony,
> Tried the latest, it works when both lag A and B are using iLBC. But there
> are still some problems:
> 1, when lag A use PCMU and lag B use iLBC or the reverse, the PCMU side hear
> only noise even if it is muted.
> 2, when added "iLBC at 30i" to codec list, it is ignored. traced the debug,
> iLBC is not compared as a choice. So adding only "iLBC" should be used as
> mitigation.
>
>
>
> On Sat, Oct 9, 2010 at 11:34 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> Try latest
>>
>> On Oct 8, 2010 2:28 PM, "xuyan yang" <xyangni at gmail.com> wrote:
>> > Hi,
>> >
>> > I am trying to use iLBC codec with both eyebeam and some iphone client.
>> > When
>> > a ivr is called, the client can here system voice and make dtmf input.
>> > but
>> > the voice recorded from client's microphone is only noise.
>> > The call between 2 clients also have such problem.
>> >
>> > Sometimes, FS may even got crashed with the following information:
>> >
>> > 2010-10-08 19:56:21.343750 [INFO] switch_rtp.c:2527 Auto Changing port
>> > from
>> > 10.20.132.244:18570 to 8
>> > 2.132.139.197:19536
>> > alloc: asked for negative size -2147483648
>> >
>> > in var.xml, I have added iLBC instead of iLBC at 30i, because iLBC at 30i is
>> > ignored during codec negotiation.
>> > <X-PRE-PROCESS cmd="set"
>> > data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h
>> > ,G722,speex at 8000h@20i,iLBC,PCMU,PCMA,GSM"/>
>> >
>> > The git version last week is used in my test. Is there anything wrong
>> > with
>> > my setup? Thanks.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From brian at ubintel.com  Sat Oct  9 02:58:50 2010
From: brian at ubintel.com (Brian Yglesias)
Date: Sat, 9 Oct 2010 05:58:50 -0400 (EDT)
Subject: [Freeswitch-users] Adding openzap channels to a call group or the
 functional equivalent.
Message-ID: <7251919.62762.1286618330278.JavaMail.root@flanders.anglerlabs.com>

I'm trying to dial 1000 AND some analog extensions on inbound calls.

Essentially, I need to add analog extensions to a call group, or the functional equivalent.

To put it another way, I'm migrating from asterisk, and I need to know how to dial(sip/1000&openzap/1/1).

This is not in the book, nor is there any documentation that I've been able to find that achieves this very basic hybrid PBX functionality.

Thanks in advance.


From davidwaf at gmail.com  Mon Oct 11 07:12:19 2010
From: davidwaf at gmail.com (David Wafula)
Date: Mon, 11 Oct 2010 16:12:19 +0200
Subject: [Freeswitch-users] SIP Registration Failing
Message-ID: <AANLkTi=1OQ4uUS8Ecb-NaoM6AT3--fgGquqb_nvvJzdV@mail.gmail.com>

Somehow, am unable to register a softphone, no matter what i do. I decided
to run wireshark from the machine running the softphone, and so i get the
following:
14485 3007.314022 146.141.76.164 146.141.76.153 ICMP Destination unreachable
(Port unreachable)

146.141.76.164 is the softphone machine, 146.141.76.153 is the server. This
certainly tells me problem, the port, right? But how could it be?

when i run sofia status on the server, it shows me , among many things:

SIP-IP:        146.141.76.153
URL:           sip:mod_sofia at 146.141.76.153:5060
BIND-URL:  sip:mod_sofia at 146.141.76.153:5060

I assume this port 5060 is the problematic one? And, am running these in a
LAN, all ports are as open as ever.
Thanks again for your help.
-- 
David Wafula
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From davidwaf at gmail.com  Mon Oct 11 07:20:55 2010
From: davidwaf at gmail.com (David Wafula)
Date: Mon, 11 Oct 2010 16:20:55 +0200
Subject: [Freeswitch-users] SIP Registration Failing
In-Reply-To: <AANLkTi=1OQ4uUS8Ecb-NaoM6AT3--fgGquqb_nvvJzdV@mail.gmail.com>
References: <AANLkTi=1OQ4uUS8Ecb-NaoM6AT3--fgGquqb_nvvJzdV@mail.gmail.com>
Message-ID: <AANLkTim5-SciJk55nzengsXNGgd2WtT5HY475JuxOG9U@mail.gmail.com>

On Mon, Oct 11, 2010 at 4:12 PM, David Wafula <davidwaf at gmail.com> wrote:

> Somehow, am unable to register a softphone, no matter what i do. I decided
> to run wireshark from the machine running the softphone, and so i get the
> following:


Just to clarify, the softphone logs the following registration request:

2010-10-11 16:16:59,213 SENT to 146.141.76.153/5060 [Transaction timer]

REGISTER sip:146.141.76.153 SIP/2.0
Via: SIP/2.0/UDP 146.141.76.164:5060;branch=z9hG4bKl6f2ot4Un
Max-Forwards: 70
To: <sip:1000 at 146.141.76.153 <sip%3A1000 at 146.141.76.153>>
From: <sip:1000 at 146.141.76.153 <sip%3A1000 at 146.141.76.153>>;tag=eCje4MAR
Call-ID: Dh8fgLjS-1286806587644 at pc164.seg76.wits.ac.za
CSeq: 0 REGISTER
Contact: <sip:1000 at 146.141.76.164:5060;transport=UDP>


But for the life of me, cant just figure why the registration fails ..
-- 
David Wafula
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From nicolas at medularis.com  Mon Oct 11 02:15:02 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 11 Oct 2010 06:15:02 -0300
Subject: [Freeswitch-users] subscribe
Message-ID: <AANLkTi=EmhJR-D4MfQxmkkDpFO9NNDP18wT-TZkOegFP@mail.gmail.com>

subscribe
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From nico at clickfono.com  Mon Oct 11 02:46:11 2010
From: nico at clickfono.com (Nicolas Brenner)
Date: Mon, 11 Oct 2010 06:46:11 -0300
Subject: [Freeswitch-users] Binding to virtual interface (ipsec vpn)
Message-ID: <AANLkTikWrRy41kskcaC-0LLEdbNd9d6hi1W26pw2ORg+@mail.gmail.com>

Hello everyone,

I'm having trouble binding FS to another ip on a virtual interface (eth0:3).
The virtual interface's ip address is the gateway to an ipsec vpn, and the
only way to access this voip provider's sip server, is through the vpn.

I tried setting up an additional sip profile, with rtp-ip and sip-ip as the
virtual interface's ip, but when I do that, no packets are sent at all when
trying to make a call with originate through the cli (according to tcpdump).
I tried also setting ext-rtp-ip and ext-sip-ip to the same virtual
interface's ip, but nothing happened, still no packets being sent (according
to tcpdump) when trying to make a call. Finally I tried only
setting ext-rtp-ip and ext-sip-ip to the virtual interface's ip, and leave
rtp-ip and sip-ip to $${local_ip_v4}, then tcpdump started seeing some
traffic (when trying to originate a call) which I was able to capture. When
I loaded the trace on wireshark though, I noticed the source ip being used
to send all the (sip) packets, is the one from eth0 (public ip), and not
eth0:3's ip (private ip for vpn).

Has anyone had any similiar experiences? how were you able to solve it?

Thanks for your help!


Nico
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From rupa at rupa.com  Mon Oct 11 08:31:48 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 11 Oct 2010 10:31:48 -0500
Subject: [Freeswitch-users] Adding openzap channels to a call group or
 the functional equivalent.
In-Reply-To: <7251919.62762.1286618330278.JavaMail.root@flanders.anglerlabs.com>
References: <7251919.62762.1286618330278.JavaMail.root@flanders.anglerlabs.com>
Message-ID: <AANLkTinS1Mz_Etx-dC-rpZcw5zJgYTzEH5U_kN=u_SBs@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Calling_multiple_destinations

On Sat, Oct 9, 2010 at 4:58 AM, Brian Yglesias <brian at ubintel.com> wrote:

> I'm trying to dial 1000 AND some analog extensions on inbound calls.
>
> Essentially, I need to add analog extensions to a call group, or the
> functional equivalent.
>
> To put it another way, I'm migrating from asterisk, and I need to know how
> to dial(sip/1000&openzap/1/1).
>
> This is not in the book, nor is there any documentation that I've been able
> to find that achieves this very basic hybrid PBX functionality.
>
> Thanks in advance.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From curriegrad2004 at gmail.com  Mon Oct 11 09:13:19 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Mon, 11 Oct 2010 09:13:19 -0700
Subject: [Freeswitch-users] SIP Registration Failing
In-Reply-To: <AANLkTim5-SciJk55nzengsXNGgd2WtT5HY475JuxOG9U@mail.gmail.com>
References: <AANLkTi=1OQ4uUS8Ecb-NaoM6AT3--fgGquqb_nvvJzdV@mail.gmail.com>
	<AANLkTim5-SciJk55nzengsXNGgd2WtT5HY475JuxOG9U@mail.gmail.com>
Message-ID: <AANLkTi=xqJEnR3eFVdKE4Yzpu-O_OrbxfoEFMimiOX=7@mail.gmail.com>

Local clients actually register at port 5080 for the internal context.
You can change that under sofia's sip profile configuration.

On Mon, Oct 11, 2010 at 7:20 AM, David Wafula <davidwaf at gmail.com> wrote:
>
>
> On Mon, Oct 11, 2010 at 4:12 PM, David Wafula <davidwaf at gmail.com> wrote:
>>
>> Somehow, am unable to register a softphone, no matter what i do. I decided
>> to run wireshark from the machine running the softphone, and so i get the
>> following:
>
>
> Just to clarify, the softphone logs the following registration request:
> 2010-10-11 16:16:59,213 SENT to 146.141.76.153/5060 [Transaction timer]
> REGISTER sip:146.141.76.153 SIP/2.0
> Via: SIP/2.0/UDP 146.141.76.164:5060;branch=z9hG4bKl6f2ot4Un
> Max-Forwards: 70
> To: <sip:1000 at 146.141.76.153>
> From: <sip:1000 at 146.141.76.153>;tag=eCje4MAR
> Call-ID: Dh8fgLjS-1286806587644 at pc164.seg76.wits.ac.za
> CSeq: 0 REGISTER
> Contact: <sip:1000 at 146.141.76.164:5060;transport=UDP>
>
> But for the life of me, cant just figure why the registration fails ..
> --
> David Wafula
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From curriegrad2004 at gmail.com  Mon Oct 11 09:15:59 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Mon, 11 Oct 2010 09:15:59 -0700
Subject: [Freeswitch-users] Increase in Voip attacks
In-Reply-To: <AANLkTin2ors5xJnSVZ_M1U1kMOi=TiCO05J60Gec7EUr@mail.gmail.com>
References: <AANLkTin2ors5xJnSVZ_M1U1kMOi=TiCO05J60Gec7EUr@mail.gmail.com>
Message-ID: <AANLkTinYfesUKHEx1aWdx-no3vgz_i6q2FujjkK1HbbA@mail.gmail.com>

Generate some random length password and hope this doesn't even happen
at all. Oh well, that's why freeswitch has 2 security contexts, one
public and one internal.

On Mon, Oct 11, 2010 at 7:11 AM, EdPimentl <edpimentl at gmail.com> wrote:
> http://it.slashdot.org/story/10/10/10/2313255/In-Australia-Rising-VoIP-Attacks-Mean-Huge-Bills-For-Victims?from=rss
> http://isc.sans.edu/diary.html?storyid=9193
>
> -E
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mario_fs at mgtech.com  Mon Oct 11 10:10:26 2010
From: mario_fs at mgtech.com (Mario)
Date: Mon, 11 Oct 2010 10:10:26 -0700
Subject: [Freeswitch-users] Does gateway tcport=#### work?
Message-ID: <4CB34502.8060202@mgtech.com>

I put contact-params" value="tport=5069 in the gateway but it still uses
5080. This was in several samples in the wiki.
Mario


From anthony.minessale at gmail.com  Mon Oct 11 10:24:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Oct 2010 12:24:38 -0500
Subject: [Freeswitch-users] Does gateway tcport=#### work?
In-Reply-To: <4CB34502.8060202@mgtech.com>
References: <4CB34502.8060202@mgtech.com>
Message-ID: <AANLkTikLutffMZ3eT8dREugRMEWGn+b0X1WbEr0irZTa@mail.gmail.com>

you are trying to use a profile param in a gateway.

On Mon, Oct 11, 2010 at 12:10 PM, Mario <mario_fs at mgtech.com> wrote:
> I put contact-params" value="tport=5069 in the gateway but it still uses
> 5080. This was in several samples in the wiki.
> Mario
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From brian at freeswitch.org  Mon Oct 11 10:33:52 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 11 Oct 2010 12:33:52 -0500
Subject: [Freeswitch-users] Binding to virtual interface (ipsec vpn)
In-Reply-To: <AANLkTikWrRy41kskcaC-0LLEdbNd9d6hi1W26pw2ORg+@mail.gmail.com>
References: <AANLkTikWrRy41kskcaC-0LLEdbNd9d6hi1W26pw2ORg+@mail.gmail.com>
Message-ID: <1F64D322-4010-4C3E-B622-93A6254B6E1B@freeswitch.org>

If you set the IP in sip-ip and rtp-ip that is abound to eth0:3 then it should bind to that interface/ip and use that... I haven't see anything out of the usual when doing this.

Show me your profile config and the output of ifconfig -a

/b

On Oct 11, 2010, at 4:46 AM, Nicolas Brenner wrote:

> Hello everyone,
> 
> I'm having trouble binding FS to another ip on a virtual interface (eth0:3). The virtual interface's ip address is the gateway to an ipsec vpn, and the only way to access this voip provider's sip server, is through the vpn.
> 
> I tried setting up an additional sip profile, with rtp-ip and sip-ip as the virtual interface's ip, but when I do that, no packets are sent at all when trying to make a call with originate through the cli (according to tcpdump). I tried also setting ext-rtp-ip and ext-sip-ip to the same virtual interface's ip, but nothing happened, still no packets being sent (according to tcpdump) when trying to make a call. Finally I tried only setting ext-rtp-ip and ext-sip-ip to the virtual interface's ip, and leave rtp-ip and sip-ip to $${local_ip_v4}, then tcpdump started seeing some traffic (when trying to originate a call) which I was able to capture. When I loaded the trace on wireshark though, I noticed the source ip being used to send all the (sip) packets, is the one from eth0 (public ip), and not eth0:3's ip (private ip for vpn).
> 
> Has anyone had any similiar experiences? how were you able to solve it?
> 
> Thanks for your help!
> 
> 
> Nico



From brian at freeswitch.org  Mon Oct 11 10:35:25 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 11 Oct 2010 12:35:25 -0500
Subject: [Freeswitch-users] Increase in Voip attacks
In-Reply-To: <AANLkTinYfesUKHEx1aWdx-no3vgz_i6q2FujjkK1HbbA@mail.gmail.com>
References: <AANLkTin2ors5xJnSVZ_M1U1kMOi=TiCO05J60Gec7EUr@mail.gmail.com>
	<AANLkTinYfesUKHEx1aWdx-no3vgz_i6q2FujjkK1HbbA@mail.gmail.com>
Message-ID: <EB49EFBB-EAF5-497A-B21E-EFF9C1BA908E@freeswitch.org>

The security context's are limitless.  The default has two (actually 3 if you count features).  see scripts/perl/blacklist.pl and scripts/perl/honeypot.pl

That friendly-scanner that hits you with register packets does it so hard and so fast that it might bring your machine to a crawl on register traffic.

/b

On Oct 11, 2010, at 11:15 AM, Jeffrey Leung wrote:

> Generate some random length password and hope this doesn't even happen
> at all. Oh well, that's why freeswitch has 2 security contexts, one
> public and one internal.



From brian at freeswitch.org  Mon Oct 11 10:38:57 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 11 Oct 2010 12:38:57 -0500
Subject: [Freeswitch-users] SIP Registration Failing
In-Reply-To: <AANLkTi=xqJEnR3eFVdKE4Yzpu-O_OrbxfoEFMimiOX=7@mail.gmail.com>
References: <AANLkTi=1OQ4uUS8Ecb-NaoM6AT3--fgGquqb_nvvJzdV@mail.gmail.com>
	<AANLkTim5-SciJk55nzengsXNGgd2WtT5HY475JuxOG9U@mail.gmail.com>
	<AANLkTi=xqJEnR3eFVdKE4Yzpu-O_OrbxfoEFMimiOX=7@mail.gmail.com>
Message-ID: <C6F2FD8F-8820-469E-833C-F60280A9BE48@freeswitch.org>

The default has 5060 for phones to register to... 5080 is where outbound register requests go from to say providers.  Never assume because something is done a certain way in the defaults that its the ONLY way you can do things.  I could make the defaults so complex that nobody would ever understand them.  I am going to do another config setup that is a bit more planned out... the defaults now do not show off a lot of stuff FreeSWITCH is capable of.

/b

On Oct 11, 2010, at 11:13 AM, Jeffrey Leung wrote:

> Local clients actually register at port 5080 for the internal context.
> You can change that under sofia's sip profile configuration.
> 
> On Mon, Oct 11, 2010 at 7:20 AM, David Wafula <davidwaf at gmail.com> wrote:
> 



From mario_fs at mgtech.com  Mon Oct 11 10:42:06 2010
From: mario_fs at mgtech.com (Mario)
Date: Mon, 11 Oct 2010 10:42:06 -0700
Subject: [Freeswitch-users] Does gateway tcport=#### work?
In-Reply-To: <AANLkTikLutffMZ3eT8dREugRMEWGn+b0X1WbEr0irZTa@mail.gmail.com>
References: <4CB34502.8060202@mgtech.com>
	<AANLkTikLutffMZ3eT8dREugRMEWGn+b0X1WbEr0irZTa@mail.gmail.com>
Message-ID: <4CB34C6E.2070206@mgtech.com>

I found the example in three gateway examples as the FS mail-list entry
below. I thought it might be controllable by gateway but it looks like
your have to make it universal in external.xml. Thanks.

http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg17020.html

On 10/11/10 10:24, Anthony Minessale wrote:
> you are trying to use a profile param in a gateway.
> 
> On Mon, Oct 11, 2010 at 12:10 PM, Mario <mario_fs at mgtech.com> wrote:
>> I put contact-params" value="tport=5069 in the gateway but it still uses
>> 5080. This was in several samples in the wiki.
>> Mario
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 


From mnhassan at usa.net  Mon Oct 11 10:45:59 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Mon, 11 Oct 2010 23:45:59 +0600
Subject: [Freeswitch-users] G723 bitrate 5.3 kbps
In-Reply-To: <COL110-W61AD88FA95894D4F32FFDC88530@phx.gbl>
References: <COL110-W61AD88FA95894D4F32FFDC88530@phx.gbl>
Message-ID: <AANLkTi=tfQodYha5Drx_OUqkkQ52KcOZu-_76BjYGxUq@mail.gmail.com>

The G.723 with FreeSwitch operates in passthrough mode only, and
should work for both bitrates. Are you trying in passthrough setups?

Regards
HASSAN


On 2010-10-11, awais nazir <awais-nazeer at hotmail.com> wrote:
>
>
>
> Hi,
>
>
>
>>
>>
>>
>>
>
>
> The default bitrate in freeswitch for g723 is 6.3 kbps and we need to use
> 5.3 kbps. Please advise?
>
> freeswitch-users at lists.freeswitch.org
>
>
>
>
>
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<,
>  		 	   		

-- 
Sent from my mobile device


From david.ponzone at ipeva.fr  Mon Oct 11 11:15:43 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 11 Oct 2010 20:15:43 +0200
Subject: [Freeswitch-users] SIP Registration Failing
In-Reply-To: <AANLkTi=1OQ4uUS8Ecb-NaoM6AT3--fgGquqb_nvvJzdV@mail.gmail.com>
References: <AANLkTi=1OQ4uUS8Ecb-NaoM6AT3--fgGquqb_nvvJzdV@mail.gmail.com>
Message-ID: <B8B73B88-95A0-44A7-A921-DB620F33AA43@ipeva.fr>

David,

that wireshark trace says that your FS sent a packet to your softphone  
on a specific UDP port, and the TCP/IP stack of the machine where the  
softphone runs is replying that the port is not opened with a ICMP  
port unreachable.
That's odd.

Can you trace the whole sequence of registration ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 11/10/2010 ? 16:12, David Wafula a ?crit :

> Somehow, am unable to register a softphone, no matter what i do. I  
> decided to run wireshark from the machine running the softphone, and  
> so i get the following:
> 14485	3007.314022	146.141.76.164	146.141.76.153	ICMP	Destination  
> unreachable (Port unreachable)
>
> 146.141.76.164 is the softphone machine, 146.141.76.153 is the  
> server. This certainly tells me problem, the port, right? But how  
> could it be?
>
> when i run sofia status on the server, it shows me , among many  
> things:
>
> SIP-IP:        146.141.76.153
> URL:           sip:mod_sofia at 146.141.76.153:5060
> BIND-URL:  sip:mod_sofia at 146.141.76.153:5060
>
> I assume this port 5060 is the problematic one? And, am running  
> these in a LAN, all ports are as open as ever.
> Thanks again for your help.
> -- 
> David Wafula
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at ipeva.fr  Mon Oct 11 11:15:43 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 11 Oct 2010 20:15:43 +0200
Subject: [Freeswitch-users] SIP Registration Failing
In-Reply-To: <AANLkTi=1OQ4uUS8Ecb-NaoM6AT3--fgGquqb_nvvJzdV@mail.gmail.com>
References: <AANLkTi=1OQ4uUS8Ecb-NaoM6AT3--fgGquqb_nvvJzdV@mail.gmail.com>
Message-ID: <B8B73B88-95A0-44A7-A921-DB620F33AA43@ipeva.fr>

David,

that wireshark trace says that your FS sent a packet to your softphone  
on a specific UDP port, and the TCP/IP stack of the machine where the  
softphone runs is replying that the port is not opened with a ICMP  
port unreachable.
That's odd.

Can you trace the whole sequence of registration ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 11/10/2010 ? 16:12, David Wafula a ?crit :

> Somehow, am unable to register a softphone, no matter what i do. I  
> decided to run wireshark from the machine running the softphone, and  
> so i get the following:
> 14485	3007.314022	146.141.76.164	146.141.76.153	ICMP	Destination  
> unreachable (Port unreachable)
>
> 146.141.76.164 is the softphone machine, 146.141.76.153 is the  
> server. This certainly tells me problem, the port, right? But how  
> could it be?
>
> when i run sofia status on the server, it shows me , among many  
> things:
>
> SIP-IP:        146.141.76.153
> URL:           sip:mod_sofia at 146.141.76.153:5060
> BIND-URL:  sip:mod_sofia at 146.141.76.153:5060
>
> I assume this port 5060 is the problematic one? And, am running  
> these in a LAN, all ports are as open as ever.
> Thanks again for your help.
> -- 
> David Wafula
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From astmac at stillnewt.org  Mon Oct 11 12:00:19 2010
From: astmac at stillnewt.org (Martin Joseph)
Date: Mon, 11 Oct 2010 12:00:19 -0700
Subject: [Freeswitch-users] Newbie (sort of) questions
Message-ID: <DFB4B88E-6834-4D22-9DEE-616AB50F0D91@stillnewt.org>

Hi Again FreeSWITCHers,

I am building a new setup for my "new" house (I have been remodeling for 5 years).

I have build FS via GIT on my "new" xServe hardware (dual Xeon 2Ghz) which I was able to do pretty easily (Hurray!).  In the past I have struggled with updating my older FS install on my anemic g3/500 ibook install which is still running by the way.

So,  now I am thinking about configuring my new installation, and I being no expert am not looking forward to figuring this all out again. My installation is overall pretty simple with a gateway to POTS (audiocodes MP114) and several Voip providors, one of whom is hopefully still in business (heh).

Anyhow,  my first request is for GUI based FS configuration systems?  These could run on my OSX box (via X11 or Cocoa), or be some kind of web based thing.  It doesn't need to be super customizable, but not having to find/edit the XML files would be lovely. I am a mac person after all.

If there are no recommended GUI's available, then pointing me to the best sample pages for configuration would also be greatly appreciated...

Thanks again for the great software and all of your efforts!
Marty

PS looking forward to killing me last Asterisk install...

From anthony.minessale at gmail.com  Mon Oct 11 12:10:18 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Oct 2010 14:10:18 -0500
Subject: [Freeswitch-users] SIP behaviour
In-Reply-To: <873888F9-9FF2-4B58-936D-3A8E2BCA17A6@ukgb.net>
References: <873888F9-9FF2-4B58-936D-3A8E2BCA17A6@ukgb.net>
Message-ID: <AANLkTinWnwVrx-py2jhdyKV9eiFtAu09OdWw2HmbNUz4@mail.gmail.com>

Maybe they are going through the same firewall on the one that doesn't
work and port mapping the same path over each other.

try setting different local sip ports on each phone.


On Mon, Oct 11, 2010 at 9:05 AM, Ken Gillett <ken at ukgb.net> wrote:
> I'm trying to get to the bottom of some strange SIP behaviour and hoped someone here might have an idea what's going on.
>
> 2 different customers have accounts with the same VOIP provider and they have several SIP accounts each. One of the customers (me) can easily have multiple devices registered to the same SIP account so they all ring on incoming calls, i.e. exactly is as it should be. But the other customer has a problem with multiple registrations. A single device can register ok and keeps working, but once a second device is simultaneously registered to the same SIP account, the first one may be dropped after a couple of minutes. If a third device is registered, then one of the others will definitely be dropped, within no more than 10 mins. While a device is registered, calls work perfectly.
>
> Initially I suspected that one of the SIP client devices was causing this behaviour, but that has now been de-registered and is no longer relevant, but the problem is still there. I am now suspecting the NAT router being used to connect this local network onto the Internet, but how possible is this? Can a NAT router be having an effect like this, making multiple registrations impossible? I must say I cannot see how, but it is one thing that is different between the 2 setups. What else could cause registrations to fail like this? Can anyone suggest any possible solutions to this?
>
> I have spoken to the VOIP provider, but they are of the opinion that multiple registrations don't work and are not supported. However, it is part of the SIP spec to support this functionality and it obviously DOES work as I don't have problems with up to 4 devices registered to the same account and they all work all of the time. However, it does mean they are reluctant to put any effort into troubleshooting it.
>
> Anyone got any ideas about why multiple registrations are failing in one instance, but not another?
>
>
>
>
> Ken G i l l e t t
>
> _/_/_/_/_/_/_/_/
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From mnhassan at usa.net  Mon Oct 11 12:10:45 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Tue, 12 Oct 2010 01:10:45 +0600
Subject: [Freeswitch-users] Newbie (sort of) questions
In-Reply-To: <DFB4B88E-6834-4D22-9DEE-616AB50F0D91@stillnewt.org>
References: <DFB4B88E-6834-4D22-9DEE-616AB50F0D91@stillnewt.org>
Message-ID: <AANLkTim+zr9DWNQ9QCVccMETYb8Y=OT89t8QhjyfWVv4@mail.gmail.com>

There is a new project, which looks quite promising:

www.2600hz.org

Regards
HASSAN


On 2010-10-12, Martin Joseph <astmac at stillnewt.org> wrote:
> Hi Again FreeSWITCHers,
>
> I am building a new setup for my "new" house (I have been remodeling for 5
> years).
>
> I have build FS via GIT on my "new" xServe hardware (dual Xeon 2Ghz) which I
> was able to do pretty easily (Hurray!).  In the past I have struggled with
> updating my older FS install on my anemic g3/500 ibook install which is
> still running by the way.
>
> So,  now I am thinking about configuring my new installation, and I being no
> expert am not looking forward to figuring this all out again. My
> installation is overall pretty simple with a gateway to POTS (audiocodes
> MP114) and several Voip providors, one of whom is hopefully still in
> business (heh).
>
> Anyhow,  my first request is for GUI based FS configuration systems?  These
> could run on my OSX box (via X11 or Cocoa), or be some kind of web based
> thing.  It doesn't need to be super customizable, but not having to
> find/edit the XML files would be lovely. I am a mac person after all.
>
> If there are no recommended GUI's available, then pointing me to the best
> sample pages for configuration would also be greatly appreciated...
>
> Thanks again for the great software and all of your efforts!
> Marty
>
> PS looking forward to killing me last Asterisk install...
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device


From davidwaf at gmail.com  Mon Oct 11 12:49:28 2010
From: davidwaf at gmail.com (David Wafula)
Date: Mon, 11 Oct 2010 21:49:28 +0200
Subject: [Freeswitch-users] SIP Registration Failing
In-Reply-To: <B8B73B88-95A0-44A7-A921-DB620F33AA43@ipeva.fr>
References: <AANLkTi=1OQ4uUS8Ecb-NaoM6AT3--fgGquqb_nvvJzdV@mail.gmail.com>
	<B8B73B88-95A0-44A7-A921-DB620F33AA43@ipeva.fr>
Message-ID: <AANLkTimcMg0Q_29vh7bHF9NKu2jXwEZQhrEnLM8HcPwX@mail.gmail.com>

On Mon, Oct 11, 2010 at 8:15 PM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> David,
>
> that wireshark trace says that your FS sent a packet to your softphone on a
> specific UDP port, and the TCP/IP stack of the machine where the softphone
> runs is replying that the port is not opened with a ICMP port unreachable.
> That's odd.
>
> Can you trace the whole sequence of registration ?
>
>
i shall certainly post it here as soon as i get back into the network. So it
is possible the problem could be with my network configuration ..

-- 
David Wafula
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From nico at clickfono.com  Mon Oct 11 13:32:28 2010
From: nico at clickfono.com (Nicolas Brenner)
Date: Mon, 11 Oct 2010 17:32:28 -0300
Subject: [Freeswitch-users] Binding to virtual interface (ipsec vpn)
In-Reply-To: <1F64D322-4010-4C3E-B622-93A6254B6E1B@freeswitch.org>
References: <AANLkTikWrRy41kskcaC-0LLEdbNd9d6hi1W26pw2ORg+@mail.gmail.com>
	<1F64D322-4010-4C3E-B622-93A6254B6E1B@freeswitch.org>
Message-ID: <AANLkTi=u6VWM-ejjjFPPws9-ZMTR+KxdRHLTBJ+5O6Jh@mail.gmail.com>

Here's the output of ifconfig -a:

http://pastebin.freeswitch.org/14199

This is the output of route:

http://pastebin.freeswitch.org/14200


This is the external.xml file under sip_profiles (is the default, only
modified sip-ip and rtp-ip):

http://pastebin.freeswitch.org/14201


This is somegateway.xml file under sip_profiles/external/

http://pastebin.freeswitch.org/14202


Output of sofia status from cli (tried with alias=true and alias=false on
internal profile):

http://pastebin.freeswitch.org/14203


This is the cli output with loglevel all set to 9 and siptrace on for
external profile:

http://pastebin.freeswitch.org/14204


Thanks


On Mon, Oct 11, 2010 at 2:33 PM, Brian West <brian at freeswitch.org> wrote:

> If you set the IP in sip-ip and rtp-ip that is abound to eth0:3 then it
> should bind to that interface/ip and use that... I haven't see anything out
> of the usual when doing this.
>
> Show me your profile config and the output of ifconfig -a
>
> /b
>
> On Oct 11, 2010, at 4:46 AM, Nicolas Brenner wrote:
>
> > Hello everyone,
> >
> > I'm having trouble binding FS to another ip on a virtual interface
> (eth0:3). The virtual interface's ip address is the gateway to an ipsec vpn,
> and the only way to access this voip provider's sip server, is through the
> vpn.
> >
> > I tried setting up an additional sip profile, with rtp-ip and sip-ip as
> the virtual interface's ip, but when I do that, no packets are sent at all
> when trying to make a call with originate through the cli (according to
> tcpdump). I tried also setting ext-rtp-ip and ext-sip-ip to the same virtual
> interface's ip, but nothing happened, still no packets being sent (according
> to tcpdump) when trying to make a call. Finally I tried only setting
> ext-rtp-ip and ext-sip-ip to the virtual interface's ip, and leave rtp-ip
> and sip-ip to $${local_ip_v4}, then tcpdump started seeing some traffic
> (when trying to originate a call) which I was able to capture. When I loaded
> the trace on wireshark though, I noticed the source ip being used to send
> all the (sip) packets, is the one from eth0 (public ip), and not eth0:3's ip
> (private ip for vpn).
> >
> > Has anyone had any similiar experiences? how were you able to solve it?
> >
> > Thanks for your help!
> >
> >
> > Nico
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Oct 11 13:53:02 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Oct 2010 13:53:02 -0700
Subject: [Freeswitch-users] Call For Assistance: Git Gurus
Message-ID: <AANLkTi=AJdkKBwSLHmr1wjFjLZom_jsKvbHOwYQXreEG@mail.gmail.com>

If you are a Git guru and are in a position to help us with a few specific
mini-projects we would be most appreciative. Right now the current need is
to create an email notification system when new commits are made to git
HEAD.

If you are experienced in this area and can help us out then please email
Ray and me offlist.
Thanks,
Michael

msc at freeswitch.org
intralanman at freeswitch.org
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From anthony.minessale at gmail.com  Mon Oct 11 13:57:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Oct 2010 15:57:45 -0500
Subject: [Freeswitch-users] Binding to virtual interface (ipsec vpn)
In-Reply-To: <AANLkTikWrRy41kskcaC-0LLEdbNd9d6hi1W26pw2ORg+@mail.gmail.com>
References: <AANLkTikWrRy41kskcaC-0LLEdbNd9d6hi1W26pw2ORg+@mail.gmail.com>
Message-ID: <AANLkTi=ABmOWZCpsNt=kYTRqnEwfpvnceSk=UQtz7FM0@mail.gmail.com>

100% you should only be setting sip-ip sip-port and rtp-ip
you don't need the ext-* family of params at all.

did you check netstat -na | grep <your ip> to see that it was bound?

I think it's a network interface configuration problem more than anything else.
It could also be iptables which you can prove by disabling it to see
any improvement and then craft appropriate rules.


On Mon, Oct 11, 2010 at 4:46 AM, Nicolas Brenner <nico at clickfono.com> wrote:
> Hello everyone,
> I'm having trouble binding FS to another ip on a virtual interface (eth0:3).
> The virtual interface's ip address is the gateway to an ipsec vpn, and the
> only way to access this voip provider's sip server, is through the vpn.
> I tried setting up an additional sip profile, with rtp-ip and sip-ip as the
> virtual interface's ip, but when I do that, no packets are sent at all when
> trying to make a call with originate through the cli (according to tcpdump).
> I tried also setting ext-rtp-ip and?ext-sip-ip to the same virtual
> interface's ip, but nothing happened, still no packets being sent (according
> to tcpdump) when trying to make a call. Finally I tried only
> setting?ext-rtp-ip and?ext-sip-ip to the virtual interface's ip, and leave
> rtp-ip and sip-ip to?$${local_ip_v4}, then tcpdump started seeing some
> traffic (when trying to originate a call) which I was able to capture. When
> I loaded the trace on wireshark though, I noticed the source ip being used
> to send all the (sip) packets, is the one from eth0 (public ip), and not
> eth0:3's ip (private ip for vpn).
> Has anyone had any similiar experiences? how were you able to solve it?
> Thanks for your help!
>
> Nico
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From anthony.minessale at gmail.com  Mon Oct 11 14:00:42 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Oct 2010 16:00:42 -0500
Subject: [Freeswitch-users] tone_detect vs. ignore_early_media
	ANDmod_fifo
In-Reply-To: <D33D5C7C9BAE42CA96354582D25F8B21@mbnet.local>
References: <429823D0454A4EB4AE179DF13803F9B9@mbnet.local>
	<AANLkTim1B_oxbCAXNPTHNoZxh6vUfbUiWsyHzxG=xwif@mail.gmail.com>
	<DAFF983A4D6A43A186D8662FE5876375@mbnet.local>
	<515E6743DC69484A8E3514E807B24353@mbnet.local>
	<D33D5C7C9BAE42CA96354582D25F8B21@mbnet.local>
Message-ID: <AANLkTimea0_Gs6jcc2ic=pYFaxFhKPAErgPnRbQnZva0@mail.gmail.com>

This was done to stop recordings and other things on callers waiting in a queue.
We may have to change it to be configurable.


On Mon, Oct 11, 2010 at 6:51 AM, Jeroen C. van Gelderen
<slim at thegreek.com> wrote:
>
>
> I have an inkling that the following commit made between 1.0.2 and 1.0.3
> might have something to do with this:
>
>
>
> ?? * mod_fifo: pause media bugs while not in a bridge (r:11466,11490)
>
>
>
> http://lists.freeswitch.org/pipermail/freeswitch-svn/2009-January/009566.html
>
> http://lists.freeswitch.org/pipermail/freeswitch-svn/2009-January/009589.html
>
>
>
> Can anyone comment on the how and why? Is there any way to reconcile this
> with use of tone_detect?
>
> Cheers,
> -Slim
> --
> Jeroen C. "Slim" van Gelderen
> Olympic Sports Data Services
> Email: slim at thegreek.com
> Phone: +1 876 953 6182 x128
>
> ________________________________
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeroen
> C. van Gelderen
> Sent: Monday, October 11, 2010 05:31
> To: 'FreeSWITCH Users Help'
> Subject: Re: [Freeswitch-users] tone_detect vs. ignore_early_media
> ANDmod_fifo
>
>
>
>
>
> Hmm? the plot thickens?
>
>
>
> When I bridge my FXO port to the SIP-GSM gateway directly (i.e. without
> using mod_fifo) I don?t seem to need monitor_early_media_fail. Using
> tone_detect on the A leg works fine when ?ignore_early_media=true? is used
> on the B leg:
>
>
>
> The following dialplan excerpt WORKS (i.e. FXO hang-ups are detected at all
> stages by tone_detect):
>
>
>
> ??? <extension name="pan_125">
>
> ????? <condition field="${freetdm_span_name}"
> expression="^span_fxo_helpdesk$">
>
> ??????? <!-- Detect busy tone from Panasonic -->
>
> ??????? <action application="tone_detect" data="busy-pan 350,440 r 0 hangup
> undef 2"/>
>
>
>
> ??????? <action application="answer"/>
>
>
>
> ??????? <action application="set" data="continue_on_fail=true"/>
>
> ??????? <action application="set" data="instant_ringback=true"/>
>
> ??????? <action application="set" data="ringback=%(1000,1500,440.0,480.0)"/>
>
>
>
> ??????? <action application="bridge"
> data="{ignore_early_media=true}sofia/internal/1??????76??@192.168.3.11:5060"/>
>
> ??????? <action application="bridge"
> data="{ignore_early_media=true}sofia/internal/1??????77??@192.168.3.11:5060"/>
>
> ????? </condition>
>
> ??? </extension>
>
>
>
> The problem seems to occur only when mod_fifo is added to the mix:
>
>
>
> ??? <extension name="test2">
>
> ????? <condition field="${freetdm_span_name}" expression="^span_fxo_test$">
>
> ??????? <!-- Detect busy tone from Panasonic -->
>
> ??????? <action application="tone_detect" data="busy-pan 350,440 r 0 hangup
> undef 2"/>
>
>
>
> ??????? <action application="answer"/>
>
>
>
> ??????? <action application="playback" data="thankyou.wav"/>
>
> ??????? <action application="set" data="fifo_music=$${hold_music}"/>
>
> ??????? <action application="fifo" data="fifo_helpdesk in"/>
>
> ????? </condition>
>
> ??? </extension>
>
>
>
> In all cases:
>
> -????????? tone_detect works BEFORE the call is handed to mod_fifo (i.e.
> during playback)
>
> -????????? tone_detect works AFTER the call is established by mod_fifo and
> audio is being exchanged between A and B leg.
>
>
>
> IF the A leg (with tone_detect enabled) is handed to mod_fifo AND there are
> no agents in the fifo THEN tone_detect does not work. Mod_fifo simply plays
> MOH to the A leg perpetually.
>
>
>
> IF the A leg (with tone_detect enabled) is handed to mod_fifo AND mod_fifo
> places an outbound call with ?ignore_early_media=true? THEN tone_detect does
> not work on the A leg during the early media phase on leg B.
>
>
>
> IF the A leg (with tone_detect enabled) is handed to mod_fifo AND mod_fifo
> places an outbound call with ?ignore_early_media=false? THEN tone_detect
> does work. I guess this is because audio is being exchanged between A and B
> legs.
>
>
>
> Any idea what would cause the tone_detect to be ?suspended? when mod_fifo is
> in the mix?
>
> Cheers,
> -Slim
> --
> Jeroen C. "Slim" van Gelderen
>
> ________________________________
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeroen
> C. van Gelderen
> Sent: Monday, October 11, 2010 03:13
> To: 'FreeSWITCH Users Help'
> Subject: Re: [Freeswitch-users] tone_detect vs. ignore_early_media (per
> leg?)
>
>
>
> Thank you for the quick response.
>
>
>
> It looks like monitor_early_media_fail should do what I need (thanks for the
> suggestion!) but I can?t seem to make it work. See below for my uneducated
> best guess why.
>
>
>
> In my case the failure conditions from the GSM side are handled out-of-band
> by SIP. That leaves only one failure condition I need to listen for in early
> media (the Panasonic far-end hang up on FXO) which can successfully be
> detected with:
>
>
>
> ? <action application="tone_detect" data="busy-pan 350,440 r 0 hangup>
>
>
>
> This results in the following log entries:
>
>
>
> [DEBUG] switch_ivr_async.c:2072 Adding tone spec 350,440 index 0 hits 1
>
> [DEBUG] switch_core_media_bug.c:360 Attaching BUG to FreeTDM/4:1/
>
>
>
> Translating tone_detect to monitor_early_media_fail I tried the following:
>
>
>
> ? <action application="set" data="result=${fifo_member(add fifo_helpdesk
> {fifo_member_wait=nowait,ignore_early_media=true,monitor_early_media_fail=busy-pan:1:350+440}sofia/internal/${caller_id_number}@192.168.3.11:5060)}"/>
>
>
>
> This results in the following relevant log entries:
>
>
>
> [DEBUG] switch_channel.c:2444 (sofia/internal/1876??????@192.168.3.11:5060)
> Callstate Change RINGING -> EARLY
>
> [DEBUG] switch_ivr_async.c:2072 Adding tone spec 350,440 index 0 hits 1
>
> [DEBUG] switch_core_media_bug.c:360 Attaching BUG to
> sofia/internal/1??????7693 at 192.168.3.11:5060
>
>
>
> So Freeswitch is listening for the right tones (tone_spec present and
> identical to the one used in tone_detect approach) but it isn?t detecting
> them. The obvious difference is in the BUG attachment.
>
>
>
> Is it possible that BUG isn?t listening to the right (A) leg in the case of
> monitor_early_media_fail? Or is this too easy? J
>
> Cheers,
> -Slim
> --
> Jeroen C. "Slim" van Gelderen
>
> ________________________________
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
> Collins
> Sent: Sunday, October 10, 2010 19:01
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] tone_detect vs. ignore_early_media (per
> leg?)
>
>
>
> Check out monitor_early_media_fail:
> http://wiki.freeswitch.org/wiki/Channel_Variables#monitor_early_media_fail
>
> It is a hybrid of ignoring early media and listening to early media for
> various conditions. In your case you'll need to play around with it. In your
> case you need to figure out which early media scenarios count as a "fail"
> and will cause processing to move on as if the call really did fail.
>
> Roll up your sleeves, you have some work to do. :)
>
> -MC
>
> On Sat, Oct 9, 2010 at 9:24 PM, Jeroen C. van Gelderen <slim at thegreek.com>
> wrote:
>
> Hi Guys,
>
> I have a legacy Panasonic PBX which does not support CPC/Disconnect
> Supervision. Calls from this PBX are sent to Freeswitch by way of
> DAHDI/FreeTDM FXO ports (on a Xorcom Astribank). I use tone_detect to deal
> with far-end hang-ups. This works fine:
>
> <!-- Detect busy tone from Panasonic -->
> <action application="tone_detect" data="busy 350,440 r 0 hangup 4"/>
>
> Some calls from the Panasonic PBX are put in a FIFO and from there they are
> sent to a SIP-to-GSM bridge (GoIP) to be terminated on one of 3 GSM phones
> in a round-robin fashion.
>
> The GSM bridge requires me to use "ignore_early_media=true" otherwise the
> caller will receive messages like "the number you are calling does not
> answer". When I set "ignore_early_media=true" the FIFO correctly keeps
> hunting for a GSM phone that is actually answered and will ignore phones
> that are busy, no-answer or turned off. This too works fine.
>
> The problem occurs when the two are combined as follows:
>
> Panasonic PBX --FXO--> Freeswitch --SIP--> GSM-bridge
>
> If I enable ignore_early_media then tone_detect doesn't work UNTIL one of
> the GSMs is answered. This is a problem when none of the GSMs are answered
> and the Panasonic PBX hangs up. The FXO port will stay "up" and the FIFO
> will keep hunting until it times out.
>
> If I don't use "ignore_early_media=true" then tone_detect works fine but
> then every telco message gets mistaken for an answered call and the hunting
> stops early.
>
> I tried changing this example line from my fifo.conf:
>
> <member timeout="30" simo="1" lag="5">
> {member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7701 at 192.1
> 68.3.11:5060
> </member>
>
> to
>
> <member timeout="30" simo="1" lag="5">
> {member_wait=nowait}[ignore_early_media=true]sofia/internal/1??????7701 at 192.
> 168.3.11:5060
> </member>
>
> in a vain attempt to ignore early media on the SIP leg only. This doesn't
> seem to do anything however.
>
> Can anyone clue me in on what I'm missing? I've snipped the relevant
> configuration bits below. I have the feeling I'm missing something obvious.
>
> Cheers,
> -Slim
>
> ----8<----8<----8<----8<----8<----
>
> <configuration name="fifo.conf" description="FIFO Configuration">
> ?<settings>
> ? ?<param name="delete-all-outbound-member-on-startup" value="false"/>
> ?</settings>
> ?<fifos>
> ? ?<fifo name="cool_fifo@$${domain}" importance="0">
>
> ? ? ?<member timeout="30" simo="1" lag="5">
> {member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7701 at 192.1
> 68.3.11:5060</member>
>
> ? ? ?<member timeout="30" simo="1" lag="5">
> {member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7693 at 192.1
> 68.3.11:5060</member>
>
> ? ? ?<member timeout="30" simo="1" lag="5">
> {member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7698 at 192.1
> 68.3.11:5060</member>
>
> ? ?</fifo>
> ?</fifos>
> </configuration>
>
> ----8<----8<----8<----8<----8<----
>
> <include>
> ?<context name="pbx_test">
>
> ? ?<extension name="test1">
> ? ? ?<condition field="${freetdm_span_name}" expression="^span_fxo_test$">
> ? ? ? ?<action application="answer"/>
>
> ? ? ? ?<!-- Detect busy tone from Panasonic -->
> ? ? ? ?<action application="tone_detect" data="busy 350,440 r 0 hangup 4"/>
>
> ? ? ? ?<action application="playback"
> data="ivr/ivr-hold_connect_call.wav"/>
>
> ? ? ? ?<action application="fifo" data="cool_fifo@$${domain} in"/>
>
> ? ? ?</condition>
> ? ?</extension>
> ?</context>
> </include>
>
> ----8<----8<----8<----8<----8<----
>
> Cheers,
> -Slim
> --
> Jeroen C. "Slim" van Gelderen
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From msc at freeswitch.org  Mon Oct 11 14:07:34 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Oct 2010 14:07:34 -0700
Subject: [Freeswitch-users] origination_caller_is_number
In-Reply-To: <0FF607C13B7F7A42B5B80DA9EB204C9109D44BD2D8@srv-ex01-dal.intervoice.int>
References: <ActnJ/ehcefI4Kb/TJK/dKhS3IErTA==>
	<0FF607C13B7F7A42B5B80DA9EB204C9109D44BD2D8@srv-ex01-dal.intervoice.int>
Message-ID: <AANLkTimJ23XXLWK6qvG2d5T5U5rGvdxBqa4b_XDodqD0@mail.gmail.com>

Can you pastebin a redacted copy of your gateway config?
-MC

On Fri, Oct 8, 2010 at 1:32 PM, Esser, Holger <Holger.Esser at convergys.com>wrote:

>  Hi,
>
>
>
> Whenever I set the origination_caller_is_number in my dial request like
> this for a local or external context,
>
>
>
> perl http.pl originate
> {origination_caller_id_number=5xxxxxxxxxsofia/internal/1 at 148.181.145.148 5000
> the caller id is set.
>
>
>
> Whenever I use the gateway, it is no longer working and the username in the
> gateway becomes the ani number in the SIP trace.
>
> perl http.pl originate
> {origination_caller_id_number=972xxxxxxx}sofia/gateway/teliax/4695xxxxxxx
> 5000
>
>
>
> Any ideas?
>
>
>
> Thx
>
>
>
>
>
> [image: cid:image001.gif at 01CA361E.BF939740]
>
> Holger Esser
>
> Staff Engineer, Continuing Engineering / RTM
> Convergys Corporation
>
> office 972-454-8167
>
> www.convergys.com | www.intervoice.com
>
>
>
>
>
> ------------------------------
> This e-mail transmission may contain information that is proprietary,
> privileged and/or confidential and is intended exclusively for the person(s)
> to whom it is addressed. Any use, copying, retention or disclosure by any
> person other than the intended recipient or the intended recipient's
> designees is strictly prohibited. If you are the intended recipient, you
> must treat the information in confidence and in accordance with all laws
> related to the privacy and confidentiality of such information. If you are
> not the intended recipient or their designee, please notify the sender
> immediately by return e-mail and delete all copies of this email, including
> all attachments.
>
> Intervoice, Inc. 17811 Waterview Parkway Dallas, TX 75252 USA
>
> Intervoice Limited, Registered in England and Wales with number 2601740, 50
> Park Road, Gatley, Cheshire, SK8 4HZ. VAT Number: 560421375
>
> Intervoice GmbH, Hagenauer Stra?e 55, 65203 Wiesbaden, Sitz der
> Gesellschaft: Wiesbaden, Handelsregister: HRB 8186 (Amtsgericht Wiesbaden),
> Gesch?ftsf?hrer: Wayne Barclay, Steffen Selbmann
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From neilp at cs.stanford.edu  Mon Oct 11 14:15:33 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Mon, 11 Oct 2010 14:15:33 -0700
Subject: [Freeswitch-users] outbound calling error: multiple PRIs with
 Sangoma + FS
In-Reply-To: <625156AD55324D3DB824F758A190C370@cune.pri>
References: <AANLkTimn7P64Exf9JTwdLZB8Vam8wBLuvrLS5MY+xAwT@mail.gmail.com>
	<AANLkTin01O6WPqXfXN+aiaPcOd+Ww1HiFwzcbUEeMz2Z@mail.gmail.com>
	<AANLkTim-UWMzVNkdnTyZnESkKTou=VqN0Y3_UU=B80oX@mail.gmail.com>
	<625156AD55324D3DB824F758A190C370@cune.pri>
Message-ID: <AANLkTik4A2jNNBAVKo6Ocna1uChLw7qnoeM9yF_P8GkL@mail.gmail.com>

I've already followed these steps... Wanpipe is compiled for FreeTDM and
installed, and I have configured FS as above, based on Sangoma's
instructions.

-Neil

On Sat, Oct 9, 2010 at 7:01 AM, Russell Mosemann
<Russell.Mosemann at cune.org>wrote:

> Neil Patel typed:
> >        <param name="q921loglevel" value="alert"/>
> >        <param name="q931loglevel" value="alert"/>
> >        <param name="mode" value="user"/>
> >        <param name="dialect" value="5ess"/>
> ...
> > 2010-10-09 18:28:58.487311 [WARNING] ftmod_sangoma_isdn_cfg.c:262
> Ignoring
> > unknown parameter q921loglevel
> > 2010-10-09 18:28:58.487369 [WARNING] ftmod_sangoma_isdn_cfg.c:262
> Ignoring
> > unknown parameter q931loglevel
> > 2010-10-09 18:28:58.487425 [WARNING] ftmod_sangoma_isdn_cfg.c:262
> Ignoring
> > unknown parameter mode
> > 2010-10-09 18:28:58.487480 [WARNING] ftmod_sangoma_isdn_cfg.c:262
> Ignoring
> > unknown parameter dialect
>
> http://wiki.sangoma.com/wanpipe-api-freetdm
> http://wiki.sangoma.com/Freeswitch-FreeTDM-Sangoma-ISDN-Library
>
>
> mose
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Russell.Mosemann at cune.org  Mon Oct 11 14:33:54 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Mon, 11 Oct 2010 21:33:54 -0000
Subject: [Freeswitch-users] Binding to virtual interface (ipsec vpn)
In-Reply-To: <AANLkTi=u6VWM-ejjjFPPws9-ZMTR+KxdRHLTBJ+5O6Jh@mail.gmail.com>
Message-ID: <20101011213354.C82E62C36FD@cuneorg-email.cune.pri>

Another thing to check is that there is only one IP address assigned to
the interface. A problem with using the "right" IP address can occur if
there is more than one IP address assigned to the same interface. Without
routing rules, the operating system chooses the IP address from the
interface that it believes is the best one for delivering the traffic,
and sometimes it is not the address you want to use.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From nico at clickfono.com  Mon Oct 11 14:41:25 2010
From: nico at clickfono.com (Nicolas Brenner)
Date: Mon, 11 Oct 2010 18:41:25 -0300
Subject: [Freeswitch-users] Binding to virtual interface (ipsec vpn)
In-Reply-To: <AANLkTi=ABmOWZCpsNt=kYTRqnEwfpvnceSk=UQtz7FM0@mail.gmail.com>
References: <AANLkTikWrRy41kskcaC-0LLEdbNd9d6hi1W26pw2ORg+@mail.gmail.com>
	<AANLkTi=ABmOWZCpsNt=kYTRqnEwfpvnceSk=UQtz7FM0@mail.gmail.com>
Message-ID: <AANLkTik7FYOsGK-j76adCxphHRw4Bs1SRUYYUFEwkzKg@mail.gmail.com>

On Mon, Oct 11, 2010 at 5:57 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> 100% you should only be setting sip-ip sip-port and rtp-ip
> you don't need the ext-* family of params at all.
>
> did you check netstat -na | grep <your ip> to see that it was bound?
>
>
The external profile is bound to port 5080 on the specified ip address,
maybe the title of my email is somewhat misleading, but the problem is that
even though it is bound to that address, it is not sending out any traffic.



> I think it's a network interface configuration problem more than anything
> else.
>


The network interface is actually working fine, I can ping the gateway and
telnet to the port where it is configured to listen. Basically, I can
communicate with it manually, but not with FS.



> It could also be iptables which you can prove by disabling it to see
> any improvement and then craft appropriate rules.
>
>
iptables is actually disabled for these tests, yet this is all I get from
tcpdump while trying to originate a call:

tcpdump -nq -s 0 -A -vvv host 200.13.15.220
tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 65535
bytes

0 packets captured
0 packets received by filter
0 packets dropped by kernel




>
> On Mon, Oct 11, 2010 at 4:46 AM, Nicolas Brenner <nico at clickfono.com>
> wrote:
> > Hello everyone,
> > I'm having trouble binding FS to another ip on a virtual interface
> (eth0:3).
> > The virtual interface's ip address is the gateway to an ipsec vpn, and
> the
> > only way to access this voip provider's sip server, is through the vpn.
> > I tried setting up an additional sip profile, with rtp-ip and sip-ip as
> the
> > virtual interface's ip, but when I do that, no packets are sent at all
> when
> > trying to make a call with originate through the cli (according to
> tcpdump).
> > I tried also setting ext-rtp-ip and ext-sip-ip to the same virtual
> > interface's ip, but nothing happened, still no packets being sent
> (according
> > to tcpdump) when trying to make a call. Finally I tried only
> > setting ext-rtp-ip and ext-sip-ip to the virtual interface's ip, and
> leave
> > rtp-ip and sip-ip to $${local_ip_v4}, then tcpdump started seeing some
> > traffic (when trying to originate a call) which I was able to capture.
> When
> > I loaded the trace on wireshark though, I noticed the source ip being
> used
> > to send all the (sip) packets, is the one from eth0 (public ip), and not
> > eth0:3's ip (private ip for vpn).
> > Has anyone had any similiar experiences? how were you able to solve it?
> > Thanks for your help!
> >
> > Nico
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Russell.Mosemann at cune.org  Mon Oct 11 14:42:25 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Mon, 11 Oct 2010 21:42:25 -0000
Subject: [Freeswitch-users] outbound calling error: multiple PRIs with
	Sangoma + FS
In-Reply-To: <AANLkTik4A2jNNBAVKo6Ocna1uChLw7qnoeM9yF_P8GkL@mail.gmail.com>
Message-ID: <20101011214225.37B2F35CFA4@cuneorg-email.cune.pri>

Neil Patel <neilp at cs.stanford.edu> said:

> I've already followed these steps... Wanpipe is compiled for FreeTDM and
> installed, and I have configured FS as above, based on Sangoma's
> instructions.


If that were true, we wouldn't see q921loglevel as a parameter below, and
FS wouldn't be complaining about it. Maybe another look at the URLs I
included would be helpful to see the valid parameters you can use for
configuration.


> On Sat, Oct 9, 2010 at 7:01 AM, Russell Mosemann
> <Russell.Mosemann at cune.org>wrote:
> 
> > Neil Patel typed:
> > >        <param name="q921loglevel" value="alert"/>

> > > 2010-10-09 18:28:58.487311 [WARNING] ftmod_sangoma_isdn_cfg.c:262
> > > Ignoring unknown parameter q921loglevel

> > http://wiki.sangoma.com/wanpipe-api-freetdm
> > http://wiki.sangoma.com/Freeswitch-FreeTDM-Sangoma-ISDN-Library


-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From nico at clickfono.com  Mon Oct 11 14:44:28 2010
From: nico at clickfono.com (Nicolas Brenner)
Date: Mon, 11 Oct 2010 18:44:28 -0300
Subject: [Freeswitch-users] Binding to virtual interface (ipsec vpn)
In-Reply-To: <20101011213354.C82E62C36FD@cuneorg-email.cune.pri>
References: <AANLkTi=u6VWM-ejjjFPPws9-ZMTR+KxdRHLTBJ+5O6Jh@mail.gmail.com>
	<20101011213354.C82E62C36FD@cuneorg-email.cune.pri>
Message-ID: <AANLkTinFuqDHbU1yLhtafSv4P6XJGG=eRpiHuR6XLEDB@mail.gmail.com>

The system routes are well configured, I can ping and telnet the gateway
from bash, all traffic to that gateway's ip is routed through the needed
interface. Now, are there other routes I should be addressing in FS config
files?


On Mon, Oct 11, 2010 at 6:33 PM, <Russell.Mosemann at cune.org> wrote:

> Another thing to check is that there is only one IP address assigned to
> the interface. A problem with using the "right" IP address can occur if
> there is more than one IP address assigned to the same interface. Without
> routing rules, the operating system chooses the IP address from the
> interface that it believes is the best one for delivering the traffic,
> and sometimes it is not the address you want to use.
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Russell.Mosemann at cune.org  Mon Oct 11 14:49:56 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Mon, 11 Oct 2010 21:49:56 -0000
Subject: [Freeswitch-users] Binding to virtual interface (ipsec vpn)
In-Reply-To: <AANLkTik7FYOsGK-j76adCxphHRw4Bs1SRUYYUFEwkzKg@mail.gmail.com>
Message-ID: <20101011214957.0BF7C39AC9A@cuneorg-email.cune.pri>

Nicolas Brenner <nico at clickfono.com> said:

> tcpdump -nq -s 0 -A -vvv host 200.13.15.220
> tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 65535
> bytes

Check if any udp traffic is going out from FS. You can restrict it by
specifying a port, in case you have DNS or DHCP on the same machine.

tcpdump -nq -s 0 -A -vvv udp

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From msc at freeswitch.org  Mon Oct 11 14:50:28 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Oct 2010 14:50:28 -0700
Subject: [Freeswitch-users] Binding to virtual interface (ipsec vpn)
In-Reply-To: <AANLkTinFuqDHbU1yLhtafSv4P6XJGG=eRpiHuR6XLEDB@mail.gmail.com>
References: <AANLkTi=u6VWM-ejjjFPPws9-ZMTR+KxdRHLTBJ+5O6Jh@mail.gmail.com>
	<20101011213354.C82E62C36FD@cuneorg-email.cune.pri>
	<AANLkTinFuqDHbU1yLhtafSv4P6XJGG=eRpiHuR6XLEDB@mail.gmail.com>
Message-ID: <AANLkTinseQ7UEp2u+QRTxuGV6fGqwwaii1-ppT_pB__P@mail.gmail.com>

When you do "sofia status" do you see the correct IP address for the profile
in question?
Also, when you make an outbound call can you see where (if?) the packets are
going? Perhaps to another interface?

-MC

On Mon, Oct 11, 2010 at 2:44 PM, Nicolas Brenner <nico at clickfono.com> wrote:

> The system routes are well configured, I can ping and telnet the gateway
> from bash, all traffic to that gateway's ip is routed through the needed
> interface. Now, are there other routes I should be addressing in FS config
> files?
>
>
> On Mon, Oct 11, 2010 at 6:33 PM, <Russell.Mosemann at cune.org> wrote:
>
>> Another thing to check is that there is only one IP address assigned to
>> the interface. A problem with using the "right" IP address can occur if
>> there is more than one IP address assigned to the same interface. Without
>> routing rules, the operating system chooses the IP address from the
>> interface that it believes is the best one for delivering the traffic,
>> and sometimes it is not the address you want to use.
>>
>> --
>> Russell Mosemann
>>
>>
>>
>> ________________________________________________________
>> Concordia University, Nebraska
>> See http://www.cune.edu/ for the latest news and events!
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mario_fs at mgtech.com  Mon Oct 11 15:17:33 2010
From: mario_fs at mgtech.com (Mario)
Date: Mon, 11 Oct 2010 15:17:33 -0700
Subject: [Freeswitch-users] Newbie (sort of) questions
In-Reply-To: <AANLkTim+zr9DWNQ9QCVccMETYb8Y=OT89t8QhjyfWVv4@mail.gmail.com>
References: <DFB4B88E-6834-4D22-9DEE-616AB50F0D91@stillnewt.org>
	<AANLkTim+zr9DWNQ9QCVccMETYb8Y=OT89t8QhjyfWVv4@mail.gmail.com>
Message-ID: <4CB38CFD.8040809@mgtech.com>

Hope this helps: I am replacing an SPA9000 (asterisk) with FS, testing
on Linux then moving to a mac mini. I tried this 8 months ago and used
some of the GUI options. My two cents: stick with XML parms. Things go
wrong and a GUI gets in the way, it "hides from view" valuable info. I
was able to get FS up and running the first try on Linux. osX 10.6.4
took some work (search for my userid in the list from the last 2 weeks)
but eventually worked. Although FS is done I am having one last tough
problem no one here (or on IRC) attempted to help with and I can't
imagine working on it without having done the XML stuff myself. To
answer your question though, this is what I did and recommend:

1. If you haven't already buy the book and study it! You definitely
won't need/want a GUI after that.
2. Search the wiki, it has some very good configs to get started, use
pieces from multiple samples. Google search help with the wiki too.
3. Search Google to find the pages on this mailing list easily.
4. As a last resort, post on this group.

On 10/11/2010 12:10 PM, Nyamul Hassan wrote:
> There is a new project, which looks quite promising:
> 
> www.2600hz.org
> 
> Regards
> HASSAN
> 
> 
> On 2010-10-12, Martin Joseph <astmac at stillnewt.org> wrote:
>> Hi Again FreeSWITCHers,
>>
>> I am building a new setup for my "new" house (I have been remodeling for 5
>> years).
>>
>> I have build FS via GIT on my "new" xServe hardware (dual Xeon 2Ghz) which I
>> was able to do pretty easily (Hurray!).  In the past I have struggled with
>> updating my older FS install on my anemic g3/500 ibook install which is
>> still running by the way.
>>
>> So,  now I am thinking about configuring my new installation, and I being no
>> expert am not looking forward to figuring this all out again. My
>> installation is overall pretty simple with a gateway to POTS (audiocodes
>> MP114) and several Voip providors, one of whom is hopefully still in
>> business (heh).
>>
>> Anyhow,  my first request is for GUI based FS configuration systems?  These
>> could run on my OSX box (via X11 or Cocoa), or be some kind of web based
>> thing.  It doesn't need to be super customizable, but not having to
>> find/edit the XML files would be lovely. I am a mac person after all.
>>
>> If there are no recommended GUI's available, then pointing me to the best
>> sample pages for configuration would also be greatly appreciated...
>>
>> Thanks again for the great software and all of your efforts!
>> Marty
>>
>> PS looking forward to killing me last Asterisk install...
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 

-- 
*Mario*


From nico at clickfono.com  Mon Oct 11 15:21:46 2010
From: nico at clickfono.com (Nicolas Brenner)
Date: Mon, 11 Oct 2010 19:21:46 -0300
Subject: [Freeswitch-users] Binding to virtual interface (ipsec vpn)
In-Reply-To: <AANLkTinseQ7UEp2u+QRTxuGV6fGqwwaii1-ppT_pB__P@mail.gmail.com>
References: <AANLkTi=u6VWM-ejjjFPPws9-ZMTR+KxdRHLTBJ+5O6Jh@mail.gmail.com>
	<20101011213354.C82E62C36FD@cuneorg-email.cune.pri>
	<AANLkTinFuqDHbU1yLhtafSv4P6XJGG=eRpiHuR6XLEDB@mail.gmail.com>
	<AANLkTinseQ7UEp2u+QRTxuGV6fGqwwaii1-ppT_pB__P@mail.gmail.com>
Message-ID: <AANLkTimqaPs0hPaeG_QtPoX1f06ETRM4Jj4X3mOaMSxk@mail.gmail.com>

Yup, here's the output:

http://pastebin.freeswitch.org/14203


I'm trying to originate a call using the gateway on 200.13.15.220,
originating it from 172.26.6.161.

Seems like no packets are going out at all, at least tcpdump is not getting
any. Oddly, if instead of setting sip-ip and rtp-ip to the ip of the
interface I need, I set ext-sip-ip and ext-rtp-ip to that ip, tcpdump starts
seeing traffic, but the traffic is originated from the wrong ip/interface
(the public one, instead of the one with the ipsec vpn).



On Mon, Oct 11, 2010 at 6:50 PM, Michael Collins <msc at freeswitch.org> wrote:

> When you do "sofia status" do you see the correct IP address for the
> profile in question?
> Also, when you make an outbound call can you see where (if?) the packets
> are going? Perhaps to another interface?
>
> -MC
>
>
> On Mon, Oct 11, 2010 at 2:44 PM, Nicolas Brenner <nico at clickfono.com>wrote:
>
>> The system routes are well configured, I can ping and telnet the gateway
>> from bash, all traffic to that gateway's ip is routed through the needed
>> interface. Now, are there other routes I should be addressing in FS config
>> files?
>>
>>
>> On Mon, Oct 11, 2010 at 6:33 PM, <Russell.Mosemann at cune.org> wrote:
>>
>>> Another thing to check is that there is only one IP address assigned to
>>> the interface. A problem with using the "right" IP address can occur if
>>> there is more than one IP address assigned to the same interface. Without
>>> routing rules, the operating system chooses the IP address from the
>>> interface that it believes is the best one for delivering the traffic,
>>> and sometimes it is not the address you want to use.
>>>
>>> --
>>> Russell Mosemann
>>>
>>>
>>>
>>> ________________________________________________________
>>> Concordia University, Nebraska
>>> See http://www.cune.edu/ for the latest news and events!
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fs-list at communicatefreely.net  Mon Oct 11 15:37:52 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Mon, 11 Oct 2010 18:37:52 -0400
Subject: [Freeswitch-users] Dead lua scripts
In-Reply-To: <F5C4BC8D-E276-4125-AAB3-655715875EFB@freeswitch.org>
References: <4CAE15F7.9090807@communicatefreely.net>	<AANLkTinKznsWB6Zzf3cexnqHjesgQ_gsK++KkKD_h+U0@mail.gmail.com>	<4CAE7EB0.2070600@communicatefreely.net>
	<F5C4BC8D-E276-4125-AAB3-655715875EFB@freeswitch.org>
Message-ID: <4CB391C0.7040108@communicatefreely.net>

Yes, I'm doing that in all scripts that loop from now on.

What I was trying to find out, was if there is some sort of fs_cli command that I can run to remove
all those calls, other than fsctl restart.

It likely won't be an issue in the future, now that I have the correct control flow, but just in
case - it helps to know these tricks.

I'm hoping to get the book ordered.  I think it will be useful, and I do like books.

-Tim

Brian West wrote:
> Well anthony gave you a hint... wrap your whole loop in a while(session:ready) it will then end the script if the session hangs up.
> 
> /b
> 
> On Oct 7, 2010, at 9:15 PM, Tim St. Pierre wrote:
> 
>> I had to completely restart freeswitch to make these calls go away.  Is there a less disruptive way
>> to terminate the scripts?
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From fs-list at communicatefreely.net  Mon Oct 11 15:40:16 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Mon, 11 Oct 2010 18:40:16 -0400
Subject: [Freeswitch-users] Cannot create any sessions at this time
In-Reply-To: <AANLkTikYZFDnkXGh1OOyupXqD+7H+s6dy7tz68jbHRfL@mail.gmail.com>
References: <4CAF2414.2040900@communicatefreely.net>
	<AANLkTikYZFDnkXGh1OOyupXqD+7H+s6dy7tz68jbHRfL@mail.gmail.com>
Message-ID: <4CB39250.5000605@communicatefreely.net>

Yep, F10 is mapped to that command.  I must have hit it when I was wrapping up.  I have gotten into
the habit of hitting F9 at the end of the day, just to make sure that all the phones are still
alive.  I think I'll be changing that!

Thanks,

-Tim

Anthony Minessale wrote:
> you would need to supply the actual logs.
> but it's possible you inadvertently entered fsctl pause or you had the
> old config where F10 was mapped to that command.
> 
> 
> On Fri, Oct 8, 2010 at 9:00 AM, Tim St. Pierre
> <fs-list at communicatefreely.net> wrote:
>> Hello,
>>
>> At some point during the night, something happened to our freeswitch blocking all outgoing calls.
>> When I looked at the console, I got this message "Cannot create any sessions at this time".  There
>> were no active channels according to show channels, and for whatever reason, I thought to try fsctl
>> hupall and then fsctl resume.  That fixed everything, but I have no idea how it got to that state.
>> Everything was working fine when I went to bed yesterday.
>>
>> I couldn't find anything on the wiki about the error message or the resume command.
>>
>> What does this error message mean, (what can put it into this condition), and what did the resume
>> command do that made it okay again?
>>
>> Just so I can avoid this in the future when we move our 300 office users over to this box!
>>
>> Thanks!
>>
>> -Tim
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 



From msc at freeswitch.org  Mon Oct 11 15:47:55 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Oct 2010 15:47:55 -0700
Subject: [Freeswitch-users] Dead lua scripts
In-Reply-To: <4CB391C0.7040108@communicatefreely.net>
References: <4CAE15F7.9090807@communicatefreely.net>
	<AANLkTinKznsWB6Zzf3cexnqHjesgQ_gsK++KkKD_h+U0@mail.gmail.com>
	<4CAE7EB0.2070600@communicatefreely.net>
	<F5C4BC8D-E276-4125-AAB3-655715875EFB@freeswitch.org>
	<4CB391C0.7040108@communicatefreely.net>
Message-ID: <AANLkTi=b2MDj7uaXOphUqDmXozZFQbs+6c99zywoUK8h@mail.gmail.com>

Well you have a few choices:

uuid_kill <uuid> (if you know the uuid)
hupall <reason> (if you don't mind dropping ALL calls on your system)

I'd say hupall is a good choice. Now, if you want to get really fancy and
you know that your Lua script always sets a particular channel variable then
you can do something like this:

hupall matching <chan_var_name> <value>

Then it will selectively hangup any channel with a chan var matching the
value you specify.
-MC

On Mon, Oct 11, 2010 at 3:37 PM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

> Yes, I'm doing that in all scripts that loop from now on.
>
> What I was trying to find out, was if there is some sort of fs_cli command
> that I can run to remove
> all those calls, other than fsctl restart.
>
> It likely won't be an issue in the future, now that I have the correct
> control flow, but just in
> case - it helps to know these tricks.
>
> I'm hoping to get the book ordered.  I think it will be useful, and I do
> like books.
>
> -Tim
>
> Brian West wrote:
> > Well anthony gave you a hint... wrap your whole loop in a
> while(session:ready) it will then end the script if the session hangs up.
> >
> > /b
> >
> > On Oct 7, 2010, at 9:15 PM, Tim St. Pierre wrote:
> >
> >> I had to completely restart freeswitch to make these calls go away.  Is
> there a less disruptive way
> >> to terminate the scripts?
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Oct 11 15:51:18 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Oct 2010 15:51:18 -0700
Subject: [Freeswitch-users] Routing DIDs to Dynamically Registered SIP
	User
In-Reply-To: <4CAEAD0B.8020704@gmail.com>
References: <4CAEAD0B.8020704@gmail.com>
Message-ID: <AANLkTimv0hvr6BYfKZW1zrdGgHrnMGxpidmpNL-5eu1T@mail.gmail.com>

What is a "dynamically registered SIP user"?
-MC

On Thu, Oct 7, 2010 at 10:32 PM, David Allen <thedjallen at gmail.com> wrote:

>  Hi,
>
> I'm trying to send multiple Direct Indial Numbers down to a dynamically
> registered SIP User. I need to ensure that both the To and Target URI
> contain the direct Indial number. I'm able to modify the SIP TO Header
> of a call that is sent to them like below:
>
> <action application="bridge"
> data="sofia/external/56778977%${domain}^61390009000"/>
>
> which sends the request as:
>
>
> ------------------------------------------------------------------------
>
>    INVITE sip:56778977 at 192.168.22.2:5061 SIP/2.0
>
>    Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j
>
>    Max-Forwards: 69
>
>    From: "0390001000" <sip:0390001000 at 192.168.2.200<sip%3A0390001000 at 192.168.2.200>
> >;tag=K4HHaZ9v1H07Q
>
>    To: <sip:61390009000 at 192.168.2.200 <sip%3A61390009000 at 192.168.2.200>>
>
>  But in order to maintain compatability with a number of PBX's/VoIP
> devices on the market, I need to be able to send the invite to the
> dynamically registered SIP user, however have it set the Target URI and
> To as the same contact number like below:
>
>
> ------------------------------------------------------------------------
>
>    INVITE sip:61390009000 at 192.168.22.2:5061 SIP/2.0
>
>    Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j
>
>    Max-Forwards: 69
>
>    From: "0390001000" <sip:0390001000 at 192.168.2.200<sip%3A0390001000 at 192.168.2.200>
> >;tag=K4HHaZ9v1H07Q
>
>    To: <sip:61390009000 at 192.168.22.2:5061>
>
> Is there a way to do this setting via variables? I can't seem to find
> any details for it.
>
> Thanks
>
> David
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From thedjallen at gmail.com  Mon Oct 11 15:59:16 2010
From: thedjallen at gmail.com (David Allen)
Date: Tue, 12 Oct 2010 09:59:16 +1100
Subject: [Freeswitch-users] Routing DIDs to Dynamically Registered SIP
	User
In-Reply-To: <AANLkTimv0hvr6BYfKZW1zrdGgHrnMGxpidmpNL-5eu1T@mail.gmail.com>
References: <4CAEAD0B.8020704@gmail.com>
	<AANLkTimv0hvr6BYfKZW1zrdGgHrnMGxpidmpNL-5eu1T@mail.gmail.com>
Message-ID: <AANLkTimCNR3KYpb0dxnikbBAFi2uL_24U8u3rq2LBhYO@mail.gmail.com>

Hi Michael,

Its a SIP UA that registers locally to Freeswitch.

Regards,
David

On Tue, Oct 12, 2010 at 9:51 AM, Michael Collins <msc at freeswitch.org> wrote:
> What is a "dynamically registered SIP user"?
> -MC
>
> On Thu, Oct 7, 2010 at 10:32 PM, David Allen <thedjallen at gmail.com> wrote:
>>
>> ?Hi,
>>
>> I'm trying to send multiple Direct Indial Numbers down to a dynamically
>> registered SIP User. I need to ensure that both the To and Target URI
>> contain the direct Indial number. I'm able to modify the SIP TO Header
>> of a call that is sent to them like below:
>>
>> <action application="bridge"
>> data="sofia/external/56778977%${domain}^61390009000"/>
>>
>> which sends the request as:
>>
>>
>> ------------------------------------------------------------------------
>>
>> ? ?INVITE sip:56778977 at 192.168.22.2:5061 SIP/2.0
>>
>> ? ?Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j
>>
>> ? ?Max-Forwards: 69
>>
>> ? ?From: "0390001000" <sip:0390001000 at 192.168.2.200>;tag=K4HHaZ9v1H07Q
>>
>> ? ?To: <sip:61390009000 at 192.168.2.200>
>>
>> ?But in order to maintain compatability with a number of PBX's/VoIP
>> devices on the market, I need to be able to send the invite to the
>> dynamically registered SIP user, however have it set the Target URI and
>> To as the same contact number like below:
>>
>>
>> ------------------------------------------------------------------------
>>
>> ? ?INVITE sip:61390009000 at 192.168.22.2:5061 SIP/2.0
>>
>> ? ?Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j
>>
>> ? ?Max-Forwards: 69
>>
>> ? ?From: "0390001000" <sip:0390001000 at 192.168.2.200>;tag=K4HHaZ9v1H07Q
>>
>> ? ?To: <sip:61390009000 at 192.168.22.2:5061>
>>
>> Is there a way to do this setting via variables? I can't seem to find
>> any details for it.
>>
>> Thanks
>>
>> David
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From peter.schrock at gmail.com  Mon Oct 11 17:22:21 2010
From: peter.schrock at gmail.com (Peter Schrock)
Date: Mon, 11 Oct 2010 17:22:21 -0700
Subject: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4
In-Reply-To: <AANLkTi=YZsrcSWbrnUrZ6+P_fnXpQVAROCNc+9okF0X9@mail.gmail.com>
References: <AANLkTingeYfPT_b6dG3bKQOVjxYsmBTKhDuLw_3BKGac@mail.gmail.com>
	<AANLkTi=YZsrcSWbrnUrZ6+P_fnXpQVAROCNc+9okF0X9@mail.gmail.com>
Message-ID: <AANLkTinFAifa6BPLDqYVEEh1onCobbgJX4DHC_Kzdn_u@mail.gmail.com>

just to make sure what am I comparing event_socket.conf.xml to? I have found
that it is set up with the default settings but am not sure how or to what I
am suppose to fix. I understand that I am comparing it to FS, but I am
wondering if there is some kind of file that I am looking for to compare the
two.

PeterS

On Fri, Oct 8, 2010 at 4:56 AM, Steven Ayre <steveayre at gmail.com> wrote:

> 1. Check FS is running
> 2. Check the port you're connecting to matches the one in
> event_socket.conf.xml
> 3. Check that event_socket.conf.xml binding to the same IP you're
> connecting to (e.g. 0.0.0.0 if you're connecting from a remote
> machine, since the default 127.0.0.1 won't work then)
> 4. Use netstat on the FS server to verify FreeSWITCH is actually
> listening on the port you're trying to connect to (I've had a syntax
> error in the config file make the module fail to load in the past
> which left FS running but with no ESL socket). (If the module fails to
> load there'll also be an error in the log file).
> 5. Check a firewall isn't blocking access to the port
>
> -Steve
>
>
>
> On 7 October 2010 01:23, Peter Schrock <peter.schrock at gmail.com> wrote:
> > Okay, so I managed to get FS working the other day and I even managed to
> > test a call and test my voicemail. All seemed to be working smoothly
> until,
> > because of the rain here, my power went out and I had to reboot my
> computer.
> > I logged in through the terminal, set up FS in background went to fs_cli
> and
> > I get this error message:
> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket Connection
> > Error]
> > I managed to figure out that in the file "fs_cli.c" on line 1206 you find
> > the code for displaying this error message. The problem is that I don't
> know
> > why this error message is occurring. Does anyone have any helpful hints
> as
> > to what I should look at to resolve this problem?
> > I even tried going to the git tree and make current, but that gave me
> > problems that forced me to turn off mod_spandsp and mod_skyopen in the
> > modules.conf, which I had running earlier. Any thoughts?
> > PeterS
> > PS
> > I am not sure if this is of any help, but in addition to the error line
> > above, it also posted this info:
> > Usage: fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x
> > command] [profile]
> >   -?,-h --help                    Usage Information
> >   -H, --host=hostname             Host to connect
> >   -P, --port=port                 Port to connect (1 - 65535)
> >   -u, --user=user at domain          user at domain
> >   -p, --password=password         Password
> >   -x, --execute=command           Execute Command and Exit
> >   -l, --loglevel=command          Log Level
> >   -q, --quiet                     Disable logging
> >   -r, --retry                     Retry connection on failure
> >   -R, --reconnect                 Reconnect if disconnected
> >   -d, --debug=level               Debug Level (0 - 7)
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Holger.Esser at Convergys.com  Mon Oct 11 17:30:56 2010
From: Holger.Esser at Convergys.com (Esser, Holger)
Date: Mon, 11 Oct 2010 19:30:56 -0500
Subject: [Freeswitch-users] Setting origination_caller_id_number
In-Reply-To: <AANLkTimWD5go=yTpC5DkHx7b8Z+0PbJMLTG4WoNsMTW9@mail.gmail.com>
References: <AANLkTimWD5go=yTpC5DkHx7b8Z+0PbJMLTG4WoNsMTW9@mail.gmail.com>
Message-ID: <0FF607C13B7F7A42B5B80DA9EB204C9109D44BD5C1@srv-ex01-dal.intervoice.int>

I am looking in old postings to the list and it seems that in 2008 people ran across the same issue. In essence, I need a way to manipulate the FROM contact header to match my changing ANI. Currently it pulls it only from the gateway user name in the profile. I a looking in the sofia_glue and it seems that these params are set in switch_channel.c. Before I bang my head on the wall too many times in trying to change the code, has this already been addressed in a manner that I cannot find?
It seems that origination_caller_id_number sets only RPID. Is that accurate? Teliax cannot set the ANI based on RPID but needs it in the FROM contact header.

Any help would be greatly appreciated.

Holger

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Holger Esser
Sent: Friday, October 08, 2010 1:33 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Setting origination_caller_id_number

Hi,

Whenever I set the origination_caller_is_number in my dial request like this for a local or external context,
perl http.pl<http://http.pl> originate {origination_caller_id_number=5xxxxxxxxxsofia/internal/1 at 148.181.145.148<mailto:1 at 148.181.145.148> 5000 the caller id is set.
Whenever I use the gateway, it is no longer working and the username in the gateway becomes the ani number in the SIP trace.
perl http.pl<http://http.pl> originate {origination_caller_id_number=972xxxxxxx}sofia/gateway/teliax/4695xxxxxxx 5000

Any ideas?

Thx

No virus found in this incoming message.
Checked by AVG - www.avg.com
Version: 9.0.862 / Virus Database: 271.1.1/3183 - Release Date: 10/08/10 13:34:00

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From Nabble at slickdeals.endjunk.com  Mon Oct 11 17:44:53 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 11 Oct 2010 17:44:53 -0700 (PDT)
Subject: [Freeswitch-users] Newbie (sort of) questions
In-Reply-To: <4CB38CFD.8040809@mgtech.com>
References: <DFB4B88E-6834-4D22-9DEE-616AB50F0D91@stillnewt.org>
	<AANLkTim+zr9DWNQ9QCVccMETYb8Y=OT89t8QhjyfWVv4@mail.gmail.com>
	<4CB38CFD.8040809@mgtech.com>
Message-ID: <1286844293187-5625104.post@n2.nabble.com>



Mario wrote:
> 1. If you haven't already buy the book and study it! You definitely
> won't need/want a GUI after that.
> 2. Search the wiki, it has some very good configs to get started, use
> pieces from multiple samples. Google search help with the wiki too.
> 3. Search Google to find the pages on this mailing list easily.
> 4. As a last resort, post on this group.
These are very simple and very good suggestions. This was how I learnt to
configure and get my FS working. When I started to learn FS, the book wasn't
out yet. At the time, the only best resources was to read Wiki and follow
the examples. Right now, I am using an inexpensive and efficient Seagate 
http://www.seagate.com/www/en-us/products/network_storage/freeagent_dockstar
DockStar  to host FS + NAS + Printer Server to serve all my needs in one
little box that consumes about 3Watts of electricity.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Newbie-sort-of-questions-tp5624290p5625104.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From david.ponzone at ipeva.fr  Mon Oct 11 17:56:48 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 12 Oct 2010 02:56:48 +0200
Subject: [Freeswitch-users] Setting origination_caller_id_number
In-Reply-To: <0FF607C13B7F7A42B5B80DA9EB204C9109D44BD5C1@srv-ex01-dal.intervoice.int>
References: <AANLkTimWD5go=yTpC5DkHx7b8Z+0PbJMLTG4WoNsMTW9@mail.gmail.com>
	<0FF607C13B7F7A42B5B80DA9EB204C9109D44BD5C1@srv-ex01-dal.intervoice.int>
Message-ID: <DE3FC56E-12E3-4408-B361-254C45F25598@ipeva.fr>

Holger,

try adding sip_cid_type=none in your {}.

And in your gateway definition, you may add:
<param name="caller-id-in-from" value="true"/>
David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 12/10/2010 ? 02:30, Esser, Holger a ?crit :

> I am looking in old postings to the list and it seems that in 2008  
> people ran across the same issue. In essence, I need a way to  
> manipulate the FROM contact header to match my changing ANI.  
> Currently it pulls it only from the gateway user name in the  
> profile. I a looking in the sofia_glue and it seems that these  
> params are set in switch_channel.c. Before I bang my head on the  
> wall too many times in trying to change the code, has this already  
> been addressed in a manner that I cannot find?
> It seems that origination_caller_id_number sets only RPID. Is that  
> accurate? Teliax cannot set the ANI based on RPID but needs it in  
> the FROM contact header.
>
> Any help would be greatly appreciated.
>
> Holger
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Holger Esser
> Sent: Friday, October 08, 2010 1:33 PM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] Setting origination_caller_id_number
>
> Hi,
>
> Whenever I set the origination_caller_is_number in my dial request  
> like this for a local or external context,
> perl http.pl originate {origination_caller_id_number=5xxxxxxxxxsofia/internal/1 at 148.181.145.148 
>  5000 the caller id is set.
> Whenever I use the gateway, it is no longer working and the username  
> in the gateway becomes the ani number in the SIP trace.
> perl http.pl originate  
> {origination_caller_id_number=972xxxxxxx}sofia/gateway/teliax/ 
> 4695xxxxxxx 5000
>
> Any ideas?
>
> Thx
> No virus found in this incoming message.
> Checked by AVG - www.avg.com
> Version: 9.0.862 / Virus Database: 271.1.1/3183 - Release Date:  
> 10/08/10 13:34:00
>
>
> This e-mail transmission may contain information that is  
> proprietary, privileged and/or confidential and is intended  
> exclusively for the person(s) to whom it is addressed. Any use,  
> copying, retention or disclosure by any person other than the  
> intended recipient or the intended recipient's designees is strictly  
> prohibited. If you are the intended recipient, you must treat the  
> information in confidence and in accordance with all laws related to  
> the privacy and confidentiality of such information. If you are not  
> the intended recipient or their designee, please notify the sender  
> immediately by return e-mail and delete all copies of this email,  
> including all attachments.
>
> Intervoice, Inc. 17811 Waterview Parkway Dallas, TX 75252 USA
>
> Intervoice Limited, Registered in England and Wales with number  
> 2601740, 50 Park Road, Gatley, Cheshire, SK8 4HZ. VAT Number:  
> 560421375
>
> Intervoice GmbH, Hagenauer Stra?e 55, 65203 Wiesbaden, Sitz der  
> Gesellschaft: Wiesbaden, Handelsregister: HRB 8186 (Amtsgericht  
> Wiesbaden), Gesch?ftsf?hrer: Wayne Barclay, Steffen Selbmann
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From Holger.Esser at Convergys.com  Mon Oct 11 18:29:06 2010
From: Holger.Esser at Convergys.com (Esser, Holger)
Date: Mon, 11 Oct 2010 20:29:06 -0500
Subject: [Freeswitch-users] Setting origination_caller_id_number
Message-ID: <0FF607C13B7F7A42B5B80DA9EB204C9109D435F2C4@srv-ex01-dal.intervoice.int>

Many many thanks!!! That addressed my issue.

Holger

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org
To: FreeSWITCH Users Help
Sent: Mon Oct 11 19:56:48 2010
Subject: Re: [Freeswitch-users] Setting origination_caller_id_number

Holger,

try adding sip_cid_type=none in your {}.

And in your gateway definition, you may add:

<param name="caller-id-in-from" value="true"/>

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr<mailto:david.ponzone at ipeva.fr>
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 12/10/2010 ? 02:30, Esser, Holger a ?crit :

I am looking in old postings to the list and it seems that in 2008 people ran across the same issue. In essence, I need a way to manipulate the FROM contact header to match my changing ANI. Currently it pulls it only from the gateway user name in the profile. I a looking in the sofia_glue and it seems that these params are set in switch_channel.c. Before I bang my head on the wall too many times in trying to change the code, has this already been addressed in a manner that I cannot find?
It seems that origination_caller_id_number sets only RPID. Is that accurate? Teliax cannot set the ANI based on RPID but needs it in the FROM contact header.

Any help would be greatly appreciated.

Holger

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Holger Esser
Sent: Friday, October 08, 2010 1:33 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Setting origination_caller_id_number

Hi,

Whenever I set the origination_caller_is_number in my dial request like this for a local or external context,
perl http.pl<http://http.pl> originate {origination_caller_id_number=5xxxxxxxxxsofia/internal/1 at 148.181.145.148<mailto:1 at 148.181.145.148> 5000 the caller id is set.
Whenever I use the gateway, it is no longer working and the username in the gateway becomes the ani number in the SIP trace.
perl http.pl<http://http.pl> originate {origination_caller_id_number=972xxxxxxx}sofia/gateway/teliax/4695xxxxxxx 5000

Any ideas?

Thx

No virus found in this incoming message.
Checked by AVG - www.avg.com<http://www.avg.com>
Version: 9.0.862 / Virus Database: 271.1.1/3183 - Release Date: 10/08/10 13:34:00

________________________________
This e-mail transmission may contain information that is proprietary, privileged and/or confidential and is intended exclusively for the person(s) to whom it is addressed. Any use, copying, retention or disclosure by any person other than the intended recipient or the intended recipient's designees is strictly prohibited. If you are the intended recipient, you must treat the information in confidence and in accordance with all laws related to the privacy and confidentiality of such information. If you are not the intended recipient or their designee, please notify the sender immediately by return e-mail and delete all copies of this email, including all attachments.

Intervoice, Inc. 17811 Waterview Parkway Dallas, TX 75252 USA

Intervoice Limited, Registered in England and Wales with number 2601740, 50 Park Road, Gatley, Cheshire, SK8 4HZ. VAT Number: 560421375

Intervoice GmbH, Hagenauer Stra?e 55, 65203 Wiesbaden, Sitz der Gesellschaft: Wiesbaden, Handelsregister: HRB 8186 (Amtsgericht Wiesbaden), Gesch?ftsf?hrer: Wayne Barclay, Steffen Selbmann
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


No virus found in this incoming message.
Checked by AVG - www.avg.com
Version: 9.0.862 / Virus Database: 271.1.1/3189 - Release Date: 10/11/10 13:34:00
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From covici at ccs.covici.com  Mon Oct 11 19:28:12 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 11 Oct 2010 22:28:12 -0400
Subject: [Freeswitch-users] Newbie (sort of) questions
In-Reply-To: <4CB38CFD.8040809@mgtech.com>
References: <DFB4B88E-6834-4D22-9DEE-616AB50F0D91@stillnewt.org>
	<AANLkTim+zr9DWNQ9QCVccMETYb8Y=OT89t8QhjyfWVv4@mail.gmail.com>
	<4CB38CFD.8040809@mgtech.com>
Message-ID: <5069.1286850492@ccs.covici.com>

I agree in most part, however I am worried about non technical people
working with xml to add/dchange extensions and routes, etc.  I have
gotten used to things somewhat, but what aboutthem.  Also, reporting
calling patterns from the database is a problem and also, conference
status -- I just tried with someone and he didn't like the cli interface
at all for conference manipulation.

Mario <mario_fs at mgtech.com> wrote:

> Hope this helps: I am replacing an SPA9000 (asterisk) with FS, testing
> on Linux then moving to a mac mini. I tried this 8 months ago and used
> some of the GUI options. My two cents: stick with XML parms. Things go
> wrong and a GUI gets in the way, it "hides from view" valuable info. I
> was able to get FS up and running the first try on Linux. osX 10.6.4
> took some work (search for my userid in the list from the last 2 weeks)
> but eventually worked. Although FS is done I am having one last tough
> problem no one here (or on IRC) attempted to help with and I can't
> imagine working on it without having done the XML stuff myself. To
> answer your question though, this is what I did and recommend:
> 
> 1. If you haven't already buy the book and study it! You definitely
> won't need/want a GUI after that.
> 2. Search the wiki, it has some very good configs to get started, use
> pieces from multiple samples. Google search help with the wiki too.
> 3. Search Google to find the pages on this mailing list easily.
> 4. As a last resort, post on this group.
> 
> On 10/11/2010 12:10 PM, Nyamul Hassan wrote:
> > There is a new project, which looks quite promising:
> > 
> > www.2600hz.org
> > 
> > Regards
> > HASSAN
> > 
> > 
> > On 2010-10-12, Martin Joseph <astmac at stillnewt.org> wrote:
> >> Hi Again FreeSWITCHers,
> >>
> >> I am building a new setup for my "new" house (I have been remodeling for 5
> >> years).
> >>
> >> I have build FS via GIT on my "new" xServe hardware (dual Xeon 2Ghz) which I
> >> was able to do pretty easily (Hurray!).  In the past I have struggled with
> >> updating my older FS install on my anemic g3/500 ibook install which is
> >> still running by the way.
> >>
> >> So,  now I am thinking about configuring my new installation, and I being no
> >> expert am not looking forward to figuring this all out again. My
> >> installation is overall pretty simple with a gateway to POTS (audiocodes
> >> MP114) and several Voip providors, one of whom is hopefully still in
> >> business (heh).
> >>
> >> Anyhow,  my first request is for GUI based FS configuration systems?  These
> >> could run on my OSX box (via X11 or Cocoa), or be some kind of web based
> >> thing.  It doesn't need to be super customizable, but not having to
> >> find/edit the XML files would be lovely. I am a mac person after all.
> >>
> >> If there are no recommended GUI's available, then pointing me to the best
> >> sample pages for configuration would also be greatly appreciated...
> >>
> >> Thanks again for the great software and all of your efforts!
> >> Marty
> >>
> >> PS looking forward to killing me last Asterisk install...
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> > 
> 
> -- 
> *Mario*
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From mnhassan at usa.net  Mon Oct 11 20:21:16 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Tue, 12 Oct 2010 09:21:16 +0600
Subject: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4
In-Reply-To: <AANLkTinFAifa6BPLDqYVEEh1onCobbgJX4DHC_Kzdn_u@mail.gmail.com>
References: <AANLkTingeYfPT_b6dG3bKQOVjxYsmBTKhDuLw_3BKGac@mail.gmail.com>
	<AANLkTi=YZsrcSWbrnUrZ6+P_fnXpQVAROCNc+9okF0X9@mail.gmail.com>
	<AANLkTinFAifa6BPLDqYVEEh1onCobbgJX4DHC_Kzdn_u@mail.gmail.com>
Message-ID: <AANLkTinRZY_DwDXqEDumMaKuXc39zveHayHXCYZa-=_q@mail.gmail.com>

That file lists the port that FS listens on for event socket connections.

Run a "netstat -nlp | grep <port>" to check which IP that port is
being listened on.

Then you can use command line switches for fs_cli to force it to
connect on that IP/port.

Regards
HASSAN


On 2010-10-12, Peter Schrock <peter.schrock at gmail.com> wrote:
> just to make sure what am I comparing event_socket.conf.xml to? I have found
> that it is set up with the default settings but am not sure how or to what I
> am suppose to fix. I understand that I am comparing it to FS, but I am
> wondering if there is some kind of file that I am looking for to compare the
> two.
>
> PeterS
>
> On Fri, Oct 8, 2010 at 4:56 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> 1. Check FS is running
>> 2. Check the port you're connecting to matches the one in
>> event_socket.conf.xml
>> 3. Check that event_socket.conf.xml binding to the same IP you're
>> connecting to (e.g. 0.0.0.0 if you're connecting from a remote
>> machine, since the default 127.0.0.1 won't work then)
>> 4. Use netstat on the FS server to verify FreeSWITCH is actually
>> listening on the port you're trying to connect to (I've had a syntax
>> error in the config file make the module fail to load in the past
>> which left FS running but with no ESL socket). (If the module fails to
>> load there'll also be an error in the log file).
>> 5. Check a firewall isn't blocking access to the port
>>
>> -Steve
>>
>>
>>
>> On 7 October 2010 01:23, Peter Schrock <peter.schrock at gmail.com> wrote:
>> > Okay, so I managed to get FS working the other day and I even managed to
>> > test a call and test my voicemail. All seemed to be working smoothly
>> until,
>> > because of the rain here, my power went out and I had to reboot my
>> computer.
>> > I logged in through the terminal, set up FS in background went to fs_cli
>> and
>> > I get this error message:
>> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
>> > Connection
>> > Error]
>> > I managed to figure out that in the file "fs_cli.c" on line 1206 you
>> > find
>> > the code for displaying this error message. The problem is that I don't
>> know
>> > why this error message is occurring. Does anyone have any helpful hints
>> as
>> > to what I should look at to resolve this problem?
>> > I even tried going to the git tree and make current, but that gave me
>> > problems that forced me to turn off mod_spandsp and mod_skyopen in the
>> > modules.conf, which I had running earlier. Any thoughts?
>> > PeterS
>> > PS
>> > I am not sure if this is of any help, but in addition to the error line
>> > above, it also posted this info:
>> > Usage: fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x
>> > command] [profile]
>> >   -?,-h --help                    Usage Information
>> >   -H, --host=hostname             Host to connect
>> >   -P, --port=port                 Port to connect (1 - 65535)
>> >   -u, --user=user at domain          user at domain
>> >   -p, --password=password         Password
>> >   -x, --execute=command           Execute Command and Exit
>> >   -l, --loglevel=command          Log Level
>> >   -q, --quiet                     Disable logging
>> >   -r, --retry                     Retry connection on failure
>> >   -R, --reconnect                 Reconnect if disconnected
>> >   -d, --debug=level               Debug Level (0 - 7)
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>

-- 
Sent from my mobile device


From msc at freeswitch.org  Mon Oct 11 22:18:51 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Oct 2010 22:18:51 -0700
Subject: [Freeswitch-users] Routing DIDs to Dynamically Registered SIP
	User
In-Reply-To: <AANLkTimCNR3KYpb0dxnikbBAFi2uL_24U8u3rq2LBhYO@mail.gmail.com>
References: <4CAEAD0B.8020704@gmail.com>
	<AANLkTimv0hvr6BYfKZW1zrdGgHrnMGxpidmpNL-5eu1T@mail.gmail.com>
	<AANLkTimCNR3KYpb0dxnikbBAFi2uL_24U8u3rq2LBhYO@mail.gmail.com>
Message-ID: <AANLkTi=PMDD2aV45pivoB69OFaUHkDsZJESO-M_A5dPV@mail.gmail.com>

Ah, I was reading too much into that.

If I understand correctly you need to send the INVITE to the actual
endpoint, but you need the target URI be the same as the URI in the To:
field? Is that even legal in SIP? (Not that legal matters since vendors do
whatever they want anyway...)

-MC

On Mon, Oct 11, 2010 at 3:59 PM, David Allen <thedjallen at gmail.com> wrote:

> Hi Michael,
>
> Its a SIP UA that registers locally to Freeswitch.
>
> Regards,
> David
>
> On Tue, Oct 12, 2010 at 9:51 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> > What is a "dynamically registered SIP user"?
> > -MC
> >
> > On Thu, Oct 7, 2010 at 10:32 PM, David Allen <thedjallen at gmail.com>
> wrote:
> >>
> >>  Hi,
> >>
> >> I'm trying to send multiple Direct Indial Numbers down to a dynamically
> >> registered SIP User. I need to ensure that both the To and Target URI
> >> contain the direct Indial number. I'm able to modify the SIP TO Header
> >> of a call that is sent to them like below:
> >>
> >> <action application="bridge"
> >> data="sofia/external/56778977%${domain}^61390009000"/>
> >>
> >> which sends the request as:
> >>
> >>
> >> ------------------------------------------------------------------------
> >>
> >>    INVITE sip:56778977 at 192.168.22.2:5061 SIP/2.0
> >>
> >>    Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j
> >>
> >>    Max-Forwards: 69
> >>
> >>    From: "0390001000" <sip:0390001000 at 192.168.2.200<sip%3A0390001000 at 192.168.2.200>
> >;tag=K4HHaZ9v1H07Q
> >>
> >>    To: <sip:61390009000 at 192.168.2.200 <sip%3A61390009000 at 192.168.2.200>
> >
> >>
> >>  But in order to maintain compatability with a number of PBX's/VoIP
> >> devices on the market, I need to be able to send the invite to the
> >> dynamically registered SIP user, however have it set the Target URI and
> >> To as the same contact number like below:
> >>
> >>
> >> ------------------------------------------------------------------------
> >>
> >>    INVITE sip:61390009000 at 192.168.22.2:5061 SIP/2.0
> >>
> >>    Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j
> >>
> >>    Max-Forwards: 69
> >>
> >>    From: "0390001000" <sip:0390001000 at 192.168.2.200<sip%3A0390001000 at 192.168.2.200>
> >;tag=K4HHaZ9v1H07Q
> >>
> >>    To: <sip:61390009000 at 192.168.22.2:5061>
> >>
> >> Is there a way to do this setting via variables? I can't seem to find
> >> any details for it.
> >>
> >> Thanks
> >>
> >> David
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Oct 11 22:24:27 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Oct 2010 22:24:27 -0700
Subject: [Freeswitch-users] Newbie (sort of) questions
In-Reply-To: <5069.1286850492@ccs.covici.com>
References: <DFB4B88E-6834-4D22-9DEE-616AB50F0D91@stillnewt.org>
	<AANLkTim+zr9DWNQ9QCVccMETYb8Y=OT89t8QhjyfWVv4@mail.gmail.com>
	<4CB38CFD.8040809@mgtech.com> <5069.1286850492@ccs.covici.com>
Message-ID: <AANLkTik4nj2vxqCMLH=SOZUmS7XfU+Uu4UkQsC9kv9hq@mail.gmail.com>

The CLI is most definitely not for end users. The CLI is for admins and
developers. End users need a purty GUI. FreeSWITCH is like being in the Lego
store: all the bricks you can imagine, but you need to put them together.

I get the impression that a conference management interface is something
that people want/need. My guess is that you won't find one for free, at
least not until some generous soul writes one and gives it to the world. :)

In the meantime I'm sure there are FreeSWITCHers around who've done some
simple Web-based conference call interfaces. They may not be free, or they
may be "free" but not open-source, etc. Ask around.

-MC

On Mon, Oct 11, 2010 at 7:28 PM, <covici at ccs.covici.com> wrote:

> I agree in most part, however I am worried about non technical people
> working with xml to add/dchange extensions and routes, etc.  I have
> gotten used to things somewhat, but what aboutthem.  Also, reporting
> calling patterns from the database is a problem and also, conference
> status -- I just tried with someone and he didn't like the cli interface
> at all for conference manipulation.
>
> Mario <mario_fs at mgtech.com> wrote:
>
> > Hope this helps: I am replacing an SPA9000 (asterisk) with FS, testing
> > on Linux then moving to a mac mini. I tried this 8 months ago and used
> > some of the GUI options. My two cents: stick with XML parms. Things go
> > wrong and a GUI gets in the way, it "hides from view" valuable info. I
> > was able to get FS up and running the first try on Linux. osX 10.6.4
> > took some work (search for my userid in the list from the last 2 weeks)
> > but eventually worked. Although FS is done I am having one last tough
> > problem no one here (or on IRC) attempted to help with and I can't
> > imagine working on it without having done the XML stuff myself. To
> > answer your question though, this is what I did and recommend:
> >
> > 1. If you haven't already buy the book and study it! You definitely
> > won't need/want a GUI after that.
> > 2. Search the wiki, it has some very good configs to get started, use
> > pieces from multiple samples. Google search help with the wiki too.
> > 3. Search Google to find the pages on this mailing list easily.
> > 4. As a last resort, post on this group.
> >
> > On 10/11/2010 12:10 PM, Nyamul Hassan wrote:
> > > There is a new project, which looks quite promising:
> > >
> > > www.2600hz.org
> > >
> > > Regards
> > > HASSAN
> > >
> > >
> > > On 2010-10-12, Martin Joseph <astmac at stillnewt.org> wrote:
> > >> Hi Again FreeSWITCHers,
> > >>
> > >> I am building a new setup for my "new" house (I have been remodeling
> for 5
> > >> years).
> > >>
> > >> I have build FS via GIT on my "new" xServe hardware (dual Xeon 2Ghz)
> which I
> > >> was able to do pretty easily (Hurray!).  In the past I have struggled
> with
> > >> updating my older FS install on my anemic g3/500 ibook install which
> is
> > >> still running by the way.
> > >>
> > >> So,  now I am thinking about configuring my new installation, and I
> being no
> > >> expert am not looking forward to figuring this all out again. My
> > >> installation is overall pretty simple with a gateway to POTS
> (audiocodes
> > >> MP114) and several Voip providors, one of whom is hopefully still in
> > >> business (heh).
> > >>
> > >> Anyhow,  my first request is for GUI based FS configuration systems?
>  These
> > >> could run on my OSX box (via X11 or Cocoa), or be some kind of web
> based
> > >> thing.  It doesn't need to be super customizable, but not having to
> > >> find/edit the XML files would be lovely. I am a mac person after all.
> > >>
> > >> If there are no recommended GUI's available, then pointing me to the
> best
> > >> sample pages for configuration would also be greatly appreciated...
> > >>
> > >> Thanks again for the great software and all of your efforts!
> > >> Marty
> > >>
> > >> PS looking forward to killing me last Asterisk install...
> > >> _______________________________________________
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >>
> > >
> >
> > --
> > *Mario*
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From covici at ccs.covici.com  Mon Oct 11 22:25:56 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 12 Oct 2010 01:25:56 -0400
Subject: [Freeswitch-users] administering a conference
Message-ID: <6851.1286861156@ccs.covici.com>

Hi.  I would like an alternate means of administering a conference other
than the command line -- for me its OK, but other prefer a web interface
and te one you use at conference.freeswitch.org would be just fine if I
could find all its pieces  -- event scripts database tables, whatever.
For me, this would help spread fs and not have people say its unusable
because of these trivialities.

any assistance would be appreciated.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From david.ponzone at ipeva.fr  Mon Oct 11 23:24:15 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 12 Oct 2010 08:24:15 +0200
Subject: [Freeswitch-users] Routing DIDs to Dynamically Registered SIP
	User
In-Reply-To: <AANLkTi=PMDD2aV45pivoB69OFaUHkDsZJESO-M_A5dPV@mail.gmail.com>
References: <4CAEAD0B.8020704@gmail.com>
	<AANLkTimv0hvr6BYfKZW1zrdGgHrnMGxpidmpNL-5eu1T@mail.gmail.com>
	<AANLkTimCNR3KYpb0dxnikbBAFi2uL_24U8u3rq2LBhYO@mail.gmail.com>
	<AANLkTi=PMDD2aV45pivoB69OFaUHkDsZJESO-M_A5dPV@mail.gmail.com>
Message-ID: <157B0F19-E2AD-4B3E-9772-69CE015BE966@ipeva.fr>

I really don't think it is as:
1/ i was not able to find a parameter to alter the INVITE URI
2/ FS itself has a parameter to change the behaviour and use the SIP  
To instead of the INVITE URI as the destination_number, so it would  
tend to prove that it's the other party's job to use the right field.

David (Allen, I am not talking to myself :) ), can you give us some  
examples of PBXs who rely only on the INVITE URI ?
Are you sure they don't have a parameter to change that ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 12/10/2010 ? 07:18, Michael Collins a ?crit :

> Ah, I was reading too much into that.
>
> If I understand correctly you need to send the INVITE to the actual  
> endpoint, but you need the target URI be the same as the URI in the  
> To: field? Is that even legal in SIP? (Not that legal matters since  
> vendors do whatever they want anyway...)
>
> -MC
>
> On Mon, Oct 11, 2010 at 3:59 PM, David Allen <thedjallen at gmail.com>  
> wrote:
> Hi Michael,
>
> Its a SIP UA that registers locally to Freeswitch.
>
> Regards,
> David
>
> On Tue, Oct 12, 2010 at 9:51 AM, Michael Collins  
> <msc at freeswitch.org> wrote:
> > What is a "dynamically registered SIP user"?
> > -MC
> >
> > On Thu, Oct 7, 2010 at 10:32 PM, David Allen  
> <thedjallen at gmail.com> wrote:
> >>
> >>  Hi,
> >>
> >> I'm trying to send multiple Direct Indial Numbers down to a  
> dynamically
> >> registered SIP User. I need to ensure that both the To and Target  
> URI
> >> contain the direct Indial number. I'm able to modify the SIP TO  
> Header
> >> of a call that is sent to them like below:
> >>
> >> <action application="bridge"
> >> data="sofia/external/56778977%${domain}^61390009000"/>
> >>
> >> which sends the request as:
> >>
> >>
> >>  
> ------------------------------------------------------------------------
> >>
> >>    INVITE sip:56778977 at 192.168.22.2:5061 SIP/2.0
> >>
> >>    Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j
> >>
> >>    Max-Forwards: 69
> >>
> >>    From: "0390001000" <sip: 
> 0390001000 at 192.168.2.200>;tag=K4HHaZ9v1H07Q
> >>
> >>    To: <sip:61390009000 at 192.168.2.200>
> >>
> >>  But in order to maintain compatability with a number of PBX's/VoIP
> >> devices on the market, I need to be able to send the invite to the
> >> dynamically registered SIP user, however have it set the Target  
> URI and
> >> To as the same contact number like below:
> >>
> >>
> >>  
> ------------------------------------------------------------------------
> >>
> >>    INVITE sip:61390009000 at 192.168.22.2:5061 SIP/2.0
> >>
> >>    Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j
> >>
> >>    Max-Forwards: 69
> >>
> >>    From: "0390001000" <sip: 
> 0390001000 at 192.168.2.200>;tag=K4HHaZ9v1H07Q
> >>
> >>    To: <sip:61390009000 at 192.168.22.2:5061>
> >>
> >> Is there a way to do this setting via variables? I can't seem to  
> find
> >> any details for it.
> >>
> >> Thanks
> >>
> >> David
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From oseslija at gmail.com  Mon Oct 11 23:50:26 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Tue, 12 Oct 2010 08:50:26 +0200
Subject: [Freeswitch-users] Routing DIDs to Dynamically Registered SIP
	User
In-Reply-To: <4CAEAFC4.3040504@gmail.com>
References: <4CAEAFC4.3040504@gmail.com>
Message-ID: <AANLkTikZDoW1_rSbDoCL+onwgj9iNkE4tbc=ycFTd_fO@mail.gmail.com>

<action application="bridge" data="{sip_invite_req_uri=
sip:61390009000 at 192.168.22.2:5061
,sip_invite_to_uri=sip:61390009000 at 192.168.22.2:5061
}sofia/external/56778977%${domain}"/>

On Fri, Oct 8, 2010 at 7:44 AM, David Allen <thedjallen at gmail.com> wrote:

>  Hi,
>
> I'm trying to send multiple Direct Indial Numbers down to a dynamically
> registered SIP User. I need to ensure that both the To and Target URI
> contain the direct Indial number. I'm able to modify the SIP TO Header
> of a call that is sent to them like below:
>
> <action application="bridge"
> data="sofia/external/56778977%${domain}^61390009000"/>
>
> which sends the request as:
>
>    ------------------------------------------------------------------------
>
>    INVITE sip:56778977 at 192.168.22.2:5061 SIP/2.0
>
>    Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j
>
>    Max-Forwards: 69
>
>    From: "0390001000" <sip:0390001000 at 192.168.2.200<sip%3A0390001000 at 192.168.2.200>
> >;tag=K4HHaZ9v1H07Q
>
>    To: <sip:61390009000 at 192.168.2.200 <sip%3A61390009000 at 192.168.2.200>>
>
>  But in order to maintain compatability with a number of PBX's/VoIP
> devices on the market, I need to be able to send the invite to the
> dynamically registered SIP user, however have it set the Target URI and
> To as the same contact number like below:
>
>    ------------------------------------------------------------------------
>
>    INVITE sip:61390009000 at 192.168.22.2:5061 SIP/2.0
>
>    Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j
>
>    Max-Forwards: 69
>
>    From: "0390001000" <sip:0390001000 at 192.168.2.200<sip%3A0390001000 at 192.168.2.200>
> >;tag=K4HHaZ9v1H07Q
>
>    To: <sip:61390009000 at 192.168.22.2:5061>
>
> Is there a way to do this setting via variables? I can't seem to find
> any details for it.
>
> Thanks
>
> David
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.ponzone at ipeva.fr  Tue Oct 12 00:17:11 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 12 Oct 2010 09:17:11 +0200
Subject: [Freeswitch-users] Routing DIDs to Dynamically Registered SIP
	User
In-Reply-To: <AANLkTikZDoW1_rSbDoCL+onwgj9iNkE4tbc=ycFTd_fO@mail.gmail.com>
References: <4CAEAFC4.3040504@gmail.com>
	<AANLkTikZDoW1_rSbDoCL+onwgj9iNkE4tbc=ycFTd_fO@mail.gmail.com>
Message-ID: <C42ECEF7-69A0-476C-85B8-6C205D680F83@ipeva.fr>

Ahhh thanks Ognjen!

it was not on the wiki, it is now :)

http://wiki.freeswitch.org/wiki/Variable_sip_invite_req_uri

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 12/10/2010 ? 08:50, Ognjen Seslija a ?crit :

> <action application="bridge" data="{sip_invite_req_uri=sip:61390009000 at 192.168.22.2:5061,sip_invite_to_uri=sip:61390009000 at 192.168.22.2:5061 
> }sofia/external/56778977%${domain}"/>
>
> On Fri, Oct 8, 2010 at 7:44 AM, David Allen <thedjallen at gmail.com>  
> wrote:
>  Hi,
>
> I'm trying to send multiple Direct Indial Numbers down to a  
> dynamically
> registered SIP User. I need to ensure that both the To and Target URI
> contain the direct Indial number. I'm able to modify the SIP TO Header
> of a call that is sent to them like below:
>
> <action application="bridge"
> data="sofia/external/56778977%${domain}^61390009000"/>
>
> which sends the request as:
>
>     
> ------------------------------------------------------------------------
>
>    INVITE sip:56778977 at 192.168.22.2:5061 SIP/2.0
>
>    Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j
>
>    Max-Forwards: 69
>
>    From: "0390001000" <sip:0390001000 at 192.168.2.200>;tag=K4HHaZ9v1H07Q
>
>    To: <sip:61390009000 at 192.168.2.200>
>
>  But in order to maintain compatability with a number of PBX's/VoIP
> devices on the market, I need to be able to send the invite to the
> dynamically registered SIP user, however have it set the Target URI  
> and
> To as the same contact number like below:
>
>     
> ------------------------------------------------------------------------
>
>    INVITE sip:61390009000 at 192.168.22.2:5061 SIP/2.0
>
>    Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j
>
>    Max-Forwards: 69
>
>    From: "0390001000" <sip:0390001000 at 192.168.2.200>;tag=K4HHaZ9v1H07Q
>
>    To: <sip:61390009000 at 192.168.22.2:5061>
>
> Is there a way to do this setting via variables? I can't seem to find
> any details for it.
>
> Thanks
>
> David
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From woodydickson at gmail.com  Tue Oct 12 00:25:28 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Tue, 12 Oct 2010 15:25:28 +0800
Subject: [Freeswitch-users] 3-way calling fails with git
Message-ID: <AANLkTi==jVtq3AosUD=pYTjCq3O-k7dtvVPbxB3fnaa4@mail.gmail.com>

Hi,

I am having problem with using the att_xfer app after upgrading to the
latest git.  After the callee presses a meta_app key and bridged a 3rd party
in, the callee got hung up immediately leaving the 3rd part and the original
caller talking to each other only.

I was able to get 3-way call to work in the previous release of fs.  Does
anyone know if this is a bug or a bad config on my end?

Thanks,
Woody
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From babak.freeswitch at gmail.com  Tue Oct 12 02:40:33 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Tue, 12 Oct 2010 13:10:33 +0330
Subject: [Freeswitch-users] fs_cli
In-Reply-To: <1286721199996-5620475.post@n2.nabble.com>
References: <AANLkTi=L4VArPuBSDV5hgGtJYr3Kbkoh1XjGTtPwM0BE@mail.gmail.com>
	<AANLkTi=HeFmsJ3BVTdBkc9YHAK7Apgcivj-cX0degiDm@mail.gmail.com>
	<AANLkTikRX3QcFfHhAA8+FHSddok0D_bDWDiqKQDCZwky@mail.gmail.com>
	<AANLkTinp4GdOyPZUcF1FoHB=dB2YbxEfi1JLrvvM2L23@mail.gmail.com>
	<1286721199996-5620475.post@n2.nabble.com>
Message-ID: <AANLkTikuppQz181vR5D52Rz7JabLB6oqo5OjjLakqP-z@mail.gmail.com>

thank u all problem solved
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From mnhassan at usa.net  Tue Oct 12 03:06:10 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Tue, 12 Oct 2010 16:06:10 +0600
Subject: [Freeswitch-users] fs_cli
In-Reply-To: <AANLkTikuppQz181vR5D52Rz7JabLB6oqo5OjjLakqP-z@mail.gmail.com>
References: <AANLkTi=L4VArPuBSDV5hgGtJYr3Kbkoh1XjGTtPwM0BE@mail.gmail.com>
	<AANLkTi=HeFmsJ3BVTdBkc9YHAK7Apgcivj-cX0degiDm@mail.gmail.com>
	<AANLkTikRX3QcFfHhAA8+FHSddok0D_bDWDiqKQDCZwky@mail.gmail.com>
	<AANLkTinp4GdOyPZUcF1FoHB=dB2YbxEfi1JLrvvM2L23@mail.gmail.com>
	<1286721199996-5620475.post@n2.nabble.com>
	<AANLkTikuppQz181vR5D52Rz7JabLB6oqo5OjjLakqP-z@mail.gmail.com>
Message-ID: <AANLkTim8SPYa6c1A+_+MxwR72wGyyf5G5eYX3e8yfnws@mail.gmail.com>

What was the problem? If you think it is needed, can you please update the Wiki?

Regards
HASSAN


On 2010-10-12, babak yakhchali <babak.freeswitch at gmail.com> wrote:
> thank u all problem solved
>

-- 
Sent from my mobile device


From babak.freeswitch at gmail.com  Tue Oct 12 03:32:14 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Tue, 12 Oct 2010 14:02:14 +0330
Subject: [Freeswitch-users] fs_cli
In-Reply-To: <AANLkTim8SPYa6c1A+_+MxwR72wGyyf5G5eYX3e8yfnws@mail.gmail.com>
References: <AANLkTi=L4VArPuBSDV5hgGtJYr3Kbkoh1XjGTtPwM0BE@mail.gmail.com>
	<AANLkTi=HeFmsJ3BVTdBkc9YHAK7Apgcivj-cX0degiDm@mail.gmail.com>
	<AANLkTikRX3QcFfHhAA8+FHSddok0D_bDWDiqKQDCZwky@mail.gmail.com>
	<AANLkTinp4GdOyPZUcF1FoHB=dB2YbxEfi1JLrvvM2L23@mail.gmail.com>
	<1286721199996-5620475.post@n2.nabble.com>
	<AANLkTikuppQz181vR5D52Rz7JabLB6oqo5OjjLakqP-z@mail.gmail.com>
	<AANLkTim8SPYa6c1A+_+MxwR72wGyyf5G5eYX3e8yfnws@mail.gmail.com>
Message-ID: <AANLkTikHjX_qhr=iNO=6xwyhfS_s_dcdmOEydTbRFOBN@mail.gmail.com>

No just my fault
I was using a wrong way to output api result instead of using context!
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From mgg at giagnocavo.net  Tue Oct 12 03:31:20 2010
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Tue, 12 Oct 2010 06:31:20 -0400
Subject: [Freeswitch-users] Mono 2.8 released and need to
	update	mod_managed
In-Reply-To: <AANLkTi=_aLLty1F_iFkf_vQRuHzWTYJrZVc-7Zkos=Dh@mail.gmail.com>
References: <AANLkTin3rM+NaW1_O2d4HaDL9yuESGmZvd8v29Kxh6Y8@mail.gmail.com>
	<AANLkTimoS2=NaMhg2dVVEuyEjwEqagLie3KDVNHYv=8B@mail.gmail.com>
	<AANLkTi=_aLLty1F_iFkf_vQRuHzWTYJrZVc-7Zkos=Dh@mail.gmail.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C670367D5C583@mse17be1.mse17.exchange.ms>

Thanks for helping out. I just saw the Mono 2.8 announcement and it looks exciting.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yitzchok
Sent: Wednesday, October 06, 2010 9:21 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Mono 2.8 released and need to update mod_managed

I got it to work and attached is the patch file with the changes I made to get it to work. (Jeff this includes the changes from my other post)

This code changes has to be reviewed and I think you should checkout mono_assembly_name_free (name); I don't call that but maybe it should be called somewhere in the code.

Thanks

Yitzchok

On Wed, Oct 6, 2010 at 9:48 PM, Yitzchok <adminjew at gmail.com<mailto:adminjew at gmail.com>> wrote:
It seems like glib itself is easy to get around by removing the glig includes and changing g_free(x) to mono_free(x)

But I am getting stuck by this one.

?         The MonoAssemblyName struct is no longer fully visible: to access its fields

you need to use the newly-provided accessors. Note also that it can't be allocated on the stack anymore and you'll need to create and destroy it with the following API:





MonoAssemlyName *aname = mono_assembly_name_new ("mscorlib");

mono_assembly_name_free (aname);
(from http://www.mono-project.com/Embedding_Mono)

Yitzchok


On Wed, Oct 6, 2010 at 9:00 PM, Yitzchok <adminjew at gmail.com<mailto:adminjew at gmail.com>> wrote:
Mono 2.8 was released today with support for C# 4 and more http://www.mono-project.com/Release_Notes_Mono_2.8

It seems like there was some changes in the way applications embed mono (which includes removing the reference to glib and replacing it with eglib from what I understand) in this link you can find information for the changes made with embedding mono http://www.mono-project.com/Embedding_Mono

I am trying to see if I can make it work but since I am just a C# windows developer (not a c or c++ dev) what I am doing is just hacking around and don't know if I will even get it to work but I think a c++ linux developer might get it to work with only a little work (the info needed I think is in the link above).

So I am wondering if anyone is interested to get it to work.


Yitzchok

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From sobolewski at gmail.com  Tue Oct 12 06:22:47 2010
From: sobolewski at gmail.com (Piotr Sobolewski)
Date: Tue, 12 Oct 2010 15:22:47 +0200
Subject: [Freeswitch-users] bad audio of recorded session
Message-ID: <AANLkTimx7+mSh8LKsj9A3Qpuewgvwax=Ou4J0qk7y1JL@mail.gmail.com>

Hi

When conversation of the call is being saved to a file
(record_session) audio tempo in such recording is like slowed down
despite that call conversations lacks this issue.
Above happens when call comes from asterisk. (Asterisk-->FS-->UA) WRONG
If the call comes from IP-Phone, recording is normal.  (UA-->FS-->Asterisk) OK

I've tried multiple options for example:
 <action application="export" data="rtp_rewrite_timestamps=true"/>
 <action application="set" data="bridge_generate_comfort_noise=true"/>

And setting  <param name="rtp-timer-name" value="none"/> in the sofia
profiles didn't solved the problem either.

Global Codecs definition.

 <X-PRE-PROCESS cmd="set"
data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU at 20i,PCMA at 20i,GSM"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU at 20i,PCMA at 20i,GSM"/>

Is this known issue?


-- 
Piotr Sobolewski
sobolewski at gmail.com


From Nabble at slickdeals.endjunk.com  Tue Oct 12 07:13:55 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 12 Oct 2010 07:13:55 -0700 (PDT)
Subject: [Freeswitch-users] Newbie (sort of) questions
In-Reply-To: <AANLkTik4nj2vxqCMLH=SOZUmS7XfU+Uu4UkQsC9kv9hq@mail.gmail.com>
References: <DFB4B88E-6834-4D22-9DEE-616AB50F0D91@stillnewt.org>
	<AANLkTim+zr9DWNQ9QCVccMETYb8Y=OT89t8QhjyfWVv4@mail.gmail.com>
	<4CB38CFD.8040809@mgtech.com> <5069.1286850492@ccs.covici.com>
	<AANLkTik4nj2vxqCMLH=SOZUmS7XfU+Uu4UkQsC9kv9hq@mail.gmail.com>
Message-ID: <1286892835983-5626871.post@n2.nabble.com>



mercutioviz wrote:
> The CLI is most definitely not for end users. The CLI is for admins and
> developers. End users need a purty GUI. FreeSWITCH is like being in the
> Lego
> store: all the bricks you can imagine, but you need to put them together.
+1.

Unfortunately, one will probably never learn the inside-out and the
capabilities an FS system can offer if GUI gets involved. In this case, one
is at the mercy of what a GUI can offer which perhaps is no more than 5% of
the total power of FS. However, for a personal daily usage as a simple PBX
system that OP wants, this will probably be different. In this case, I
reckon a small foot-print, i.e. an inexpensive Linux embedded, system that
consumes no more than 5Watts of electricity can easily and smoothly
host/operate FS + GUI these days.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Newbie-sort-of-questions-tp5624290p5626871.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at freeswitch.org  Tue Oct 12 07:30:54 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Oct 2010 09:30:54 -0500
Subject: [Freeswitch-users] Mono 2.8 released and need to update
	mod_managed
In-Reply-To: <AANLkTin3rM+NaW1_O2d4HaDL9yuESGmZvd8v29Kxh6Y8@mail.gmail.com>
References: <AANLkTin3rM+NaW1_O2d4HaDL9yuESGmZvd8v29Kxh6Y8@mail.gmail.com>
Message-ID: <031F577B-CBFB-44B9-9764-68B4745657B4@freeswitch.org>

Patches belong on http://jira.freeswitch.org if you can please.

/b

On Oct 6, 2010, at 8:00 PM, Yitzchok wrote:

> Mono 2.8 was released today with support for C# 4 and more http://www.mono-project.com/Release_Notes_Mono_2.8
> 
> It seems like there was some changes in the way applications embed mono (which includes removing the reference to glib and replacing it with eglib from what I understand) in this link you can find information for the changes made with embedding mono http://www.mono-project.com/Embedding_Mono
> 
> I am trying to see if I can make it work but since I am just a C# windows developer (not a c or c++ dev) what I am doing is just hacking around and don't know if I will even get it to work but I think a c++ linux developer might get it to work with only a little work (the info needed I think is in the link above).
> 
> So I am wondering if anyone is interested to get it to work.
> 
> 
> Yitzchok



From anthony.minessale at gmail.com  Tue Oct 12 07:40:52 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Oct 2010 09:40:52 -0500
Subject: [Freeswitch-users] bad audio of recorded session
In-Reply-To: <AANLkTimx7+mSh8LKsj9A3Qpuewgvwax=Ou4J0qk7y1JL@mail.gmail.com>
References: <AANLkTimx7+mSh8LKsj9A3Qpuewgvwax=Ou4J0qk7y1JL@mail.gmail.com>
Message-ID: <AANLkTimznWj2kXGrmOnfOHQsNXUB4vnX2X6-x7S9_Afj@mail.gmail.com>

It is a known-issue that you never know what you're going to get with
Asterisk.. yes.


On Tue, Oct 12, 2010 at 8:22 AM, Piotr Sobolewski <sobolewski at gmail.com> wrote:
> Hi
>
> When conversation of the call is being saved to a file
> (record_session) audio tempo in such recording is like slowed down
> despite that call conversations lacks this issue.
> Above happens when call comes from asterisk. (Asterisk-->FS-->UA) WRONG
> If the call comes from IP-Phone, recording is normal. ?(UA-->FS-->Asterisk) OK
>
> I've tried multiple options for example:
> ?<action application="export" data="rtp_rewrite_timestamps=true"/>
> ?<action application="set" data="bridge_generate_comfort_noise=true"/>
>
> And setting ?<param name="rtp-timer-name" value="none"/> in the sofia
> profiles didn't solved the problem either.
>
> Global Codecs definition.
>
> ?<X-PRE-PROCESS cmd="set"
> data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU at 20i,PCMA at 20i,GSM"/>
> ?<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU at 20i,PCMA at 20i,GSM"/>
>
> Is this known issue?
>
>
> --
> Piotr Sobolewski
> sobolewski at gmail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From anthony.minessale at gmail.com  Tue Oct 12 07:42:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Oct 2010 09:42:49 -0500
Subject: [Freeswitch-users] Newbie (sort of) questions
In-Reply-To: <1286892835983-5626871.post@n2.nabble.com>
References: <DFB4B88E-6834-4D22-9DEE-616AB50F0D91@stillnewt.org>
	<AANLkTim+zr9DWNQ9QCVccMETYb8Y=OT89t8QhjyfWVv4@mail.gmail.com>
	<4CB38CFD.8040809@mgtech.com> <5069.1286850492@ccs.covici.com>
	<AANLkTik4nj2vxqCMLH=SOZUmS7XfU+Uu4UkQsC9kv9hq@mail.gmail.com>
	<1286892835983-5626871.post@n2.nabble.com>
Message-ID: <AANLkTimiBkgs3KYHKeF4D_wo6kg+PDxGSRZNXQC55zLH@mail.gmail.com>

like CudaTel for instance 


On Tue, Oct 12, 2010 at 9:13 AM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
>
> mercutioviz wrote:
>> The CLI is most definitely not for end users. The CLI is for admins and
>> developers. End users need a purty GUI. FreeSWITCH is like being in the
>> Lego
>> store: all the bricks you can imagine, but you need to put them together.
> +1.
>
> Unfortunately, one will probably never learn the inside-out and the
> capabilities an FS system can offer if GUI gets involved. In this case, one
> is at the mercy of what a GUI can offer which perhaps is no more than 5% of
> the total power of FS. However, for a personal daily usage as a simple PBX
> system that OP wants, this will probably be different. In this case, I
> reckon a small foot-print, i.e. an inexpensive Linux embedded, system that
> consumes no more than 5Watts of electricity can easily and smoothly
> host/operate FS + GUI these days.
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Newbie-sort-of-questions-tp5624290p5626871.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From awais-nazeer at hotmail.com  Tue Oct 12 04:09:38 2010
From: awais-nazeer at hotmail.com (awais nazir)
Date: Tue, 12 Oct 2010 17:09:38 +0600
Subject: [Freeswitch-users] 64 bit CPU and 32 bit OS,
	is it good for freeswitch
Message-ID: <COL110-W230774B749D9D716972D1688540@phx.gbl>


Hi

I am newbie using and appreciating this remarkable software.

I have heard that CENTOS 5.3 is best for freeswitch deployment  but I am using 64 bit CPU and 32 bit OS.

Can it give any problem?

Earliest response will be appreciated.


Thanks and Best Regards.

Awais Nazeer
 		 	   		  
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From riccardo.magliocchetti at gmail.com  Tue Oct 12 05:32:58 2010
From: riccardo.magliocchetti at gmail.com (Riccardo Magliocchetti)
Date: Tue, 12 Oct 2010 14:32:58 +0200
Subject: [Freeswitch-users] can't load python scripts
Message-ID: <4CB4557A.3090809@gmail.com>

Hello,

i'm trying to do an ivr in python but all i get is:

2010-10-12 12:39:56.053054 [DEBUG] mod_python.c:188 Call python script
2010-10-12 12:39:56.054128 [DEBUG] mod_python.c:191 Finished calling 
python script
2010-10-12 12:39:56.054128 [ERR] mod_python.c:200 Error calling python 
script

This script fails, even without the import:

from freeswitch import *

def handler(uuid):
     return

FreeSWITCH is 1.0.6. Any hints?

thanks,
riccardo


From kdjakovic at hotmail.com  Tue Oct 12 06:12:40 2010
From: kdjakovic at hotmail.com (katarina djakovic)
Date: Tue, 12 Oct 2010 06:12:40 -0700
Subject: [Freeswitch-users] ACL and Digest authentication problem
Message-ID: <SNT137-w5155113C52492E566CEC53B3540@phx.gbl>


Dear FreeSwitch users,

we need some help about ACL and Digest authenication.  

This is what we want:

1) We want certain users to be authenticated through ACL (certain IP addresses) including both Register and Invite messages. In other words, we want those users to be granted access to our FS withouth having to authenticate with username and password when registering or calling.
2) On the other hand, if users don't fall into our ACL list (registering/calling from other IP addresses) we want them to authenticate normally throught Digest authentication (username/password).

We tried to configure FS for our needs, but we didn't acomplished what we wanted. Namely, now, for any users that do not belong to the ACL list our FS will reject their registration and will NOT fall back to Digest authentication. In other words, our FS will let all users that fall into ACL list register and call without authenticating --- but all others will be rejected on the attempt to register (debug trace says: sofia_reg.c IP YY.YY.YY.YY Rejected by register acl "domains") and will not let them fall back to Digest authentication.

These are our settings:

    a) acl.conf.xml: 
        <configuration name="acl.conf" description="Network Lists">
          <network-lists>

          <!--
        This will traverse the directory adding all users 
         with the cidr= tag to this ACL, when this ACL matches
        the users variables and params apply as if they 
        digest authenticated.
          -->
          <list name="domains" default="deny">
            <node type="allow" domain="$${domain}"/>
            <node type="allow" domain="XX.XX.XX.XX/32"/>
      
          </list>

          </network-lists>
        </configuration>

b) sip profile:

   <param name="apply-inbound-acl" value="domains"/>
   <param name="apply-register-acl" value="domains"/>
   <param name="auth-calls" value="true"/>

c) users that fall into ACL will have a cidr parameter set aproprietelly <user id="2000" mailbox="2000" cidr="XX.XX.XX.XX/32">

Other users, that we want to be authenticated through Digest authentication will not have anything related to ACL in their user profiles in the Directory.

2) On the other hand, if we remove the <param name="apply-register-acl" value="domains"/> from the sip profile, then users that do not belong to the ACL list will register normally and when calling - their calls (Invite) will fall back to digest authentication (here is the debug: "sofia.c:5847 IP YY.YY.YY.YY Rejected by acl "domains". Falling back to Digest auth.).

That is fine with us - but then we have a different problem, then the users from the ACL list will be asked to register by username/password credentials, i.e. their registration will have to authenticated and that is not what we wanted.


We are mistaging somewhere. Hopefully what I wrote makes sense and maybe someone could help us configure the system to fit our needs.

Many thanks in advance,
Katarina 		 	   		  
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From fernando.berretta at gmail.com  Tue Oct 12 07:37:54 2010
From: fernando.berretta at gmail.com (Fernando Berretta)
Date: Tue, 12 Oct 2010 11:37:54 -0300
Subject: [Freeswitch-users] Originate Calls From an external program
In-Reply-To: <4CAEAD0B.8020704@gmail.com>
References: <4CAEAD0B.8020704@gmail.com>
Message-ID: <4CB472C2.4030706@gmail.com>

  Hi,

I'm newbie in FreeSwitch.

Is there some way to instruct FreeSwitch from an external program in 
order to originate calls to certain numbers an then bridge the calls etc. ?

I'm looking fore something similar to Asterisk AMI  where we we use the 
action Originate to achieve this
Is there any other way to do that in FreeSwitch

Any help will be appreciated.

Best Regards,
Fernando



From mnhassan at usa.net  Tue Oct 12 07:51:09 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Tue, 12 Oct 2010 20:51:09 +0600
Subject: [Freeswitch-users] 64 bit CPU and 32 bit OS,
	is it good for freeswitch
In-Reply-To: <COL110-W230774B749D9D716972D1688540@phx.gbl>
References: <COL110-W230774B749D9D716972D1688540@phx.gbl>
Message-ID: <AANLkTin9=dFMe=R4Ek2qgqE9G7MO_Axbdmn+17QrzeDd@mail.gmail.com>

What is your processor model?

64 bit OS is recommended for
best results.

Regards
HASSAN


On 2010-10-12, awais nazir <awais-nazeer at hotmail.com> wrote:
>
> Hi
>
> I am newbie using and appreciating this remarkable software.
>
> I have heard that CENTOS 5.3 is best for freeswitch deployment  but I am
> using 64 bit CPU and 32 bit OS.
>
> Can it give any problem?
>
> Earliest response will be appreciated.
>
>
> Thanks and Best Regards.
>
> Awais Nazeer
>  		 	   		

-- 
Sent from my mobile device


From anthony.minessale at gmail.com  Tue Oct 12 07:55:10 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Oct 2010 09:55:10 -0500
Subject: [Freeswitch-users] 64 bit CPU and 32 bit OS,
	is it good for freeswitch
In-Reply-To: <COL110-W230774B749D9D716972D1688540@phx.gbl>
References: <COL110-W230774B749D9D716972D1688540@phx.gbl>
Message-ID: <AANLkTi=6Mmk+DnaGvOwKOo7EFoG1J0MBf8OFYqSBMkSP@mail.gmail.com>

Yes its a problem because we don't support it at all.
By support it, I mean won't help you.  We support it technically as in
it will work but when it starts to act strange we will not assist.


On Tue, Oct 12, 2010 at 6:09 AM, awais nazir <awais-nazeer at hotmail.com> wrote:
> Hi
>
> I am newbie using and appreciating this remarkable software.
>
> I have heard that CENTOS 5.3 is best for freeswitch deployment? but I am
> using 64 bit CPU and 32 bit OS.
>
> Can it give any problem?
>
> Earliest response will be appreciated.
>
>
> Thanks and Best Regards.
>
> Awais Nazeer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From david.ponzone at ipeva.fr  Tue Oct 12 07:58:19 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 12 Oct 2010 16:58:19 +0200
Subject: [Freeswitch-users] Originate Calls From an external program
In-Reply-To: <4CB472C2.4030706@gmail.com>
References: <4CAEAD0B.8020704@gmail.com> <4CB472C2.4030706@gmail.com>
Message-ID: <73E1ABB8-57C1-4975-8443-4F9E9C384C03@ipeva.fr>

Check ESL on the wiki.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 12/10/2010 ? 16:37, Fernando Berretta a ?crit :

>  Hi,
>
> I'm newbie in FreeSwitch.
>
> Is there some way to instruct FreeSwitch from an external program in
> order to originate calls to certain numbers an then bridge the calls  
> etc. ?
>
> I'm looking fore something similar to Asterisk AMI  where we we use  
> the
> action Originate to achieve this
> Is there any other way to do that in FreeSwitch
>
> Any help will be appreciated.
>
> Best Regards,
> Fernando
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From sobolewski at gmail.com  Tue Oct 12 07:58:56 2010
From: sobolewski at gmail.com (Piotr Sobolewski)
Date: Tue, 12 Oct 2010 16:58:56 +0200
Subject: [Freeswitch-users] bad audio of recorded session
In-Reply-To: <AANLkTimznWj2kXGrmOnfOHQsNXUB4vnX2X6-x7S9_Afj@mail.gmail.com>
References: <AANLkTimx7+mSh8LKsj9A3Qpuewgvwax=Ou4J0qk7y1JL@mail.gmail.com>
	<AANLkTimznWj2kXGrmOnfOHQsNXUB4vnX2X6-x7S9_Afj@mail.gmail.com>
Message-ID: <AANLkTimK_7dh-KQJOnwujz5js8nt+CpdWAw367XTkXk3@mail.gmail.com>

On Tue, Oct 12, 2010 at 4:40 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> It is a known-issue that you never know what you're going to get with
> Asterisk.. yes.

Is there a way to solve this issue without taking away asterisk?

>
> On Tue, Oct 12, 2010 at 8:22 AM, Piotr Sobolewski <sobolewski at gmail.com> wrote:
>> Hi
>>
>> When conversation of the call is being saved to a file
>> (record_session) audio tempo in such recording is like slowed down
>> despite that call conversations lacks this issue.
>> Above happens when call comes from asterisk. (Asterisk-->FS-->UA) WRONG
>> If the call comes from IP-Phone, recording is normal. ?(UA-->FS-->Asterisk) OK
>>
>> I've tried multiple options for example:
>> ?<action application="export" data="rtp_rewrite_timestamps=true"/>
>> ?<action application="set" data="bridge_generate_comfort_noise=true"/>
>>
>> And setting ?<param name="rtp-timer-name" value="none"/> in the sofia
>> profiles didn't solved the problem either.
>>
>> Global Codecs definition.
>>
>> ?<X-PRE-PROCESS cmd="set"
>> data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU at 20i,PCMA at 20i,GSM"/>
>> ?<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU at 20i,PCMA at 20i,GSM"/>
>>
>> Is this known issue?
>>
>>
>> --
>> Piotr Sobolewski
>> sobolewski at gmail.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Piotr Sobolewski
sobolewski at gmail.com


From mrene_lists at avgs.ca  Tue Oct 12 07:59:48 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 12 Oct 2010 10:59:48 -0400
Subject: [Freeswitch-users] 64 bit CPU and 32 bit OS,
	is it good for freeswitch
In-Reply-To: <COL110-W230774B749D9D716972D1688540@phx.gbl>
References: <COL110-W230774B749D9D716972D1688540@phx.gbl>
Message-ID: <B664E6B3-4844-4B3C-ACB9-7C89027D3F9F@avgs.ca>

It shouldn't pose any direct problems as the CPU will perfectly emulate 32 bits, but be aware that you won't get the same performance as using a complete 64 bits system. I would definitely get a 64 bits OS since your CPU supports it, or else its wasting your money/hardware.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-10-12, at 7:09 AM, awais nazir wrote:

> Hi
> 
> I am newbie using and appreciating this remarkable software.
> 
> I have heard that CENTOS 5.3 is best for freeswitch deployment  but I am using 64 bit CPU and 32 bit OS.
> 
> Can it give any problem?
> 
> Earliest response will be appreciated.
> 
> 
> Thanks and Best Regards.
> 
> Awais Nazeer
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Tue Oct 12 08:01:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Oct 2010 10:01:51 -0500
Subject: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4
In-Reply-To: <AANLkTinRZY_DwDXqEDumMaKuXc39zveHayHXCYZa-=_q@mail.gmail.com>
References: <AANLkTingeYfPT_b6dG3bKQOVjxYsmBTKhDuLw_3BKGac@mail.gmail.com>
	<AANLkTi=YZsrcSWbrnUrZ6+P_fnXpQVAROCNc+9okF0X9@mail.gmail.com>
	<AANLkTinFAifa6BPLDqYVEEh1onCobbgJX4DHC_Kzdn_u@mail.gmail.com>
	<AANLkTinRZY_DwDXqEDumMaKuXc39zveHayHXCYZa-=_q@mail.gmail.com>
Message-ID: <AANLkTi=ectqfWOymqRuPK_urDSZ1q0rPLxA2g8nCtB2v@mail.gmail.com>

Try telnet to localhost port 8021 and see if that connects.






On Mon, Oct 11, 2010 at 10:21 PM, Nyamul Hassan <mnhassan at usa.net> wrote:
> That file lists the port that FS listens on for event socket connections.
>
> Run a "netstat -nlp | grep <port>" to check which IP that port is
> being listened on.
>
> Then you can use command line switches for fs_cli to force it to
> connect on that IP/port.
>
> Regards
> HASSAN
>
>
> On 2010-10-12, Peter Schrock <peter.schrock at gmail.com> wrote:
>> just to make sure what am I comparing event_socket.conf.xml to? I have found
>> that it is set up with the default settings but am not sure how or to what I
>> am suppose to fix. I understand that I am comparing it to FS, but I am
>> wondering if there is some kind of file that I am looking for to compare the
>> two.
>>
>> PeterS
>>
>> On Fri, Oct 8, 2010 at 4:56 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> 1. Check FS is running
>>> 2. Check the port you're connecting to matches the one in
>>> event_socket.conf.xml
>>> 3. Check that event_socket.conf.xml binding to the same IP you're
>>> connecting to (e.g. 0.0.0.0 if you're connecting from a remote
>>> machine, since the default 127.0.0.1 won't work then)
>>> 4. Use netstat on the FS server to verify FreeSWITCH is actually
>>> listening on the port you're trying to connect to (I've had a syntax
>>> error in the config file make the module fail to load in the past
>>> which left FS running but with no ESL socket). (If the module fails to
>>> load there'll also be an error in the log file).
>>> 5. Check a firewall isn't blocking access to the port
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 7 October 2010 01:23, Peter Schrock <peter.schrock at gmail.com> wrote:
>>> > Okay, so I managed to get FS working the other day and I even managed to
>>> > test a call and test my voicemail. All seemed to be working smoothly
>>> until,
>>> > because of the rain here, my power went out and I had to reboot my
>>> computer.
>>> > I logged in through the terminal, set up FS in background went to fs_cli
>>> and
>>> > I get this error message:
>>> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
>>> > Connection
>>> > Error]
>>> > I managed to figure out that in the file "fs_cli.c" on line 1206 you
>>> > find
>>> > the code for displaying this error message. The problem is that I don't
>>> know
>>> > why this error message is occurring. Does anyone have any helpful hints
>>> as
>>> > to what I should look at to resolve this problem?
>>> > I even tried going to the git tree and make current, but that gave me
>>> > problems that forced me to turn off mod_spandsp and mod_skyopen in the
>>> > modules.conf, which I had running earlier. Any thoughts?
>>> > PeterS
>>> > PS
>>> > I am not sure if this is of any help, but in addition to the error line
>>> > above, it also posted this info:
>>> > Usage: fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x
>>> > command] [profile]
>>> > ? -?,-h --help ? ? ? ? ? ? ? ? ? ?Usage Information
>>> > ? -H, --host=hostname ? ? ? ? ? ? Host to connect
>>> > ? -P, --port=port ? ? ? ? ? ? ? ? Port to connect (1 - 65535)
>>> > ? -u, --user=user at domain ? ? ? ? ?user at domain
>>> > ? -p, --password=password ? ? ? ? Password
>>> > ? -x, --execute=command ? ? ? ? ? Execute Command and Exit
>>> > ? -l, --loglevel=command ? ? ? ? ?Log Level
>>> > ? -q, --quiet ? ? ? ? ? ? ? ? ? ? Disable logging
>>> > ? -r, --retry ? ? ? ? ? ? ? ? ? ? Retry connection on failure
>>> > ? -R, --reconnect ? ? ? ? ? ? ? ? Reconnect if disconnected
>>> > ? -d, --debug=level ? ? ? ? ? ? ? Debug Level (0 - 7)
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>
> --
> Sent from my mobile device
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From Holger.Esser at Convergys.com  Tue Oct 12 08:08:11 2010
From: Holger.Esser at Convergys.com (Esser, Holger)
Date: Tue, 12 Oct 2010 10:08:11 -0500
Subject: [Freeswitch-users] Originate Calls From an external program
In-Reply-To: <4CB472C2.4030706@gmail.com>
References: <4CAEAD0B.8020704@gmail.com> <4CB472C2.4030706@gmail.com>
Message-ID: <0FF607C13B7F7A42B5B80DA9EB204C9109D44BD6D2@srv-ex01-dal.intervoice.int>

Fernando,

Take a look at the socket interface.
http://wiki.freeswitch.org/wiki/Event_Socket

It is as close to AMI as you can get, just better ;)

Here is a sample perl script that I found somewhere.
#!/usr/bin/perl -w

# This autodials via the manager API on an Asterisk box.
# VERY useful for fast and easy scanning with asterisk! -natas

use RPC::XML::Client;
use Data::Dumper;
use Net::Telnet;
use warnings;
use FreeSWITCH::Client;
require ESL;





my $command = shift;
my $args = join(" ", @ARGV);

my $con = new ESL::ESLconnection("127.0.0.1", "8021", "ClueCon");




for ($Loop=0; $Loop < 1; $Loop++) {



       my $e = $con->bgapi($command, $args);
       #print $e->getBody();

        select(undef,undef,undef,0.1);  # sleep 250ms


#     select(undef,undef,undef,0.25); # sleep 1/4 second
#     sleep (1.5);
  # They hung up.  Move on.
  print "Placing next call $Loop\n";
}

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fernando Berretta
Sent: Tuesday, October 12, 2010 9:38 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Originate Calls From an external program

  Hi,

I'm newbie in FreeSwitch.

Is there some way to instruct FreeSwitch from an external program in
order to originate calls to certain numbers an then bridge the calls etc. ?

I'm looking fore something similar to Asterisk AMI  where we we use the
action Originate to achieve this
Is there any other way to do that in FreeSwitch

Any help will be appreciated.

Best Regards,
Fernando


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

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From oseslija at gmail.com  Tue Oct 12 08:15:57 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Tue, 12 Oct 2010 17:15:57 +0200
Subject: [Freeswitch-users] ACL and Digest authentication problem
In-Reply-To: <SNT137-w5155113C52492E566CEC53B3540@phx.gbl>
References: <SNT137-w5155113C52492E566CEC53B3540@phx.gbl>
Message-ID: <AANLkTinWQ65Ah_Vy6QDt=GC60g28eukv01M1Yz9BSocO@mail.gmail.com>

Hello Katarina,

I can answer your questions in (I believe) our mother tongue.


On Tue, Oct 12, 2010 at 3:12 PM, katarina djakovic <kdjakovic at hotmail.com>wrote:

>  Dear FreeSwitch users,
>
> we need some help about ACL and Digest authenication.
>
> This is what we want:
>
> 1) We want certain users to be authenticated through ACL (certain IP
> addresses) including both Register and Invite messages. In other words, we
> want those users to be granted access to our FS withouth having to
> authenticate with username and password when registering or calling.
> 2) On the other hand, if users don't fall into our ACL list
> (registering/calling from other IP addresses) we want them to authenticate
> normally throught Digest authentication (username/password).
>
>
2) je FreeSWITCH-ov default konfiguracija.


> We tried to configure FS for our needs, but we didn't acomplished what we
> wanted. Namely, now, for any users that do not belong to the ACL list our FS
> will reject their registration and will NOT fall back to Digest
> authentication. In other words, our FS will let all users that fall into ACL
> list register and call without authenticating --- but all others will be
> rejected on the attempt to register (debug trace says: sofia_reg.c IP
> YY.YY.YY.YY Rejected by register acl "domains") and will not let them fall
> back to Digest authentication.
>
>
Ako se koristi register acl FS ne koristi fallback na Digest. Ovo ne vazi za
INVITE-e gde to radi.



> These are our settings:
>
>     a) acl.conf.xml:
>         <configuration name="acl.conf" description="Network Lists">
>           <network-lists>
>
>           <!--
>         This will traverse the directory adding all users
>          with the cidr= tag to this ACL, when this ACL matches
>         the users variables and params apply as if they
>         digest authenticated.
>           -->
>           <list name="domains" default="deny">
>             <node type="allow" domain="$${domain}"/>
>             <node type="allow" domain="XX.XX.XX.XX/32"/>
>
>           </list>
>
>           </network-lists>
>         </configuration>
>
> b) sip profile:
>
>    <param name="apply-inbound-acl" value="domains"/>
>    <param name="apply-register-acl" value="domains"/>
>    <param name="auth-calls" value="true"/>
>
> c) users that fall into ACL will have a cidr parameter set aproprietelly
> <user id="2000" mailbox="2000" cidr="XX.XX.XX.XX/32">
>
> Other users, that we want to be authenticated through Digest authentication
> will not have anything related to ACL in their user profiles in the
> Directory.
>
> 2) On the other hand, if we remove the <param name="apply-register-acl"
> value="domains"/> from the sip profile, then users that do not belong to the
> ACL list will register normally and when calling - their calls (Invite) will
> fall back to digest authentication (here is the debug: "sofia.c:5847 IP
> YY.YY.YY.YY Rejected by acl "domains". Falling back to Digest auth.).
>
> That is fine with us - but then we have a different problem, then the users
> from the ACL list will be asked to register by username/password
> credentials, i.e. their registration will have to authenticated and that is
> not what we wanted.
>
>
> We are mistaging somewhere. Hopefully what I wrote makes sense and maybe
> someone could help us configure the system to fit our needs.
>
>
Kao sto sam rekao ovo je podrazumevana opcija.



>
>
>
Many thanks in advance,
> Katarina
>
>
Regards,
Ognjen

irc #freeswitch: sekil


> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Tue Oct 12 08:19:26 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Oct 2010 10:19:26 -0500
Subject: [Freeswitch-users] bad audio of recorded session
In-Reply-To: <AANLkTimK_7dh-KQJOnwujz5js8nt+CpdWAw367XTkXk3@mail.gmail.com>
References: <AANLkTimx7+mSh8LKsj9A3Qpuewgvwax=Ou4J0qk7y1JL@mail.gmail.com>
	<AANLkTimznWj2kXGrmOnfOHQsNXUB4vnX2X6-x7S9_Afj@mail.gmail.com>
	<AANLkTimK_7dh-KQJOnwujz5js8nt+CpdWAw367XTkXk3@mail.gmail.com>
Message-ID: <AANLkTi=urA12xEioZ_6LvdR3bkana-Oo=fT1_tgkQLY6@mail.gmail.com>

possibly but I am not quite sure what the issue is caused by.
you say the audio is slower in the recording?
Does it sound normal during the actual conversation?
What format are you recording to?

On Tue, Oct 12, 2010 at 9:58 AM, Piotr Sobolewski <sobolewski at gmail.com> wrote:
> On Tue, Oct 12, 2010 at 4:40 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> It is a known-issue that you never know what you're going to get with
>> Asterisk.. yes.
>
> Is there a way to solve this issue without taking away asterisk?
>
>>
>> On Tue, Oct 12, 2010 at 8:22 AM, Piotr Sobolewski <sobolewski at gmail.com> wrote:
>>> Hi
>>>
>>> When conversation of the call is being saved to a file
>>> (record_session) audio tempo in such recording is like slowed down
>>> despite that call conversations lacks this issue.
>>> Above happens when call comes from asterisk. (Asterisk-->FS-->UA) WRONG
>>> If the call comes from IP-Phone, recording is normal. ?(UA-->FS-->Asterisk) OK
>>>
>>> I've tried multiple options for example:
>>> ?<action application="export" data="rtp_rewrite_timestamps=true"/>
>>> ?<action application="set" data="bridge_generate_comfort_noise=true"/>
>>>
>>> And setting ?<param name="rtp-timer-name" value="none"/> in the sofia
>>> profiles didn't solved the problem either.
>>>
>>> Global Codecs definition.
>>>
>>> ?<X-PRE-PROCESS cmd="set"
>>> data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU at 20i,PCMA at 20i,GSM"/>
>>> ?<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU at 20i,PCMA at 20i,GSM"/>
>>>
>>> Is this known issue?
>>>
>>>
>>> --
>>> Piotr Sobolewski
>>> sobolewski at gmail.com
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Piotr Sobolewski
> sobolewski at gmail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From anthony.minessale at gmail.com  Tue Oct 12 08:19:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Oct 2010 10:19:49 -0500
Subject: [Freeswitch-users] bad audio of recorded session
In-Reply-To: <AANLkTimK_7dh-KQJOnwujz5js8nt+CpdWAw367XTkXk3@mail.gmail.com>
References: <AANLkTimx7+mSh8LKsj9A3Qpuewgvwax=Ou4J0qk7y1JL@mail.gmail.com>
	<AANLkTimznWj2kXGrmOnfOHQsNXUB4vnX2X6-x7S9_Afj@mail.gmail.com>
	<AANLkTimK_7dh-KQJOnwujz5js8nt+CpdWAw367XTkXk3@mail.gmail.com>
Message-ID: <AANLkTimApxh2fwUNiVSD-uhhPdYocQ9BT7StwKuvQEW5@mail.gmail.com>

possibly but I am not quite sure what the issue is caused by.
you say the audio is slower in the recording?
Does it sound normal during the actual conversation?
What format are you recording to?

On Tue, Oct 12, 2010 at 9:58 AM, Piotr Sobolewski <sobolewski at gmail.com> wrote:
> On Tue, Oct 12, 2010 at 4:40 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> It is a known-issue that you never know what you're going to get with
>> Asterisk.. yes.
>
> Is there a way to solve this issue without taking away asterisk?
>
>>
>> On Tue, Oct 12, 2010 at 8:22 AM, Piotr Sobolewski <sobolewski at gmail.com> wrote:
>>> Hi
>>>
>>> When conversation of the call is being saved to a file
>>> (record_session) audio tempo in such recording is like slowed down
>>> despite that call conversations lacks this issue.
>>> Above happens when call comes from asterisk. (Asterisk-->FS-->UA) WRONG
>>> If the call comes from IP-Phone, recording is normal. ?(UA-->FS-->Asterisk) OK
>>>
>>> I've tried multiple options for example:
>>> ?<action application="export" data="rtp_rewrite_timestamps=true"/>
>>> ?<action application="set" data="bridge_generate_comfort_noise=true"/>
>>>
>>> And setting ?<param name="rtp-timer-name" value="none"/> in the sofia
>>> profiles didn't solved the problem either.
>>>
>>> Global Codecs definition.
>>>
>>> ?<X-PRE-PROCESS cmd="set"
>>> data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU at 20i,PCMA at 20i,GSM"/>
>>> ?<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU at 20i,PCMA at 20i,GSM"/>
>>>
>>> Is this known issue?
>>>
>>>
>>> --
>>> Piotr Sobolewski
>>> sobolewski at gmail.com
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Piotr Sobolewski
> sobolewski at gmail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From Holger.Esser at Convergys.com  Tue Oct 12 08:21:04 2010
From: Holger.Esser at Convergys.com (Esser, Holger)
Date: Tue, 12 Oct 2010 10:21:04 -0500
Subject: [Freeswitch-users] Originate Calls From an external program
In-Reply-To: <0FF607C13B7F7A42B5B80DA9EB204C9109D44BD6D2@srv-ex01-dal.intervoice.int>
References: <4CAEAD0B.8020704@gmail.com> <4CB472C2.4030706@gmail.com>
	<0FF607C13B7F7A42B5B80DA9EB204C9109D44BD6D2@srv-ex01-dal.intervoice.int>
Message-ID: <0FF607C13B7F7A42B5B80DA9EB204C9109D44BD6F6@srv-ex01-dal.intervoice.int>

Here is the call to the perl script:
perl http.pl originate \{origination_caller_id_number=xxxxxxxxxx\}sofia/gateway/xxxx/xxxxxxxxxx 5000

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Esser, Holger
Sent: Tuesday, October 12, 2010 10:08 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Originate Calls From an external program

Fernando,

Take a look at the socket interface.
http://wiki.freeswitch.org/wiki/Event_Socket

It is as close to AMI as you can get, just better ;)

Here is a sample perl script that I found somewhere.
#!/usr/bin/perl -w

# This autodials via the manager API on an Asterisk box.
# VERY useful for fast and easy scanning with asterisk! -natas

use RPC::XML::Client;
use Data::Dumper;
use Net::Telnet;
use warnings;
use FreeSWITCH::Client;
require ESL;





my $command = shift;
my $args = join(" ", @ARGV);

my $con = new ESL::ESLconnection("127.0.0.1", "8021", "ClueCon");




for ($Loop=0; $Loop < 1; $Loop++) {



       my $e = $con->bgapi($command, $args);
       #print $e->getBody();

        select(undef,undef,undef,0.1);  # sleep 250ms


#     select(undef,undef,undef,0.25); # sleep 1/4 second
#     sleep (1.5);
  # They hung up.  Move on.
  print "Placing next call $Loop\n";
}

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fernando Berretta
Sent: Tuesday, October 12, 2010 9:38 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Originate Calls From an external program

  Hi,

I'm newbie in FreeSwitch.

Is there some way to instruct FreeSwitch from an external program in
order to originate calls to certain numbers an then bridge the calls etc. ?

I'm looking fore something similar to Asterisk AMI  where we we use the
action Originate to achieve this
Is there any other way to do that in FreeSwitch

Any help will be appreciated.

Best Regards,
Fernando


_______________________________________________
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This e-mail transmission may contain information that is proprietary, privileged and/or confidential and is intended exclusively for the person(s) to whom it is addressed. Any use, copying, retention or disclosure by any person other than the intended recipient or the intended recipient's designees is strictly prohibited. If you are the intended recipient, you must treat the information in confidence and in accordance with all laws related to the privacy and confidentiality of such information.  If you are not the intended recipient or their designee, please notify the sender immediately by return e-mail and delete all copies of this email, including all attachments.

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No virus found in this incoming message.
Checked by AVG - www.avg.com 
Version: 9.0.862 / Virus Database: 271.1.1/3189 - Release Date: 10/12/10 01:34:00

No virus found in this outgoing message.
Checked by AVG - www.avg.com 
Version: 9.0.862 / Virus Database: 271.1.1/3189 - Release Date: 10/12/10 01:34:00


From sobolewski at gmail.com  Tue Oct 12 08:45:48 2010
From: sobolewski at gmail.com (Piotr Sobolewski)
Date: Tue, 12 Oct 2010 17:45:48 +0200
Subject: [Freeswitch-users] bad audio of recorded session
In-Reply-To: <AANLkTi=urA12xEioZ_6LvdR3bkana-Oo=fT1_tgkQLY6@mail.gmail.com>
References: <AANLkTimx7+mSh8LKsj9A3Qpuewgvwax=Ou4J0qk7y1JL@mail.gmail.com>
	<AANLkTimznWj2kXGrmOnfOHQsNXUB4vnX2X6-x7S9_Afj@mail.gmail.com>
	<AANLkTimK_7dh-KQJOnwujz5js8nt+CpdWAw367XTkXk3@mail.gmail.com>
	<AANLkTi=urA12xEioZ_6LvdR3bkana-Oo=fT1_tgkQLY6@mail.gmail.com>
Message-ID: <AANLkTik8c134MF3KnDF0RCK1ir1YsSN1Sd3+8GRia_ZW@mail.gmail.com>

On Tue, Oct 12, 2010 at 5:19 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> possibly but I am not quite sure what the issue is caused by.
> you say the audio is slower in the recording?

yes, it's slower to such extent that it's unreadable

> Does it sound normal during the actual conversation?

yes, it's normal

> What format are you recording to?

wav



-- 
Piotr Sobolewski
sobolewski at gmail.com


From sobolewski at gmail.com  Tue Oct 12 08:45:48 2010
From: sobolewski at gmail.com (Piotr Sobolewski)
Date: Tue, 12 Oct 2010 17:45:48 +0200
Subject: [Freeswitch-users] bad audio of recorded session
In-Reply-To: <AANLkTi=urA12xEioZ_6LvdR3bkana-Oo=fT1_tgkQLY6@mail.gmail.com>
References: <AANLkTimx7+mSh8LKsj9A3Qpuewgvwax=Ou4J0qk7y1JL@mail.gmail.com>
	<AANLkTimznWj2kXGrmOnfOHQsNXUB4vnX2X6-x7S9_Afj@mail.gmail.com>
	<AANLkTimK_7dh-KQJOnwujz5js8nt+CpdWAw367XTkXk3@mail.gmail.com>
	<AANLkTi=urA12xEioZ_6LvdR3bkana-Oo=fT1_tgkQLY6@mail.gmail.com>
Message-ID: <AANLkTik8c134MF3KnDF0RCK1ir1YsSN1Sd3+8GRia_ZW@mail.gmail.com>

On Tue, Oct 12, 2010 at 5:19 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> possibly but I am not quite sure what the issue is caused by.
> you say the audio is slower in the recording?

yes, it's slower to such extent that it's unreadable

> Does it sound normal during the actual conversation?

yes, it's normal

> What format are you recording to?

wav



-- 
Piotr Sobolewski
sobolewski at gmail.com


From sobolewski at gmail.com  Tue Oct 12 08:45:48 2010
From: sobolewski at gmail.com (Piotr Sobolewski)
Date: Tue, 12 Oct 2010 17:45:48 +0200
Subject: [Freeswitch-users] bad audio of recorded session
In-Reply-To: <AANLkTi=urA12xEioZ_6LvdR3bkana-Oo=fT1_tgkQLY6@mail.gmail.com>
References: <AANLkTimx7+mSh8LKsj9A3Qpuewgvwax=Ou4J0qk7y1JL@mail.gmail.com>
	<AANLkTimznWj2kXGrmOnfOHQsNXUB4vnX2X6-x7S9_Afj@mail.gmail.com>
	<AANLkTimK_7dh-KQJOnwujz5js8nt+CpdWAw367XTkXk3@mail.gmail.com>
	<AANLkTi=urA12xEioZ_6LvdR3bkana-Oo=fT1_tgkQLY6@mail.gmail.com>
Message-ID: <AANLkTik8c134MF3KnDF0RCK1ir1YsSN1Sd3+8GRia_ZW@mail.gmail.com>

On Tue, Oct 12, 2010 at 5:19 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> possibly but I am not quite sure what the issue is caused by.
> you say the audio is slower in the recording?

yes, it's slower to such extent that it's unreadable

> Does it sound normal during the actual conversation?

yes, it's normal

> What format are you recording to?

wav



-- 
Piotr Sobolewski
sobolewski at gmail.com


From anthony.minessale at gmail.com  Tue Oct 12 08:54:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Oct 2010 10:54:38 -0500
Subject: [Freeswitch-users] bad audio of recorded session
In-Reply-To: <AANLkTik8c134MF3KnDF0RCK1ir1YsSN1Sd3+8GRia_ZW@mail.gmail.com>
References: <AANLkTimx7+mSh8LKsj9A3Qpuewgvwax=Ou4J0qk7y1JL@mail.gmail.com>
	<AANLkTimznWj2kXGrmOnfOHQsNXUB4vnX2X6-x7S9_Afj@mail.gmail.com>
	<AANLkTimK_7dh-KQJOnwujz5js8nt+CpdWAw367XTkXk3@mail.gmail.com>
	<AANLkTi=urA12xEioZ_6LvdR3bkana-Oo=fT1_tgkQLY6@mail.gmail.com>
	<AANLkTik8c134MF3KnDF0RCK1ir1YsSN1Sd3+8GRia_ZW@mail.gmail.com>
Message-ID: <AANLkTi=stRpj1HxL63QVgHtGZPdQH9MXgwaBGti6mHFm@mail.gmail.com>

maybe you can produce a sample recording

On Tue, Oct 12, 2010 at 10:45 AM, Piotr Sobolewski <sobolewski at gmail.com> wrote:
> On Tue, Oct 12, 2010 at 5:19 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> possibly but I am not quite sure what the issue is caused by.
>> you say the audio is slower in the recording?
>
> yes, it's slower to such extent that it's unreadable
>
>> Does it sound normal during the actual conversation?
>
> yes, it's normal
>
>> What format are you recording to?
>
> wav
>
>
>
> --
> Piotr Sobolewski
> sobolewski at gmail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From babak.freeswitch at gmail.com  Tue Oct 12 09:01:35 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Tue, 12 Oct 2010 19:31:35 +0330
Subject: [Freeswitch-users] mod_managed socket binding problem
Message-ID: <AANLkTinqR-8GTQ7qAOM=u5TFEoDbX__qAt7p8V++D06c@mail.gmail.com>

Hi
I'm creating a module in mod_managed to accept snmp request and respond by
returning the result of calling some apis. I'm creating a udp socket to
listen for snmp reqs. problem is whenever the module is reloaded, because
first a new load occurs, in the new one the socket can not be bind, because
of the previous socket still running. how can I stop the old socket?
thanx
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From sobolewski at gmail.com  Tue Oct 12 09:02:41 2010
From: sobolewski at gmail.com (Piotr Sobolewski)
Date: Tue, 12 Oct 2010 18:02:41 +0200
Subject: [Freeswitch-users] bad audio of recorded session
In-Reply-To: <AANLkTi=urA12xEioZ_6LvdR3bkana-Oo=fT1_tgkQLY6@mail.gmail.com>
References: <AANLkTimx7+mSh8LKsj9A3Qpuewgvwax=Ou4J0qk7y1JL@mail.gmail.com>
	<AANLkTimznWj2kXGrmOnfOHQsNXUB4vnX2X6-x7S9_Afj@mail.gmail.com>
	<AANLkTimK_7dh-KQJOnwujz5js8nt+CpdWAw367XTkXk3@mail.gmail.com>
	<AANLkTi=urA12xEioZ_6LvdR3bkana-Oo=fT1_tgkQLY6@mail.gmail.com>
Message-ID: <AANLkTik_PCebN+Sy-9vtoFJGDH2NgG7dYeiDT9OXMrEY@mail.gmail.com>

On Tue, Oct 12, 2010 at 5:19 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> possibly but I am not quite sure what the issue is caused by.
> you say the audio is slower in the recording?

yes, it's slower to such extent that it's unreadable

> Does it sound normal during the actual conversation?

yes, it's normal

> What format are you recording to?

wav



-- 
Piotr Sobolewski
sobolewski at gmail.com


From brian at freeswitch.org  Tue Oct 12 09:16:28 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Oct 2010 11:16:28 -0500
Subject: [Freeswitch-users] mod_managed socket binding problem
In-Reply-To: <AANLkTinqR-8GTQ7qAOM=u5TFEoDbX__qAt7p8V++D06c@mail.gmail.com>
References: <AANLkTinqR-8GTQ7qAOM=u5TFEoDbX__qAt7p8V++D06c@mail.gmail.com>
Message-ID: <D51C1FE5-C34E-4B04-95AF-1D84B6AAFCA0@freeswitch.org>

You would have to shutdown and close the socket in your module shutdown routine.

/b

On Oct 12, 2010, at 11:01 AM, babak yakhchali wrote:

> Hi
> I'm creating a module in mod_managed to accept snmp request and respond by returning the result of calling some apis. I'm creating a udp socket to listen for snmp reqs. problem is whenever the module is reloaded, because first a new load occurs, in the new one the socket can not be bind, because of the previous socket still running. how can I stop the old socket?
> thanx
> _________



From neil.burgess at redmatter.com  Tue Oct 12 09:24:52 2010
From: neil.burgess at redmatter.com (Neil Burgess)
Date: Tue, 12 Oct 2010 17:24:52 +0100
Subject: [Freeswitch-users] Click to Dial using REFER
Message-ID: <787302A89ACCE24DA8F56DA101E77C842B52774835@THHS2E12BE1X.hostedservice2.net>

Hi,

We are looking at how to implement some "Click to Dial" functionality, and have come across a "REFER based" method documented on several sites, e.g. http://www.tech-invite.com/Ti-sip-service-19.html, which shows the possibility of a Switch or some other agent sending a SIP REFER to a Client which causes that client to make an outbound call to the "referred to" URI.

This is also documented elsewhere as:-

"First, the SIP server sends an INVITE to one of the phones because phones usually do not accept
REFER without prior invitation.The INVITE contains 0.0.0.0 as the IP address in SDP, because
there is no remote phone (the message is sent by user agent within the SIP server which does not
deal with media).
After that, the server sends a REFER method which will ask the phone to send INVITE
somewhere else.The URI of the called party is passed to the phone in a Refer-To header field of
the REFER method."

My question relates to whether or not it is possible to cause FS to issue this form of REFER via the Event Socket interface to implement this kind of click to dial functionality.

Rgds,
Neil

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From brian at freeswitch.org  Tue Oct 12 09:30:14 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Oct 2010 11:30:14 -0500
Subject: [Freeswitch-users] Click to Dial using REFER
In-Reply-To: <787302A89ACCE24DA8F56DA101E77C842B52774835@THHS2E12BE1X.hostedservice2.net>
References: <787302A89ACCE24DA8F56DA101E77C842B52774835@THHS2E12BE1X.hostedservice2.net>
Message-ID: <A22ED745-F05D-42D5-8DFD-1E97C2BCBBC2@freeswitch.org>

Why would you go round about to do this type of click to call when a simple example of doing this is in scripts/perl/call.cgi that uses ESL to do the entire call.

/b

On Oct 12, 2010, at 11:24 AM, Neil Burgess wrote:

> My question relates to whether or not it is possible to cause FS to issue this form of REFER via the Event Socket interface to implement this kind of click to dial functionality.
>  
> Rgds,
> Neil



From anthony.minessale at gmail.com  Tue Oct 12 09:36:12 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Oct 2010 11:36:12 -0500
Subject: [Freeswitch-users] Click to Dial using REFER
In-Reply-To: <787302A89ACCE24DA8F56DA101E77C842B52774835@THHS2E12BE1X.hostedservice2.net>
References: <787302A89ACCE24DA8F56DA101E77C842B52774835@THHS2E12BE1X.hostedservice2.net>
Message-ID: <AANLkTi=h18_HHZCRWJBSSVG=ScRp2Lbm5fjDe4RErMPn@mail.gmail.com>

it's probably possible but it would require additional features coded
into mod_sofia.


On Tue, Oct 12, 2010 at 11:24 AM, Neil Burgess
<neil.burgess at redmatter.com> wrote:
> Hi,
>
>
>
> We are looking at how to implement some ?Click to Dial? functionality, and
> have come across a ?REFER based? method documented on several sites, e.g.
> http://www.tech-invite.com/Ti-sip-service-19.html, which shows the
> possibility of a Switch or some other agent sending a SIP REFER to a Client
> which causes that client to make an outbound call to the ?referred to? URI.
>
>
>
> This is also documented elsewhere as:-
>
>
>
> ?First, the SIP server sends an INVITE to one of the phones because phones
> usually do not accept
>
> REFER without prior invitation.The INVITE contains 0.0.0.0 as the IP address
> in SDP, because
>
> there is no remote phone (the message is sent by user agent within the SIP
> server which does not
>
> deal with media).
>
> After that, the server sends a REFER method which will ask the phone to send
> INVITE
>
> somewhere else.The URI of the called party is passed to the phone in a
> Refer-To header field of
>
> the REFER method.?
>
>
>
> My question relates to whether or not it is possible to cause FS to issue
> this form of REFER via the Event Socket interface to implement this kind of
> click to dial functionality.
>
>
>
> Rgds,
>
> Neil
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From neil.burgess at redmatter.com  Tue Oct 12 10:07:32 2010
From: neil.burgess at redmatter.com (Neil Burgess)
Date: Tue, 12 Oct 2010 18:07:32 +0100
Subject: [Freeswitch-users] Click to Dial using REFER
In-Reply-To: <A22ED745-F05D-42D5-8DFD-1E97C2BCBBC2@freeswitch.org>
References: <787302A89ACCE24DA8F56DA101E77C842B52774835@THHS2E12BE1X.hostedservice2.net>
	<A22ED745-F05D-42D5-8DFD-1E97C2BCBBC2@freeswitch.org>
Message-ID: <787302A89ACCE24DA8F56DA101E77C842B52774867@THHS2E12BE1X.hostedservice2.net>

I guess I was wondering whether the refer method might give a better overall feel in that the phone will presumably act as though it is making an outbound call, rather than it ringing as an inbound call, which then requires it to be manually answered, before it is then bridged to other end! Although, I guess if I also use a Call-Info header, I can get an auto-answer out of it.  I will go and experiment with this approach for the moment! Thanks for the pointer.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: 12 October 2010 17:30
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Click to Dial using REFER


Why would you go round about to do this type of click to call when a simple example of doing this is in scripts/perl/call.cgi that uses ESL to do the entire call.

/b

On Oct 12, 2010, at 11:24 AM, Neil Burgess wrote:

> My question relates to whether or not it is possible to cause FS to issue this form of REFER via the Event Socket interface to implement this kind of click to dial functionality.
>
> Rgds,
> Neil


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

________________________________

No virus found in this message.
Checked by AVG - www.avg.com<http://www.avg.com>
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From adminjew at gmail.com  Tue Oct 12 10:16:38 2010
From: adminjew at gmail.com (Yitzchok)
Date: Tue, 12 Oct 2010 13:16:38 -0400
Subject: [Freeswitch-users] Mono 2.8 released and need to update
	mod_managed
In-Reply-To: <031F577B-CBFB-44B9-9764-68B4745657B4@freeswitch.org>
References: <AANLkTin3rM+NaW1_O2d4HaDL9yuESGmZvd8v29Kxh6Y8@mail.gmail.com>
	<031F577B-CBFB-44B9-9764-68B4745657B4@freeswitch.org>
Message-ID: <AANLkTik=zPcdCNg+YBzkH4q_zvyZ853abKXrru2AgYOz@mail.gmail.com>

Here it is.

http://jira.freeswitch.org/browse/FS-2774

I would be a good idea if someone has a look through the code since I am not
a c(++) developer.
 <http://jira.freeswitch.org/browse/FS-2774>
Yitzchok


On Tue, Oct 12, 2010 at 10:30 AM, Brian West <brian at freeswitch.org> wrote:

> Patches belong on http://jira.freeswitch.org if you can please.
>
> /b
>
> On Oct 6, 2010, at 8:00 PM, Yitzchok wrote:
>
> > Mono 2.8 was released today with support for C# 4 and more
> http://www.mono-project.com/Release_Notes_Mono_2.8
> >
> > It seems like there was some changes in the way applications embed mono
> (which includes removing the reference to glib and replacing it with eglib
> from what I understand) in this link you can find information for the
> changes made with embedding mono
> http://www.mono-project.com/Embedding_Mono
> >
> > I am trying to see if I can make it work but since I am just a C# windows
> developer (not a c or c++ dev) what I am doing is just hacking around and
> don't know if I will even get it to work but I think a c++ linux developer
> might get it to work with only a little work (the info needed I think is in
> the link above).
> >
> > So I am wondering if anyone is interested to get it to work.
> >
> >
> > Yitzchok
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Joshua.Foshee at LogixCom.com  Tue Oct 12 10:27:59 2010
From: Joshua.Foshee at LogixCom.com (Joshua Foshee)
Date: Tue, 12 Oct 2010 12:27:59 -0500
Subject: [Freeswitch-users] SIP Registration DNS Error
Message-ID: <06502C073AD9394AADB3CA7FD94931BC0519DF46@okc1x1.Logixcom.com>

I have setup up two sip providers that I can connect to fine but after a
while I then start to get these messages.

 

2010-10-12 07:27:57.292742 [NOTICE] sofia_reg.c:342 Registering
flowroute

2010-10-12 07:27:57.295138 [ERR] sofia_reg.c:1611 flowroute Registration
Failed with status DNS Error [503]. failure #1391

2010-10-12 07:27:59.959022 [WARNING] sofia_reg.c:387 flowroute Failed
Registration, setting retry to 30 seconds.

 

Here is the output of the gateway status

 

Name            flowroute

Profile         external

Scheme          Digest

Realm           sip.flowroute.com

Username        xxxxxxx

Password        yes

>From            <sip:xxxxxxxxx at sip.flowroute.com;transport=udp>

Contact
<sip:gw+flowroute at xxxxxxxx:5080;transport=udp;gw=flowroute>

Exten           xxxxxxx

To              sip:xxxxxxxxx at sip.flowroute.com

Proxy           sip:sip.flowroute.com

Context         public

Expires         600

Freq            600

Ping            1286835829

PingFreq        25

PingState       -1/0/1

State           FAIL_WAIT

Status          DOWN

CallsIN         0

CallsOUT        5

FailedCallsIN   0

FailedCallsOUT  5

 

 

Name            broadvoice

Profile         external

Scheme          Digest

Realm           BroadWorks

Username        xxxxxxxx

Password        yes

>From            <sip:xxxxxxxxxxxx at sip.broadvoice.com;transport=udp>

Contact
<sip:gw+broadvoice at xxxxxxxxxx:5080;transport=udp;gw=broadvoice>

Exten           xxxxxxxx

To              sip:xxxxxxxxxx at sip.broadvoice.com

Proxy           sip:sip.broadvoice.com

Context         public

Expires         30

Freq            30

Ping            0

PingFreq        0

PingState       0/0/0

State           FAIL_WAIT

Status          DOWN

CallsIN         0

CallsOUT        2

FailedCallsIN   0

FailedCallsOUT  3

 

If I restart Freeswitch process they both come up and Reg just fine for
a while till it fails again.

 

Thanks in advance,

Josh

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From anthony.minessale at gmail.com  Tue Oct 12 10:33:52 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Oct 2010 12:33:52 -0500
Subject: [Freeswitch-users] tone_detect vs. ignore_early_media
	ANDmod_fifo
In-Reply-To: <AANLkTimea0_Gs6jcc2ic=pYFaxFhKPAErgPnRbQnZva0@mail.gmail.com>
References: <429823D0454A4EB4AE179DF13803F9B9@mbnet.local>
	<AANLkTim1B_oxbCAXNPTHNoZxh6vUfbUiWsyHzxG=xwif@mail.gmail.com>
	<DAFF983A4D6A43A186D8662FE5876375@mbnet.local>
	<515E6743DC69484A8E3514E807B24353@mbnet.local>
	<D33D5C7C9BAE42CA96354582D25F8B21@mbnet.local>
	<AANLkTimea0_Gs6jcc2ic=pYFaxFhKPAErgPnRbQnZva0@mail.gmail.com>
Message-ID: <AANLkTinun0CHg5+yjEkZuC=jFNMQUUvqhhqqA_GqWG0L@mail.gmail.com>

try latest GIT

On Mon, Oct 11, 2010 at 4:00 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> This was done to stop recordings and other things on callers waiting in a queue.
> We may have to change it to be configurable.
>
>
> On Mon, Oct 11, 2010 at 6:51 AM, Jeroen C. van Gelderen
> <slim at thegreek.com> wrote:
>>
>>
>> I have an inkling that the following commit made between 1.0.2 and 1.0.3
>> might have something to do with this:
>>
>>
>>
>> ?? * mod_fifo: pause media bugs while not in a bridge (r:11466,11490)
>>
>>
>>
>> http://lists.freeswitch.org/pipermail/freeswitch-svn/2009-January/009566.html
>>
>> http://lists.freeswitch.org/pipermail/freeswitch-svn/2009-January/009589.html
>>
>>
>>
>> Can anyone comment on the how and why? Is there any way to reconcile this
>> with use of tone_detect?
>>
>> Cheers,
>> -Slim
>> --
>> Jeroen C. "Slim" van Gelderen
>> Olympic Sports Data Services
>> Email: slim at thegreek.com
>> Phone: +1 876 953 6182 x128
>>
>> ________________________________
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeroen
>> C. van Gelderen
>> Sent: Monday, October 11, 2010 05:31
>> To: 'FreeSWITCH Users Help'
>> Subject: Re: [Freeswitch-users] tone_detect vs. ignore_early_media
>> ANDmod_fifo
>>
>>
>>
>>
>>
>> Hmm? the plot thickens?
>>
>>
>>
>> When I bridge my FXO port to the SIP-GSM gateway directly (i.e. without
>> using mod_fifo) I don?t seem to need monitor_early_media_fail. Using
>> tone_detect on the A leg works fine when ?ignore_early_media=true? is used
>> on the B leg:
>>
>>
>>
>> The following dialplan excerpt WORKS (i.e. FXO hang-ups are detected at all
>> stages by tone_detect):
>>
>>
>>
>> ??? <extension name="pan_125">
>>
>> ????? <condition field="${freetdm_span_name}"
>> expression="^span_fxo_helpdesk$">
>>
>> ??????? <!-- Detect busy tone from Panasonic -->
>>
>> ??????? <action application="tone_detect" data="busy-pan 350,440 r 0 hangup
>> undef 2"/>
>>
>>
>>
>> ??????? <action application="answer"/>
>>
>>
>>
>> ??????? <action application="set" data="continue_on_fail=true"/>
>>
>> ??????? <action application="set" data="instant_ringback=true"/>
>>
>> ??????? <action application="set" data="ringback=%(1000,1500,440.0,480.0)"/>
>>
>>
>>
>> ??????? <action application="bridge"
>> data="{ignore_early_media=true}sofia/internal/1??????76??@192.168.3.11:5060"/>
>>
>> ??????? <action application="bridge"
>> data="{ignore_early_media=true}sofia/internal/1??????77??@192.168.3.11:5060"/>
>>
>> ????? </condition>
>>
>> ??? </extension>
>>
>>
>>
>> The problem seems to occur only when mod_fifo is added to the mix:
>>
>>
>>
>> ??? <extension name="test2">
>>
>> ????? <condition field="${freetdm_span_name}" expression="^span_fxo_test$">
>>
>> ??????? <!-- Detect busy tone from Panasonic -->
>>
>> ??????? <action application="tone_detect" data="busy-pan 350,440 r 0 hangup
>> undef 2"/>
>>
>>
>>
>> ??????? <action application="answer"/>
>>
>>
>>
>> ??????? <action application="playback" data="thankyou.wav"/>
>>
>> ??????? <action application="set" data="fifo_music=$${hold_music}"/>
>>
>> ??????? <action application="fifo" data="fifo_helpdesk in"/>
>>
>> ????? </condition>
>>
>> ??? </extension>
>>
>>
>>
>> In all cases:
>>
>> -????????? tone_detect works BEFORE the call is handed to mod_fifo (i.e.
>> during playback)
>>
>> -????????? tone_detect works AFTER the call is established by mod_fifo and
>> audio is being exchanged between A and B leg.
>>
>>
>>
>> IF the A leg (with tone_detect enabled) is handed to mod_fifo AND there are
>> no agents in the fifo THEN tone_detect does not work. Mod_fifo simply plays
>> MOH to the A leg perpetually.
>>
>>
>>
>> IF the A leg (with tone_detect enabled) is handed to mod_fifo AND mod_fifo
>> places an outbound call with ?ignore_early_media=true? THEN tone_detect does
>> not work on the A leg during the early media phase on leg B.
>>
>>
>>
>> IF the A leg (with tone_detect enabled) is handed to mod_fifo AND mod_fifo
>> places an outbound call with ?ignore_early_media=false? THEN tone_detect
>> does work. I guess this is because audio is being exchanged between A and B
>> legs.
>>
>>
>>
>> Any idea what would cause the tone_detect to be ?suspended? when mod_fifo is
>> in the mix?
>>
>> Cheers,
>> -Slim
>> --
>> Jeroen C. "Slim" van Gelderen
>>
>> ________________________________
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeroen
>> C. van Gelderen
>> Sent: Monday, October 11, 2010 03:13
>> To: 'FreeSWITCH Users Help'
>> Subject: Re: [Freeswitch-users] tone_detect vs. ignore_early_media (per
>> leg?)
>>
>>
>>
>> Thank you for the quick response.
>>
>>
>>
>> It looks like monitor_early_media_fail should do what I need (thanks for the
>> suggestion!) but I can?t seem to make it work. See below for my uneducated
>> best guess why.
>>
>>
>>
>> In my case the failure conditions from the GSM side are handled out-of-band
>> by SIP. That leaves only one failure condition I need to listen for in early
>> media (the Panasonic far-end hang up on FXO) which can successfully be
>> detected with:
>>
>>
>>
>> ? <action application="tone_detect" data="busy-pan 350,440 r 0 hangup>
>>
>>
>>
>> This results in the following log entries:
>>
>>
>>
>> [DEBUG] switch_ivr_async.c:2072 Adding tone spec 350,440 index 0 hits 1
>>
>> [DEBUG] switch_core_media_bug.c:360 Attaching BUG to FreeTDM/4:1/
>>
>>
>>
>> Translating tone_detect to monitor_early_media_fail I tried the following:
>>
>>
>>
>> ? <action application="set" data="result=${fifo_member(add fifo_helpdesk
>> {fifo_member_wait=nowait,ignore_early_media=true,monitor_early_media_fail=busy-pan:1:350+440}sofia/internal/${caller_id_number}@192.168.3.11:5060)}"/>
>>
>>
>>
>> This results in the following relevant log entries:
>>
>>
>>
>> [DEBUG] switch_channel.c:2444 (sofia/internal/1876??????@192.168.3.11:5060)
>> Callstate Change RINGING -> EARLY
>>
>> [DEBUG] switch_ivr_async.c:2072 Adding tone spec 350,440 index 0 hits 1
>>
>> [DEBUG] switch_core_media_bug.c:360 Attaching BUG to
>> sofia/internal/1??????7693 at 192.168.3.11:5060
>>
>>
>>
>> So Freeswitch is listening for the right tones (tone_spec present and
>> identical to the one used in tone_detect approach) but it isn?t detecting
>> them. The obvious difference is in the BUG attachment.
>>
>>
>>
>> Is it possible that BUG isn?t listening to the right (A) leg in the case of
>> monitor_early_media_fail? Or is this too easy? J
>>
>> Cheers,
>> -Slim
>> --
>> Jeroen C. "Slim" van Gelderen
>>
>> ________________________________
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
>> Collins
>> Sent: Sunday, October 10, 2010 19:01
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] tone_detect vs. ignore_early_media (per
>> leg?)
>>
>>
>>
>> Check out monitor_early_media_fail:
>> http://wiki.freeswitch.org/wiki/Channel_Variables#monitor_early_media_fail
>>
>> It is a hybrid of ignoring early media and listening to early media for
>> various conditions. In your case you'll need to play around with it. In your
>> case you need to figure out which early media scenarios count as a "fail"
>> and will cause processing to move on as if the call really did fail.
>>
>> Roll up your sleeves, you have some work to do. :)
>>
>> -MC
>>
>> On Sat, Oct 9, 2010 at 9:24 PM, Jeroen C. van Gelderen <slim at thegreek.com>
>> wrote:
>>
>> Hi Guys,
>>
>> I have a legacy Panasonic PBX which does not support CPC/Disconnect
>> Supervision. Calls from this PBX are sent to Freeswitch by way of
>> DAHDI/FreeTDM FXO ports (on a Xorcom Astribank). I use tone_detect to deal
>> with far-end hang-ups. This works fine:
>>
>> <!-- Detect busy tone from Panasonic -->
>> <action application="tone_detect" data="busy 350,440 r 0 hangup 4"/>
>>
>> Some calls from the Panasonic PBX are put in a FIFO and from there they are
>> sent to a SIP-to-GSM bridge (GoIP) to be terminated on one of 3 GSM phones
>> in a round-robin fashion.
>>
>> The GSM bridge requires me to use "ignore_early_media=true" otherwise the
>> caller will receive messages like "the number you are calling does not
>> answer". When I set "ignore_early_media=true" the FIFO correctly keeps
>> hunting for a GSM phone that is actually answered and will ignore phones
>> that are busy, no-answer or turned off. This too works fine.
>>
>> The problem occurs when the two are combined as follows:
>>
>> Panasonic PBX --FXO--> Freeswitch --SIP--> GSM-bridge
>>
>> If I enable ignore_early_media then tone_detect doesn't work UNTIL one of
>> the GSMs is answered. This is a problem when none of the GSMs are answered
>> and the Panasonic PBX hangs up. The FXO port will stay "up" and the FIFO
>> will keep hunting until it times out.
>>
>> If I don't use "ignore_early_media=true" then tone_detect works fine but
>> then every telco message gets mistaken for an answered call and the hunting
>> stops early.
>>
>> I tried changing this example line from my fifo.conf:
>>
>> <member timeout="30" simo="1" lag="5">
>> {member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7701 at 192.1
>> 68.3.11:5060
>> </member>
>>
>> to
>>
>> <member timeout="30" simo="1" lag="5">
>> {member_wait=nowait}[ignore_early_media=true]sofia/internal/1??????7701 at 192.
>> 168.3.11:5060
>> </member>
>>
>> in a vain attempt to ignore early media on the SIP leg only. This doesn't
>> seem to do anything however.
>>
>> Can anyone clue me in on what I'm missing? I've snipped the relevant
>> configuration bits below. I have the feeling I'm missing something obvious.
>>
>> Cheers,
>> -Slim
>>
>> ----8<----8<----8<----8<----8<----
>>
>> <configuration name="fifo.conf" description="FIFO Configuration">
>> ?<settings>
>> ? ?<param name="delete-all-outbound-member-on-startup" value="false"/>
>> ?</settings>
>> ?<fifos>
>> ? ?<fifo name="cool_fifo@$${domain}" importance="0">
>>
>> ? ? ?<member timeout="30" simo="1" lag="5">
>> {member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7701 at 192.1
>> 68.3.11:5060</member>
>>
>> ? ? ?<member timeout="30" simo="1" lag="5">
>> {member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7693 at 192.1
>> 68.3.11:5060</member>
>>
>> ? ? ?<member timeout="30" simo="1" lag="5">
>> {member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7698 at 192.1
>> 68.3.11:5060</member>
>>
>> ? ?</fifo>
>> ?</fifos>
>> </configuration>
>>
>> ----8<----8<----8<----8<----8<----
>>
>> <include>
>> ?<context name="pbx_test">
>>
>> ? ?<extension name="test1">
>> ? ? ?<condition field="${freetdm_span_name}" expression="^span_fxo_test$">
>> ? ? ? ?<action application="answer"/>
>>
>> ? ? ? ?<!-- Detect busy tone from Panasonic -->
>> ? ? ? ?<action application="tone_detect" data="busy 350,440 r 0 hangup 4"/>
>>
>> ? ? ? ?<action application="playback"
>> data="ivr/ivr-hold_connect_call.wav"/>
>>
>> ? ? ? ?<action application="fifo" data="cool_fifo@$${domain} in"/>
>>
>> ? ? ?</condition>
>> ? ?</extension>
>> ?</context>
>> </include>
>>
>> ----8<----8<----8<----8<----8<----
>>
>> Cheers,
>> -Slim
>> --
>> Jeroen C. "Slim" van Gelderen
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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
>



-- 
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


From fernando.berretta at gmail.com  Tue Oct 12 11:09:43 2010
From: fernando.berretta at gmail.com (Fernando Berretta)
Date: Tue, 12 Oct 2010 15:09:43 -0300
Subject: [Freeswitch-users] Originate Calls From an external program
In-Reply-To: <73E1ABB8-57C1-4975-8443-4F9E9C384C03@ipeva.fr>
References: <4CAEAD0B.8020704@gmail.com> <4CB472C2.4030706@gmail.com>
	<73E1ABB8-57C1-4975-8443-4F9E9C384C03@ipeva.fr>
Message-ID: <4CB4A467.7020007@gmail.com>

  I'm gonna check it thanks

On 10/12/2010 11:58 AM, David Ponzone wrote:
> Check ESL on the wiki.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 12/10/2010 ? 16:37, Fernando Berretta a ?crit :
>
>>  Hi,
>>
>> I'm newbie in FreeSwitch.
>>
>> Is there some way to instruct FreeSwitch from an external program in
>> order to originate calls to certain numbers an then bridge the calls 
>> etc. ?
>>
>> I'm looking fore something similar to Asterisk AMI  where we we use the
>> action Originate to achieve this
>> Is there any other way to do that in FreeSwitch
>>
>> Any help will be appreciated.
>>
>> Best Regards,
>> Fernando
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From fernando.berretta at gmail.com  Tue Oct 12 11:13:22 2010
From: fernando.berretta at gmail.com (Fernando Berretta)
Date: Tue, 12 Oct 2010 15:13:22 -0300
Subject: [Freeswitch-users] Originate Calls From an external program
In-Reply-To: <0FF607C13B7F7A42B5B80DA9EB204C9109D44BD6D2@srv-ex01-dal.intervoice.int>
References: <4CAEAD0B.8020704@gmail.com> <4CB472C2.4030706@gmail.com>
	<0FF607C13B7F7A42B5B80DA9EB204C9109D44BD6D2@srv-ex01-dal.intervoice.int>
Message-ID: <4CB4A542.6000901@gmail.com>

  Esser,

Your answer was very helpfull, thanks !

Is ESL reliable to production apps ?

Best Regards,
Fernando

On 10/12/2010 12:08 PM, Esser, Holger wrote:
> Fernando,
>
> Take a look at the socket interface.
> http://wiki.freeswitch.org/wiki/Event_Socket
>
> It is as close to AMI as you can get, just better ;)
>
> Here is a sample perl script that I found somewhere.
> #!/usr/bin/perl -w
>
> # This autodials via the manager API on an Asterisk box.
> # VERY useful for fast and easy scanning with asterisk! -natas
>
> use RPC::XML::Client;
> use Data::Dumper;
> use Net::Telnet;
> use warnings;
> use FreeSWITCH::Client;
> require ESL;
>
>
>
>
>
> my $command = shift;
> my $args = join(" ", @ARGV);
>
> my $con = new ESL::ESLconnection("127.0.0.1", "8021", "ClueCon");
>
>
>
>
> for ($Loop=0; $Loop<  1; $Loop++) {
>
>
>
>         my $e = $con->bgapi($command, $args);
>         #print $e->getBody();
>
>          select(undef,undef,undef,0.1);  # sleep 250ms
>
>
> #     select(undef,undef,undef,0.25); # sleep 1/4 second
> #     sleep (1.5);
>    # They hung up.  Move on.
>    print "Placing next call $Loop\n";
> }
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fernando Berretta
> Sent: Tuesday, October 12, 2010 9:38 AM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] Originate Calls From an external program
>
>    Hi,
>
> I'm newbie in FreeSwitch.
>
> Is there some way to instruct FreeSwitch from an external program in
> order to originate calls to certain numbers an then bridge the calls etc. ?
>
> I'm looking fore something similar to Asterisk AMI  where we we use the
> action Originate to achieve this
> Is there any other way to do that in FreeSwitch
>
> Any help will be appreciated.
>
> Best Regards,
> Fernando
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> No virus found in this incoming message.
> Checked by AVG - www.avg.com
> Version: 9.0.862 / Virus Database: 271.1.1/3189 - Release Date: 10/12/10 01:34:00
>
> No virus found in this outgoing message.
> Checked by AVG - www.avg.com
> Version: 9.0.862 / Virus Database: 271.1.1/3189 - Release Date: 10/12/10 01:34:00
>
> This e-mail transmission may contain information that is proprietary, privileged and/or confidential and is intended exclusively for the person(s) to whom it is addressed. Any use, copying, retention or disclosure by any person other than the intended recipient or the intended recipient's designees is strictly prohibited. If you are the intended recipient, you must treat the information in confidence and in accordance with all laws related to the privacy and confidentiality of such information.  If you are not the intended recipient or their designee, please notify the sender immediately by return e-mail and delete all copies of this email, including all attachments.
>
> Intervoice, Inc. 17811 Waterview Parkway Dallas, TX 75252 USA
>
> Intervoice Limited, Registered in England and Wales with number 2601740, 50 Park Road, Gatley, Cheshire, SK8 4HZ. VAT Number: 560421375
>
> Intervoice GmbH, Hagenauer Stra?e 55, 65203 Wiesbaden, Sitz der Gesellschaft: Wiesbaden, Handelsregister: HRB 8186 (Amtsgericht Wiesbaden), Gesch?ftsf?hrer: Wayne Barclay, Steffen Selbmann
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From curriegrad2004 at gmail.com  Tue Oct 12 13:12:03 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Tue, 12 Oct 2010 13:12:03 -0700
Subject: [Freeswitch-users] 64 bit CPU and 32 bit OS,
	is it good for freeswitch
In-Reply-To: <B664E6B3-4844-4B3C-ACB9-7C89027D3F9F@avgs.ca>
References: <COL110-W230774B749D9D716972D1688540@phx.gbl>
	<B664E6B3-4844-4B3C-ACB9-7C89027D3F9F@avgs.ca>
Message-ID: <AANLkTi=nMk7NSuPrdxLbjhbjyrXpBK=fFjh7RJ_nf-b_@mail.gmail.com>

I'm running a FreeSwitch box under 32-bit version of Fedora 13. So far
not too many problems except that it probably has a bad copy of gnutls
as mod_dingaling is unstable on the 32-bit Fedora 13 box I use.

On Tue, Oct 12, 2010 at 7:59 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> It shouldn't pose any direct problems as the CPU will perfectly emulate 32
> bits, but be aware that you won't get the same performance as using a
> complete 64 bits system. I would definitely get a 64 bits OS since your CPU
> supports it, or else its wasting your money/hardware.
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
> On 2010-10-12, at 7:09 AM, awais nazir wrote:
>
> Hi
>
> I am newbie using and appreciating this remarkable software.
>
> I have heard that CENTOS 5.3 is best for freeswitch deployment? but I am
> using 64 bit CPU and 32 bit OS.
>
> Can it give any problem?
>
> Earliest response will be appreciated.
>
>
> Thanks and Best Regards.
>
> Awais Nazeer
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Tue Oct 12 13:51:40 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 12 Oct 2010 13:51:40 -0700
Subject: [Freeswitch-users] 3-way calling fails with git
In-Reply-To: <AANLkTi==jVtq3AosUD=pYTjCq3O-k7dtvVPbxB3fnaa4@mail.gmail.com>
References: <AANLkTi==jVtq3AosUD=pYTjCq3O-k7dtvVPbxB3fnaa4@mail.gmail.com>
Message-ID: <AANLkTi=Oxw3tzX=EMppHk4oqEpgoo8LJKuoDK6B-iPr5@mail.gmail.com>

Malfunction! Need input!

Can you get a debug log, preferably with SIP trace? Drop on
pastebin.freeswitch.org and we'll take a peek.

-MC

On Tue, Oct 12, 2010 at 12:25 AM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hi,
>
> I am having problem with using the att_xfer app after upgrading to the
> latest git.  After the callee presses a meta_app key and bridged a 3rd party
> in, the callee got hung up immediately leaving the 3rd part and the original
> caller talking to each other only.
>
> I was able to get 3-way call to work in the previous release of fs.  Does
> anyone know if this is a bug or a bad config on my end?
>
> Thanks,
> Woody
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Kyle.Haefner at colostate.edu  Tue Oct 12 11:19:15 2010
From: Kyle.Haefner at colostate.edu (Kyle Haefner)
Date: Tue, 12 Oct 2010 12:19:15 -0600
Subject: [Freeswitch-users] fifo help
Message-ID: <AANLkTinv11QYE0UKnOcJE=v8ZADgX3q6AosyXAcCF1jS@mail.gmail.com>

Hi All,

I'm pretty new to freeswitch :)

I'm trying to set up the following scenario:

1. A caller calls in is greeted by a message that says, "Please hold while
your party is located"
2. The caller is placed on hold
3. An outbound call sequentially rings cell phones through a gateway
4. If a cellphone answers the "agent" must push a digit (to verify they are
not voice mail)
5. If a digit is pushed the agent is connected to the caller

I'm trying to implement this using mod_fifo, however when I add the caller
to the fifo, processing of that extension stops and I can't ever call and
test the cellphone to add them as a member of the fifo.

<context name="custom">
  <extension name="FIFO-Call">
    <condition field="destination_number" expression="^(100)$">
-       <action application="answer"/>
        <action application="playback"
data="/usr/local/freeswitch/sounds/en/us/callie/misc/noc_intro.wav"/>
        <action application="fifo" data="noc_fifo in nowait"/>
        <action application="sleep" data="300"/>
        <action application="set"
data="effective_caller_id_number=${outbound_caller_id_number}"/>
        <action application="set"
data="effective_caller_id_name=${outbound_caller_id_name}"/>
        <action application="set" data="ringback=${us-ring}"/>
        <action application="set" data="call_timeout=60"/>
        <action application="set"
data="group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/misc/prove_youre_human.wav"/>
        <action application="set" data="group_confirm_key=4"/>
        <action application="set" data="fifo_member(add noc_fifo
{fifo_member_wait=nowait}sofia/custom_dialplan/89883757 at pstn-gateway
:5060)"/>
    </condition>
  </extension>
</context>


Any help would be greatly appreciated!

Kyle
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From shamun.toha at gmail.com  Tue Oct 12 15:52:33 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Wed, 13 Oct 2010 00:52:33 +0200
Subject: [Freeswitch-users] BUG BUG -- mod_spidermonkey or mod_spidermonkey
	or mod_curl --- BUG BUG???
Message-ID: <AANLkTinng4=jMXJLXdAMZEJSn2V_s2MqRy02C+_0wS40@mail.gmail.com>

Hello,


I am almost 2 week on the same script and same problem.  It gets OK, and
then automatically hangup the call, instead of moving/transfer the call to
somewhere else.

**In fedora 12, the same script same setup is working 100% accurate. Then i
installed it to CentOS 5.5, there it doesnt work. What is the problem,
please advise kindly.

Should work:
===========
Here you can see i am receiving 'OK' via curl 2010-10-13 00:39:04.570826
[INFO] conference.js:45 OK
Normally it should now execute this:

function my_callback(string,arg)
{
  console_log("info", string);

   session.sayPhrase("valid", menuselection, "en");
   session.setVariable("myroom", menuselection);
   session.execute("conference" , menuselection, "en");



  return true;
}


Debug:
=======

2010-10-13 00:38:46.946636 [DEBUG] switch_ivr_play_say.c:244 Handle
speak-text:[Enter your room number.] (en:en)
2010-10-13 00:38:46.946636 [DEBUG] switch_ivr_play_say.c:2160 OPEN TTS
cepstral
2010-10-13 00:38:46.946636 [DEBUG] switch_ivr_play_say.c:2169 Raw Codec
Activated


2010-10-13 00:38:46.947636 [DEBUG] switch_ivr_play_say.c:1878 Speaking text:
Enter your room number.
2010-10-13 00:38:58.046751 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 3:800
2010-10-13 00:38:58.046751 [DEBUG] conference.js:126 digit: 3
2010-10-13 00:38:58.626757 [DEBUG] switch_ivr_play_say.c:2050 done speaking
text
2010-10-13 00:38:58.726760 [DEBUG] conference.js:140 Prompt
done=[DialByNumberMenu] Collected 1 digits [3]


2010-10-13 00:38:58.907732 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 2:800
2010-10-13 00:38:59.767758 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 3:800
2010-10-13 00:39:00.546774 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 2:800
2010-10-13 00:39:01.986791 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 9:800
2010-10-13 00:39:02.727801 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 8:800
2010-10-13 00:39:03.626796 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 2:800
2010-10-13 00:39:04.366833 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 1:800
2010-10-13 00:39:04.366833 [DEBUG] mod_spidermonkey_curl.c:182 Running:
method: [POST] url: [http://portal.x86_64.com/conference/query] data:
[room=32329821&security=___whyyoudothis___] cred=[] cb: [yes]
2010-10-13 00:39:04.570826 [INFO] conference.js:45 OK




2010-10-13 00:39:04.576826 [DEBUG] switch_ivr_play_say.c:244 Handle
execute:[sleep(500)] (en:en)
EXECUTE sofia/internal/1002 at 3.x86_64.com sleep(500)
2010-10-13 00:39:05.206844 [DEBUG] switch_ivr_play_say.c:244 Handle
speak-text:[Room number is invalid, Please try again.] (en:en)


2010-10-13 00:39:05.206844 [DEBUG] switch_ivr_play_say.c:2160 OPEN TTS
cepstral
2010-10-13 00:39:05.206844 [DEBUG] switch_ivr_play_say.c:2169 Raw Codec
Activated
2010-10-13 00:39:05.207837 [DEBUG] switch_ivr_play_say.c:1878 Speaking text:
Room number is invalid, Please try again.
2010-10-13 00:39:18.606958 [DEBUG] switch_ivr_play_say.c:2050 done speaking
text
2010-10-13 00:39:18.707961 [NOTICE] switch_core_state_machine.c:185
sofia/internal/1002 at 3.x86_64.com has executed the last dialplan instruction,
hanging up.
2010-10-13 00:39:18.707961 [DEBUG] switch_channel.c:2357 (sofia/internal/
1002 at 3.x86_64.com) Callstate Change ACTIVE -> HANGUP


2010-10-13 00:39:18.707961 [NOTICE] switch_core_state_machine.c:187 Hangup
sofia/internal/1002 at 3.x86_64.com [CS_EXECUTE] [NORMAL_CLEARING]
2010-10-13 00:39:18.707961 [DEBUG] switch_channel.c:2373 Send signal
sofia/internal/1002 at 3.x86_64.com [KILL]
2010-10-13 00:39:18.707961 [DEBUG] switch_core_session.c:1047 Send signal
sofia/internal/1002 at 3.x86_64.com [BREAK]
2010-10-13 00:39:18.707961 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/1002 at 3.x86_64.com) State EXECUTE going to sleep
2010-10-13 00:39:18.707961 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1002 at 3.x86_64.com) Running State Change CS_HANGUP
2010-10-13 00:39:18.707961 [DEBUG] switch_core_state_machine.c:535
(sofia/internal/1002 at 3.x86_64.com) State HANGUP
2010-10-13 00:39:18.707961 [DEBUG] mod_sofia.c:457 Channel sofia/internal/
1002 at 3.x86_64.com hanging up, cause: NORMAL_CLEARING
2010-10-13 00:39:18.723960 [DEBUG] mod_sofia.c:500 Sending BYE to
sofia/internal/1002 at 3.x86_64.com
2010-10-13 00:39:18.723960 [DEBUG] switch_core_state_machine.c:46
sofia/internal/1002 at 3.x86_64.com Standard HANGUP, cause: NORMAL_CLEARING
2010-10-13 00:39:18.723960 [DEBUG] switch_core_state_machine.c:535
(sofia/internal/1002 at 3.x86_64.com) State HANGUP going to sleep
2010-10-13 00:39:18.724961 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/1002 at 3.x86_64.com) State Change CS_HANGUP -> CS_REPORTING
2010-10-13 00:39:18.724961 [DEBUG] switch_core_session.c:1047 Send signal
sofia/internal/1002 at 3.x86_64.com [BREAK]
2010-10-13 00:39:18.724961 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1002 at 3.x86_64.com) Running State Change CS_REPORTING
2010-10-13 00:39:18.724961 [DEBUG] switch_core_state_machine.c:595
(sofia/internal/1002 at 3.x86_64.com) State REPORTING
2010-10-13 00:39:18.724961 [DEBUG] switch_core_state_machine.c:53
sofia/internal/1002 at 3.x86_64.com Standard REPORTING, cause: NORMAL_CLEARING
2010-10-13 00:39:18.724961 [DEBUG] switch_core_state_machine.c:595
(sofia/internal/1002 at 3.x86_64.com) State REPORTING going to sleep
2010-10-13 00:39:18.724961 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/1002 at 3.x86_64.com) State Change CS_REPORTING -> CS_DESTROY
2010-10-13 00:39:18.724961 [DEBUG] switch_core_session.c:1047 Send signal
sofia/internal/1002 at 3.x86_64.com [BREAK]
2010-10-13 00:39:18.724961 [DEBUG] switch_core_session.c:1210 Session 10
(sofia/internal/1002 at 3.x86_64.com) Locked, Waiting on external entities
2010-10-13 00:39:18.724961 [NOTICE] switch_core_session.c:1228 Session 10
(sofia/internal/1002 at 3.x86_64.com) Ended
2010-10-13 00:39:18.724961 [NOTICE] switch_core_session.c:1230 Close Channel
sofia/internal/1002 at 3.x86_64.com [CS_DESTROY]
2010-10-13 00:39:18.725963 [DEBUG] switch_core_state_machine.c:427
(sofia/internal/1002 at 3.x86_64.com) Callstate Change HANGUP -> DOWN
2010-10-13 00:39:18.725963 [DEBUG] switch_core_state_machine.c:430
(sofia/internal/1002 at 3.x86_64.com) Running State Change CS_DESTROY
2010-10-13 00:39:18.725963 [DEBUG] switch_core_state_machine.c:440
(sofia/internal/1002 at 3.x86_64.com) State DESTROY
2010-10-13 00:39:18.725963 [DEBUG] mod_sofia.c:362 sofia/internal/
1002 at 3.x86_64.com SOFIA DESTROY
2010-10-13 00:39:18.725963 [DEBUG] switch_core_state_machine.c:60
sofia/internal/1002 at 3.x86_64.com Standard DESTROY
2010-10-13 00:39:18.725963 [DEBUG] switch_core_state_machine.c:440
(sofia/internal/1002 at 3.x86_64.com) State DESTROY going to sleep
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From shamun.toha at gmail.com  Tue Oct 12 16:04:38 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Wed, 13 Oct 2010 01:04:38 +0200
Subject: [Freeswitch-users] BUG BUG -- mod_spidermonkey or
 mod_spidermonkey or mod_curl --- BUG BUG???
In-Reply-To: <AANLkTinng4=jMXJLXdAMZEJSn2V_s2MqRy02C+_0wS40@mail.gmail.com>
References: <AANLkTinng4=jMXJLXdAMZEJSn2V_s2MqRy02C+_0wS40@mail.gmail.com>
Message-ID: <AANLkTi=AzzT6SFtesR8RsJO9q-BKCVYJUB8KjJXrbhHK@mail.gmail.com>

Never mind its solved. It was receiving 4 length instead of 2.

class iCanSolveNasaProblemsLolController extends Zend_Controller_Action
{

  public function init()
  {
    $this->_helper->layout()->disableLayout();
    $this->_helper->viewRenderer->setNoRender();
    //$this->getHelper('layout')->disableLayout();
    //$this->getHelper('ViewRenderer')->setNoRender();
    //Zend_Controller_Action_HelperBroker::removeHelper('Layout');
    $a = 'OK';

    $this->getResponse()
         ->setHeader('Content-Type', 'application/text')
         ->setBody($a)
         ->sendResponse();
    exit;
  }


}


On Wed, Oct 13, 2010 at 12:52 AM, Shamun toha md <shamun.toha at gmail.com>wrote:

> Hello,
>
>
> I am almost 2 week on the same script and same problem.  It gets OK, and
> then automatically hangup the call, instead of moving/transfer the call to
> somewhere else.
>
> **In fedora 12, the same script same setup is working 100% accurate. Then i
> installed it to CentOS 5.5, there it doesnt work. What is the problem,
> please advise kindly.
>
> Should work:
> ===========
> Here you can see i am receiving 'OK' via curl 2010-10-13 00:39:04.570826
> [INFO] conference.js:45 OK
> Normally it should now execute this:
>
> function my_callback(string,arg)
> {
>   console_log("info", string);
>
>    session.sayPhrase("valid", menuselection, "en");
>    session.setVariable("myroom", menuselection);
>    session.execute("conference" , menuselection, "en");
>
>
>
>   return true;
> }
>
>
> Debug:
> =======
>
> 2010-10-13 00:38:46.946636 [DEBUG] switch_ivr_play_say.c:244 Handle
> speak-text:[Enter your room number.] (en:en)
> 2010-10-13 00:38:46.946636 [DEBUG] switch_ivr_play_say.c:2160 OPEN TTS
> cepstral
> 2010-10-13 00:38:46.946636 [DEBUG] switch_ivr_play_say.c:2169 Raw Codec
> Activated
>
>
> 2010-10-13 00:38:46.947636 [DEBUG] switch_ivr_play_say.c:1878 Speaking
> text: Enter your room number.
> 2010-10-13 00:38:58.046751 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 3:800
> 2010-10-13 00:38:58.046751 [DEBUG] conference.js:126 digit: 3
> 2010-10-13 00:38:58.626757 [DEBUG] switch_ivr_play_say.c:2050 done speaking
> text
> 2010-10-13 00:38:58.726760 [DEBUG] conference.js:140 Prompt
> done=[DialByNumberMenu] Collected 1 digits [3]
>
>
> 2010-10-13 00:38:58.907732 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 2:800
> 2010-10-13 00:38:59.767758 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 3:800
> 2010-10-13 00:39:00.546774 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 2:800
> 2010-10-13 00:39:01.986791 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 9:800
> 2010-10-13 00:39:02.727801 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 8:800
> 2010-10-13 00:39:03.626796 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 2:800
> 2010-10-13 00:39:04.366833 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 1:800
> 2010-10-13 00:39:04.366833 [DEBUG] mod_spidermonkey_curl.c:182 Running:
> method: [POST] url: [http://portal.x86_64.com/conference/query] data:
> [room=32329821&security=___whyyoudothis___] cred=[] cb: [yes]
> 2010-10-13 00:39:04.570826 [INFO] conference.js:45 OK
>
>
>
>
> 2010-10-13 00:39:04.576826 [DEBUG] switch_ivr_play_say.c:244 Handle
> execute:[sleep(500)] (en:en)
> EXECUTE sofia/internal/1002 at 3.x86_64.com sleep(500)
> 2010-10-13 00:39:05.206844 [DEBUG] switch_ivr_play_say.c:244 Handle
> speak-text:[Room number is invalid, Please try again.] (en:en)
>
>
> 2010-10-13 00:39:05.206844 [DEBUG] switch_ivr_play_say.c:2160 OPEN TTS
> cepstral
> 2010-10-13 00:39:05.206844 [DEBUG] switch_ivr_play_say.c:2169 Raw Codec
> Activated
> 2010-10-13 00:39:05.207837 [DEBUG] switch_ivr_play_say.c:1878 Speaking
> text: Room number is invalid, Please try again.
> 2010-10-13 00:39:18.606958 [DEBUG] switch_ivr_play_say.c:2050 done speaking
> text
> 2010-10-13 00:39:18.707961 [NOTICE] switch_core_state_machine.c:185
> sofia/internal/1002 at 3.x86_64.com has executed the last dialplan
> instruction, hanging up.
> 2010-10-13 00:39:18.707961 [DEBUG] switch_channel.c:2357 (sofia/internal/
> 1002 at 3.x86_64.com) Callstate Change ACTIVE -> HANGUP
>
>
> 2010-10-13 00:39:18.707961 [NOTICE] switch_core_state_machine.c:187 Hangup
> sofia/internal/1002 at 3.x86_64.com [CS_EXECUTE] [NORMAL_CLEARING]
> 2010-10-13 00:39:18.707961 [DEBUG] switch_channel.c:2373 Send signal
> sofia/internal/1002 at 3.x86_64.com [KILL]
> 2010-10-13 00:39:18.707961 [DEBUG] switch_core_session.c:1047 Send signal
> sofia/internal/1002 at 3.x86_64.com [BREAK]
> 2010-10-13 00:39:18.707961 [DEBUG] switch_core_state_machine.c:348
> (sofia/internal/1002 at 3.x86_64.com) State EXECUTE going to sleep
> 2010-10-13 00:39:18.707961 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1002 at 3.x86_64.com) Running State Change CS_HANGUP
> 2010-10-13 00:39:18.707961 [DEBUG] switch_core_state_machine.c:535
> (sofia/internal/1002 at 3.x86_64.com) State HANGUP
> 2010-10-13 00:39:18.707961 [DEBUG] mod_sofia.c:457 Channel sofia/internal/
> 1002 at 3.x86_64.com hanging up, cause: NORMAL_CLEARING
> 2010-10-13 00:39:18.723960 [DEBUG] mod_sofia.c:500 Sending BYE to
> sofia/internal/1002 at 3.x86_64.com
> 2010-10-13 00:39:18.723960 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/1002 at 3.x86_64.com Standard HANGUP, cause: NORMAL_CLEARING
> 2010-10-13 00:39:18.723960 [DEBUG] switch_core_state_machine.c:535
> (sofia/internal/1002 at 3.x86_64.com) State HANGUP going to sleep
> 2010-10-13 00:39:18.724961 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/1002 at 3.x86_64.com) State Change CS_HANGUP -> CS_REPORTING
> 2010-10-13 00:39:18.724961 [DEBUG] switch_core_session.c:1047 Send signal
> sofia/internal/1002 at 3.x86_64.com [BREAK]
> 2010-10-13 00:39:18.724961 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1002 at 3.x86_64.com) Running State Change CS_REPORTING
> 2010-10-13 00:39:18.724961 [DEBUG] switch_core_state_machine.c:595
> (sofia/internal/1002 at 3.x86_64.com) State REPORTING
> 2010-10-13 00:39:18.724961 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/1002 at 3.x86_64.com Standard REPORTING, cause:
> NORMAL_CLEARING
> 2010-10-13 00:39:18.724961 [DEBUG] switch_core_state_machine.c:595
> (sofia/internal/1002 at 3.x86_64.com) State REPORTING going to sleep
> 2010-10-13 00:39:18.724961 [DEBUG] switch_core_state_machine.c:327
> (sofia/internal/1002 at 3.x86_64.com) State Change CS_REPORTING -> CS_DESTROY
> 2010-10-13 00:39:18.724961 [DEBUG] switch_core_session.c:1047 Send signal
> sofia/internal/1002 at 3.x86_64.com [BREAK]
> 2010-10-13 00:39:18.724961 [DEBUG] switch_core_session.c:1210 Session 10
> (sofia/internal/1002 at 3.x86_64.com) Locked, Waiting on external entities
> 2010-10-13 00:39:18.724961 [NOTICE] switch_core_session.c:1228 Session 10
> (sofia/internal/1002 at 3.x86_64.com) Ended
> 2010-10-13 00:39:18.724961 [NOTICE] switch_core_session.c:1230 Close
> Channel sofia/internal/1002 at 3.x86_64.com [CS_DESTROY]
> 2010-10-13 00:39:18.725963 [DEBUG] switch_core_state_machine.c:427
> (sofia/internal/1002 at 3.x86_64.com) Callstate Change HANGUP -> DOWN
> 2010-10-13 00:39:18.725963 [DEBUG] switch_core_state_machine.c:430
> (sofia/internal/1002 at 3.x86_64.com) Running State Change CS_DESTROY
> 2010-10-13 00:39:18.725963 [DEBUG] switch_core_state_machine.c:440
> (sofia/internal/1002 at 3.x86_64.com) State DESTROY
> 2010-10-13 00:39:18.725963 [DEBUG] mod_sofia.c:362 sofia/internal/
> 1002 at 3.x86_64.com SOFIA DESTROY
> 2010-10-13 00:39:18.725963 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/1002 at 3.x86_64.com Standard DESTROY
> 2010-10-13 00:39:18.725963 [DEBUG] switch_core_state_machine.c:440
> (sofia/internal/1002 at 3.x86_64.com) State DESTROY going to sleep
>
>
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From thedjallen at gmail.com  Tue Oct 12 16:08:10 2010
From: thedjallen at gmail.com (David Allen)
Date: Wed, 13 Oct 2010 10:08:10 +1100
Subject: [Freeswitch-users] Routing DIDs to Dynamically Registered SIP
 User
In-Reply-To: <157B0F19-E2AD-4B3E-9772-69CE015BE966@ipeva.fr>
References: <4CAEAD0B.8020704@gmail.com>	<AANLkTimv0hvr6BYfKZW1zrdGgHrnMGxpidmpNL-5eu1T@mail.gmail.com>	<AANLkTimCNR3KYpb0dxnikbBAFi2uL_24U8u3rq2LBhYO@mail.gmail.com>	<AANLkTi=PMDD2aV45pivoB69OFaUHkDsZJESO-M_A5dPV@mail.gmail.com>
	<157B0F19-E2AD-4B3E-9772-69CE015BE966@ipeva.fr>
Message-ID: <4CB4EA5A.6000600@gmail.com>

  Hi David,

     On other soft-switches I've worked with in the past I have been 
able to do this. Some of the PBX's which specifically use the Invite URI 
rather than to TO include Asterisk, Zultys and a few others and also 
Gateways such as a Quintum.

Regards,
David

On 12/10/2010 5:24 PM, David Ponzone wrote:
> I really don't think it is as:
> 1/ i was not able to find a parameter to alter the INVITE URI
> 2/ FS itself has a parameter to change the behaviour and use the SIP 
> To instead of the INVITE URI as the destination_number, so it would 
> tend to prove that it's the other party's job to use the right field.
>
> David (Allen, I am not talking to myself :) ), can you give us some 
> examples of PBXs who rely only on the INVITE URI ?
> Are you sure they don't have a parameter to change that ?
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 12/10/2010 ? 07:18, Michael Collins a ?crit :
>
>> Ah, I was reading too much into that.
>>
>> If I understand correctly you need to send the INVITE to the actual 
>> endpoint, but you need the target URI be the same as the URI in the 
>> To: field? Is that even legal in SIP? (Not that legal matters since 
>> vendors do whatever they want anyway...)
>>
>> -MC
>>
>> On Mon, Oct 11, 2010 at 3:59 PM, David Allen <thedjallen at gmail.com 
>> <mailto:thedjallen at gmail.com>> wrote:
>>
>>     Hi Michael,
>>
>>     Its a SIP UA that registers locally to Freeswitch.
>>
>>     Regards,
>>     David
>>
>>     On Tue, Oct 12, 2010 at 9:51 AM, Michael Collins
>>     <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>>     > What is a "dynamically registered SIP user"?
>>     > -MC
>>     >
>>     > On Thu, Oct 7, 2010 at 10:32 PM, David Allen
>>     <thedjallen at gmail.com <mailto:thedjallen at gmail.com>> wrote:
>>     >>
>>     >>  Hi,
>>     >>
>>     >> I'm trying to send multiple Direct Indial Numbers down to a
>>     dynamically
>>     >> registered SIP User. I need to ensure that both the To and
>>     Target URI
>>     >> contain the direct Indial number. I'm able to modify the SIP
>>     TO Header
>>     >> of a call that is sent to them like below:
>>     >>
>>     >> <action application="bridge"
>>     >> data="sofia/external/56778977%${domain}^61390009000"/>
>>     >>
>>     >> which sends the request as:
>>     >>
>>     >>
>>     >>
>>     ------------------------------------------------------------------------
>>     >>
>>     >>    INVITE sip:56778977 at 192.168.22.2:5061
>>     <http://sip:56778977 at 192.168.22.2:5061> SIP/2.0
>>     >>
>>     >>    Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j
>>     >>
>>     >>    Max-Forwards: 69
>>     >>
>>     >>    From: "0390001000" <sip:0390001000 at 192.168.2.200
>>     <mailto:sip%3A0390001000 at 192.168.2.200>>;tag=K4HHaZ9v1H07Q
>>     >>
>>     >>    To: <sip:61390009000 at 192.168.2.200
>>     <mailto:sip%3A61390009000 at 192.168.2.200>>
>>     >>
>>     >>  But in order to maintain compatability with a number of
>>     PBX's/VoIP
>>     >> devices on the market, I need to be able to send the invite to the
>>     >> dynamically registered SIP user, however have it set the
>>     Target URI and
>>     >> To as the same contact number like below:
>>     >>
>>     >>
>>     >>
>>     ------------------------------------------------------------------------
>>     >>
>>     >>    INVITE sip:61390009000 at 192.168.22.2:5061
>>     <http://sip:61390009000 at 192.168.22.2:5061> SIP/2.0
>>     >>
>>     >>    Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j
>>     >>
>>     >>    Max-Forwards: 69
>>     >>
>>     >>    From: "0390001000" <sip:0390001000 at 192.168.2.200
>>     <mailto:sip%3A0390001000 at 192.168.2.200>>;tag=K4HHaZ9v1H07Q
>>     >>
>>     >>    To: <sip:61390009000 at 192.168.22.2:5061
>>     <http://sip:61390009000 at 192.168.22.2:5061>>
>>     >>
>>     >> Is there a way to do this setting via variables? I can't seem
>>     to find
>>     >> any details for it.
>>     >>
>>     >> Thanks
>>     >>
>>     >> David
>>     >>
>>     >> _______________________________________________
>>     >> FreeSWITCH-users mailing list
>>     >> FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     >>
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     >> http://www.freeswitch.org
>>     >
>>     >
>>     > _______________________________________________
>>     > FreeSWITCH-users mailing list
>>     > FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     >
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     > http://www.freeswitch.org
>>     >
>>     >
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From thedjallen at gmail.com  Tue Oct 12 16:11:49 2010
From: thedjallen at gmail.com (David Allen)
Date: Wed, 13 Oct 2010 10:11:49 +1100
Subject: [Freeswitch-users] Routing DIDs to Dynamically Registered SIP
 User
In-Reply-To: <AANLkTikZDoW1_rSbDoCL+onwgj9iNkE4tbc=ycFTd_fO@mail.gmail.com>
References: <4CAEAFC4.3040504@gmail.com>
	<AANLkTikZDoW1_rSbDoCL+onwgj9iNkE4tbc=ycFTd_fO@mail.gmail.com>
Message-ID: <4CB4EB35.7040503@gmail.com>

  Hi,

Thanks for this - I'll give this a try as well. I ended up using ODBC to 
get the internal Freeswitch DB into MySQL. Since I serve the config via 
XML I did a lookup in the DB to get the Network Address and Port of the 
user and sent a custom dial/bridge string to Freeswitch with the Direct 
Indials number which is working a treat.

Thanks for everyones input.

David

On 12/10/2010 5:50 PM, Ognjen Seslija wrote:
> <action application="bridge" 
> data="{sip_invite_req_uri=sip:61390009000 at 192.168.22.2:5061 
> <http://sip:61390009000 at 192.168.22.2:5061>,sip_invite_to_uri=sip:61390009000 at 192.168.22.2:5061}sofia/external/56778977%${domain}"/>
>
> On Fri, Oct 8, 2010 at 7:44 AM, David Allen <thedjallen at gmail.com 
> <mailto:thedjallen at gmail.com>> wrote:
>
>      Hi,
>
>     I'm trying to send multiple Direct Indial Numbers down to a
>     dynamically
>     registered SIP User. I need to ensure that both the To and Target URI
>     contain the direct Indial number. I'm able to modify the SIP TO Header
>     of a call that is sent to them like below:
>
>     <action application="bridge"
>     data="sofia/external/56778977%${domain}^61390009000"/>
>
>     which sends the request as:
>
>      
>      ------------------------------------------------------------------------
>
>        INVITE sip:56778977 at 192.168.22.2:5061
>     <http://sip:56778977 at 192.168.22.2:5061> SIP/2.0
>
>        Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j
>
>        Max-Forwards: 69
>
>        From: "0390001000" <sip:0390001000 at 192.168.2.200
>     <mailto:sip%3A0390001000 at 192.168.2.200>>;tag=K4HHaZ9v1H07Q
>
>        To: <sip:61390009000 at 192.168.2.200
>     <mailto:sip%3A61390009000 at 192.168.2.200>>
>
>      But in order to maintain compatability with a number of PBX's/VoIP
>     devices on the market, I need to be able to send the invite to the
>     dynamically registered SIP user, however have it set the Target
>     URI and
>     To as the same contact number like below:
>
>      
>      ------------------------------------------------------------------------
>
>        INVITE sip:61390009000 at 192.168.22.2:5061
>     <http://sip:61390009000 at 192.168.22.2:5061> SIP/2.0
>
>        Via: SIP/2.0/UDP 202.0.155.73;rport;branch=z9hG4bKXeND9cHtgKF2j
>
>        Max-Forwards: 69
>
>        From: "0390001000" <sip:0390001000 at 192.168.2.200
>     <mailto:sip%3A0390001000 at 192.168.2.200>>;tag=K4HHaZ9v1H07Q
>
>        To: <sip:61390009000 at 192.168.22.2:5061
>     <http://sip:61390009000 at 192.168.22.2:5061>>
>
>     Is there a way to do this setting via variables? I can't seem to find
>     any details for it.
>
>     Thanks
>
>     David
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From dujinfang at gmail.com  Tue Oct 12 17:01:22 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 13 Oct 2010 08:01:22 +0800
Subject: [Freeswitch-users] Originate Calls From an external program
In-Reply-To: <4CB4A542.6000901@gmail.com>
References: <4CAEAD0B.8020704@gmail.com> <4CB472C2.4030706@gmail.com>
	<0FF607C13B7F7A42B5B80DA9EB204C9109D44BD6D2@srv-ex01-dal.intervoice.int>
	<4CB4A542.6000901@gmail.com>
Message-ID: <AANLkTimTO6_gCLPOqXeq0nY_=TG=LvOG1Mb8vNLMAYKh@mail.gmail.com>

On Wed, Oct 13, 2010 at 2:13 AM, Fernando Berretta
<fernando.berretta at gmail.com> wrote:
> ?Esser,
>
> Your answer was very helpfull, thanks !
>
> Is ESL reliable to production apps ?
>

That's what ESL is designed to do.

> Best Regards,
> Fernando
>
> On 10/12/2010 12:08 PM, Esser, Holger wrote:
>> Fernando,
>>
>> Take a look at the socket interface.
>> http://wiki.freeswitch.org/wiki/Event_Socket
>>
>> It is as close to AMI as you can get, just better ;)
>>
>> Here is a sample perl script that I found somewhere.
>> #!/usr/bin/perl -w
>>
>> # This autodials via the manager API on an Asterisk box.
>> # VERY useful for fast and easy scanning with asterisk! -natas
>>
>> use RPC::XML::Client;
>> use Data::Dumper;
>> use Net::Telnet;
>> use warnings;
>> use FreeSWITCH::Client;
>> require ESL;
>>
>>
>>
>>
>>
>> my $command = shift;
>> my $args = join(" ", @ARGV);
>>
>> my $con = new ESL::ESLconnection("127.0.0.1", "8021", "ClueCon");
>>
>>
>>
>>
>> for ($Loop=0; $Loop< ?1; $Loop++) {
>>
>>
>>
>> ? ? ? ? my $e = $con->bgapi($command, $args);
>> ? ? ? ? #print $e->getBody();
>>
>> ? ? ? ? ?select(undef,undef,undef,0.1); ?# sleep 250ms
>>
>>
>> # ? ? select(undef,undef,undef,0.25); # sleep 1/4 second
>> # ? ? sleep (1.5);
>> ? ?# They hung up. ?Move on.
>> ? ?print "Placing next call $Loop\n";
>> }
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fernando Berretta
>> Sent: Tuesday, October 12, 2010 9:38 AM
>> To: FreeSWITCH Users Help
>> Subject: [Freeswitch-users] Originate Calls From an external program
>>
>> ? ?Hi,
>>
>> I'm newbie in FreeSwitch.
>>
>> Is there some way to instruct FreeSwitch from an external program in
>> order to originate calls to certain numbers an then bridge the calls etc. ?
>>
>> I'm looking fore something similar to Asterisk AMI ?where we we use the
>> action Originate to achieve this
>> Is there any other way to do that in FreeSwitch
>>
>> Any help will be appreciated.
>>
>> Best Regards,
>> Fernando
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> No virus found in this incoming message.
>> Checked by AVG - www.avg.com
>> Version: 9.0.862 / Virus Database: 271.1.1/3189 - Release Date: 10/12/10 01:34:00
>>
>> No virus found in this outgoing message.
>> Checked by AVG - www.avg.com
>> Version: 9.0.862 / Virus Database: 271.1.1/3189 - Release Date: 10/12/10 01:34:00
>>
>> This e-mail transmission may contain information that is proprietary, privileged and/or confidential and is intended exclusively for the person(s) to whom it is addressed. Any use, copying, retention or disclosure by any person other than the intended recipient or the intended recipient's designees is strictly prohibited. If you are the intended recipient, you must treat the information in confidence and in accordance with all laws related to the privacy and confidentiality of such information. ?If you are not the intended recipient or their designee, please notify the sender immediately by return e-mail and delete all copies of this email, including all attachments.
>>
>> Intervoice, Inc. 17811 Waterview Parkway Dallas, TX 75252 USA
>>
>> Intervoice Limited, Registered in England and Wales with number 2601740, 50 Park Road, Gatley, Cheshire, SK8 4HZ. VAT Number: 560421375
>>
>> Intervoice GmbH, Hagenauer Stra?e 55, 65203 Wiesbaden, Sitz der Gesellschaft: Wiesbaden, Handelsregister: HRB 8186 (Amtsgericht Wiesbaden), Gesch?ftsf?hrer: Wayne Barclay, Steffen Selbmann
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From dujinfang at gmail.com  Tue Oct 12 17:12:29 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 13 Oct 2010 08:12:29 +0800
Subject: [Freeswitch-users] fifo help
In-Reply-To: <AANLkTinv11QYE0UKnOcJE=v8ZADgX3q6AosyXAcCF1jS@mail.gmail.com>
References: <AANLkTinv11QYE0UKnOcJE=v8ZADgX3q6AosyXAcCF1jS@mail.gmail.com>
Message-ID: <AANLkTi=EJeQL-V2f1acYOmM5xetyvTzUTV+rNBVb6-3=@mail.gmail.com>

add your members in fifo.conf.xml  in the <fifos></fifos> section and
reload mod_fifo.

http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback

http://wiki.freeswitch.org/wiki/Simple_call_center_using_mod_fifo

On Wed, Oct 13, 2010 at 2:19 AM, Kyle Haefner
<Kyle.Haefner at colostate.edu> wrote:
> Hi All,
> I'm pretty new to freeswitch :)
> I'm trying to set up the following scenario:
> 1. A caller calls in is greeted by a message that says, "Please hold while
> your party is located"
> 2. The caller is placed on hold
> 3. An outbound call sequentially rings cell phones through a gateway
> 4. If a cellphone answers the "agent" must push a digit (to verify they are
> not voice mail)
> 5. If a digit is pushed the agent is connected to the caller
> I'm trying to implement this using mod_fifo, however when I add the caller
> to the fifo, processing of that extension stops and I can't ever call and
> test the cellphone to add them as a member of the fifo.
> <context name="custom">
> ??<extension name="FIFO-Call">
> ?? ?<condition field="destination_number" expression="^(100)$">
> - ? ? ? <action application="answer"/>
> ?? ? ? ?<action application="playback"
> data="/usr/local/freeswitch/sounds/en/us/callie/misc/noc_intro.wav"/>
> ?? ? ? ?<action application="fifo" data="noc_fifo in nowait"/>
> ?? ? ? ?<action application="sleep" data="300"/>
> ?? ? ? ?<action application="set"
> data="effective_caller_id_number=${outbound_caller_id_number}"/>
> ?? ? ? ?<action application="set"
> data="effective_caller_id_name=${outbound_caller_id_name}"/>
> ?? ? ? ?<action application="set" data="ringback=${us-ring}"/>
> ?? ? ? ?<action application="set" data="call_timeout=60"/>
> ?? ? ? ?<action application="set"
> data="group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/misc/prove_youre_human.wav"/>
> ?? ? ? ?<action application="set" data="group_confirm_key=4"/>
> ?? ? ? ?<action application="set" data="fifo_member(add noc_fifo
> {fifo_member_wait=nowait}sofia/custom_dialplan/89883757 at pstn-gateway:5060)"/>
> ?? ?</condition>
> ??</extension>
> </context>
>
> Any help would be greatly appreciated!
> Kyle
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From mario_fs at mgtech.com  Tue Oct 12 17:22:58 2010
From: mario_fs at mgtech.com (Mario)
Date: Tue, 12 Oct 2010 17:22:58 -0700
Subject: [Freeswitch-users] Can uPNP be turned off?
Message-ID: <4CB4FBE2.30903@mgtech.com>

I am using ext... autonat:1.2.3.4 in sip internal and external as per
wiki. The sofia status is ok for the profiles bu the nat_map shows the
wrong address because the router is sending the wrong wan IP in the pnp
request. Since I have autonat:1.2.3.4 coded is there a way to force FS
not to use uPNP so it does not request nat info from the router? I need
it on in the router for other things. Thanks. Mario


From msc at freeswitch.org  Tue Oct 12 17:58:31 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 12 Oct 2010 17:58:31 -0700
Subject: [Freeswitch-users] valet_park timeout and spot announcement
In-Reply-To: <4CAB6A30.7070608@lightnex.com>
References: <4CAB6A30.7070608@lightnex.com>
Message-ID: <AANLkTi=F=YV+5vG6OvLTNAGJ9pFvRO131w_pcDkjk7Ee@mail.gmail.com>

Jeremy,

My apologies for taking so long to reply. I was kicking around ways to do
this. Comments inline.

On Tue, Oct 5, 2010 at 11:10 AM, Jeremy Stricker <jstricker at lightnex.com>wrote:

> Hello all,
>
> Is there anyway to set a call timeout when a call is valet parked so
> that it will ring back to the extension that parked it if it isn't
> answered by the intended recipient?  We have an user who parks calls and
> then blindly announces the parking spot over an intercom.  With this
> setup there is a great risk of a caller getting stuck in park.  If there
> isn't a timeout method, can anyone recommend any alternate setup that
> would allow at least two parking spots with visual indication of calls
> parked on the handsets (Aastra 6730i and 6757i).
>

I couldn't find a simple way to do this but the following method worked for
me in a lab environment. Assume that 6100 is the park extension and
6101-6199 are the parking stalls. Add these <condition> blocks to the end of
your "global" extension (it needs to be executed for every call):

      <condition field="${ani}" expression="^(10\d\d)$"> <!-- include only
'parker' extensions -->
      <condition field="destination_number" expression="^(6100)$">
        <anti-action application="hash"
data="insert/valet_recall/${destination_number}/${uuid}"/>
      </condition>

Then create the valet_park extensions:

    <extension name="valet_park_retrieve">
      <condition field="destination_number" expression="^(61\d[1-9])$">
        <action application="answer"/>
        <action application="lua" data="cancel_valet_recall.lua valet_lot
$1"/>
        <action application="valet_park" data="valet_lot $1"/>
      </condition>
    </extension>

    <extension name="valet_park_with_recall_timer">
      <condition field="destination_number" expression="^(6100)$">
        <action application="answer"/>
        <action application="set"
data="valet_uuid=${hash(select/valet_recall/${caller_id_number})}"/>
        <action application="log" data="INFO Pulled [${valet_uuid}] from
hash valet_recall/${caller_id_number}"/>
        <!-- the magic happens on the next line. The +30 means 30 seconds
later x-fer the call back to the parker -->
        <action application="set" data="api_res=${sched_api +30
uuid_transfer ${valet_uuid} ${valet_uuid} ${ani}}"/>
        <action application="valet_park" data="valet_lot auto in 6101
6199"/>
      </condition>
    </extension>

The trick is to use "sched_api" API to schedule the other leg to be
transferred back to the parker. We use the hash API to store some
information, namely the parker's extension number and the uuid of the parked
leg. Change the +30 to however many seconds you want the call to be parked
before recalling. Here's the catch: if someone does come along and grab that
call out of the parking stall then we need to remove the scheduled transfer!
I thought a Lua script would be the easiest way to handle that, so when the
person picking up the parked call dials 6001 e.g. then it launches a quickie
Lua script that removes the sched_api from the task list. Here's the Lua
script:

-- cancel_valet_recall.lua
-- 
-- parse the valet lot in question and find the uuid for the extension
-- remove the uuid from the sched task list since this call is now being
unparked
--


valet_lot = argv[1]
valet_ext = argv[2]
uuid      = session:getVariable('uuid')


--freeswitch.consoleLog('INFO','Lot: ' .. valet_lot .. ' , Ext: ' ..
valet_ext .. ' , uuid: ' .. uuid .. "\n")


api = freeswitch.API()


if ( valet_lot == nil and valet_ext == nil ) then
    -- improper args... (feel free to do some better error handling)
else
    -- sched_del uuid for the uuid in valet_lot, valet_ext
    valet_info = api:executeString('valet_info ' .. valet_lot)
    -- freeswitch.consoleLog("INFO","\n" .. valet_info .. "\n\n")


    valet_data = string.match(valet_info,"<extension uuid=.->" .. valet_ext
.. "</extension>")
    -- freeswitch.consoleLog("INFO","uuid data line is: " .. valet_data ..
"\n\n")


    valet_uuid = string.gsub(valet_data,'<extension uuid="(.-)">' ..
valet_ext .. "</extension>","%1")
    --freeswitch.consoleLog("INFO","valet uuid is: " .. valet_uuid ..
"\n\n")


    -- perform the sched_del
    api_res = api:executeString('sched_del ' .. valet_uuid)
    --freeswitch.consoleLog("INFO","result of sched_del " .. valet_uuid .. "
is " .. api_res .. "\n\n")
end

Throw this into conf/scripts/cancel_valet_recall.lua and give it a shot. Let
me know how it goes. If it works and if no one has any obvious improvements
then I will toss it up on the wiki as an example of how to do a valet_park
recall timer.

If you want to learn more about how this works then uncomment the log
statements in the Lua script. Also after parking but before picking up the
call go to the fs_cli and do "show tasks" and you'll see what the task looks
like. I used the parked leg's uuid as the 'group id' in sched_api so that it
is easy to find and remove when unparking the call.

Have fun and let me know if this helps or not.
-MC


> Secondly, is there anyway to stop read-back of the parking spot on the
> parkee's side of the call?  Currently, the parker performs an attended
> transfer, waits for and hears the spot read-back.  The parkee hears MOH
> while the transfer is being performed, but then the spot announcement
> themselves as soon as the parker finishes the attended transfer by
> hanging up.
>

The parker is hanging up too soon. If he/she waits until he/she hears music
and then completes the transfer it should be okay. I just tested this.
-MC
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From msc at freeswitch.org  Tue Oct 12 17:59:37 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 12 Oct 2010 17:59:37 -0700
Subject: [Freeswitch-users] Can uPNP be turned off?
In-Reply-To: <4CB4FBE2.30903@mgtech.com>
References: <4CB4FBE2.30903@mgtech.com>
Message-ID: <AANLkTinvkRkJsnyC98_Uc2PS633ohta3QB8C7C8gPHXx@mail.gmail.com>

launch freeswitch with -nonat like this:

freeswitch -nc -nonat

-MC

On Tue, Oct 12, 2010 at 5:22 PM, Mario <mario_fs at mgtech.com> wrote:

> I am using ext... autonat:1.2.3.4 in sip internal and external as per
> wiki. The sofia status is ok for the profiles bu the nat_map shows the
> wrong address because the router is sending the wrong wan IP in the pnp
> request. Since I have autonat:1.2.3.4 coded is there a way to force FS
> not to use uPNP so it does not request nat info from the router? I need
> it on in the router for other things. Thanks. Mario
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From peter.schrock at gmail.com  Tue Oct 12 18:15:11 2010
From: peter.schrock at gmail.com (Peter Schrock)
Date: Tue, 12 Oct 2010 18:15:11 -0700
Subject: [Freeswitch-users] Problems with fs_cli on OS X 10.6.4
In-Reply-To: <AANLkTi=ectqfWOymqRuPK_urDSZ1q0rPLxA2g8nCtB2v@mail.gmail.com>
References: <AANLkTingeYfPT_b6dG3bKQOVjxYsmBTKhDuLw_3BKGac@mail.gmail.com>
	<AANLkTi=YZsrcSWbrnUrZ6+P_fnXpQVAROCNc+9okF0X9@mail.gmail.com>
	<AANLkTinFAifa6BPLDqYVEEh1onCobbgJX4DHC_Kzdn_u@mail.gmail.com>
	<AANLkTinRZY_DwDXqEDumMaKuXc39zveHayHXCYZa-=_q@mail.gmail.com>
	<AANLkTi=ectqfWOymqRuPK_urDSZ1q0rPLxA2g8nCtB2v@mail.gmail.com>
Message-ID: <AANLkTinaudYzr_T_6CGXCCtQnB2Jopx=-_AHtaucRuM5@mail.gmail.com>

Thanks for your help. I don't know what happen but I did the "make current"
and rebuilt the whole thing and now it seems to be working fine. At least
now I know some tricks to try out to see if I can get it working should it
happen again. Thanks again.

PeterS

On Tue, Oct 12, 2010 at 8:01 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Try telnet to localhost port 8021 and see if that connects.
>
>
>
>
>
>
> On Mon, Oct 11, 2010 at 10:21 PM, Nyamul Hassan <mnhassan at usa.net> wrote:
> > That file lists the port that FS listens on for event socket connections.
> >
> > Run a "netstat -nlp | grep <port>" to check which IP that port is
> > being listened on.
> >
> > Then you can use command line switches for fs_cli to force it to
> > connect on that IP/port.
> >
> > Regards
> > HASSAN
> >
> >
> > On 2010-10-12, Peter Schrock <peter.schrock at gmail.com> wrote:
> >> just to make sure what am I comparing event_socket.conf.xml to? I have
> found
> >> that it is set up with the default settings but am not sure how or to
> what I
> >> am suppose to fix. I understand that I am comparing it to FS, but I am
> >> wondering if there is some kind of file that I am looking for to compare
> the
> >> two.
> >>
> >> PeterS
> >>
> >> On Fri, Oct 8, 2010 at 4:56 AM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >>
> >>> 1. Check FS is running
> >>> 2. Check the port you're connecting to matches the one in
> >>> event_socket.conf.xml
> >>> 3. Check that event_socket.conf.xml binding to the same IP you're
> >>> connecting to (e.g. 0.0.0.0 if you're connecting from a remote
> >>> machine, since the default 127.0.0.1 won't work then)
> >>> 4. Use netstat on the FS server to verify FreeSWITCH is actually
> >>> listening on the port you're trying to connect to (I've had a syntax
> >>> error in the config file make the module fail to load in the past
> >>> which left FS running but with no ESL socket). (If the module fails to
> >>> load there'll also be an error in the log file).
> >>> 5. Check a firewall isn't blocking access to the port
> >>>
> >>> -Steve
> >>>
> >>>
> >>>
> >>> On 7 October 2010 01:23, Peter Schrock <peter.schrock at gmail.com>
> wrote:
> >>> > Okay, so I managed to get FS working the other day and I even managed
> to
> >>> > test a call and test my voicemail. All seemed to be working smoothly
> >>> until,
> >>> > because of the rain here, my power went out and I had to reboot my
> >>> computer.
> >>> > I logged in through the terminal, set up FS in background went to
> fs_cli
> >>> and
> >>> > I get this error message:
> >>> > [ERROR] libs/esl/fs_cli.c:1206 main() Error Connecting [Socket
> >>> > Connection
> >>> > Error]
> >>> > I managed to figure out that in the file "fs_cli.c" on line 1206 you
> >>> > find
> >>> > the code for displaying this error message. The problem is that I
> don't
> >>> know
> >>> > why this error message is occurring. Does anyone have any helpful
> hints
> >>> as
> >>> > to what I should look at to resolve this problem?
> >>> > I even tried going to the git tree and make current, but that gave me
> >>> > problems that forced me to turn off mod_spandsp and mod_skyopen in
> the
> >>> > modules.conf, which I had running earlier. Any thoughts?
> >>> > PeterS
> >>> > PS
> >>> > I am not sure if this is of any help, but in addition to the error
> line
> >>> > above, it also posted this info:
> >>> > Usage: fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x
> >>> > command] [profile]
> >>> >   -?,-h --help                    Usage Information
> >>> >   -H, --host=hostname             Host to connect
> >>> >   -P, --port=port                 Port to connect (1 - 65535)
> >>> >   -u, --user=user at domain          user at domain
> >>> >   -p, --password=password         Password
> >>> >   -x, --execute=command           Execute Command and Exit
> >>> >   -l, --loglevel=command          Log Level
> >>> >   -q, --quiet                     Disable logging
> >>> >   -r, --retry                     Retry connection on failure
> >>> >   -R, --reconnect                 Reconnect if disconnected
> >>> >   -d, --debug=level               Debug Level (0 - 7)
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >
> > --
> > Sent from my mobile device
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Tue Oct 12 18:34:58 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Oct 2010 20:34:58 -0500
Subject: [Freeswitch-users] Can uPNP be turned off?
In-Reply-To: <4CB4FBE2.30903@mgtech.com>
References: <4CB4FBE2.30903@mgtech.com>
Message-ID: <AANLkTi=8XEG-r=0aD3kchSvGcoFECGqJb31ZVgMQz1Yb@mail.gmail.com>

Start fs with -nonat
On Oct 12, 2010 7:27 PM, "Mario" <mario_fs at mgtech.com> wrote:
> I am using ext... autonat:1.2.3.4 in sip internal and external as per
> wiki. The sofia status is ok for the profiles bu the nat_map shows the
> wrong address because the router is sending the wrong wan IP in the pnp
> request. Since I have autonat:1.2.3.4 coded is there a way to force FS
> not to use uPNP so it does not request nat info from the router? I need
> it on in the router for other things. Thanks. Mario
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Tue Oct 12 19:28:42 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Oct 2010 21:28:42 -0500
Subject: [Freeswitch-users] Can uPNP be turned off?
In-Reply-To: <4CB4FBE2.30903@mgtech.com>
References: <4CB4FBE2.30903@mgtech.com>
Message-ID: <AANLkTikLaD2BWiiySGy0cSfAVA0q-GwodWrjF6BQ1U28@mail.gmail.com>

Start fs with -nonat
On Oct 12, 2010 7:27 PM, "Mario" <mario_fs at mgtech.com> wrote:
> I am using ext... autonat:1.2.3.4 in sip internal and external as per
> wiki. The sofia status is ok for the profiles bu the nat_map shows the
> wrong address because the router is sending the wrong wan IP in the pnp
> request. Since I have autonat:1.2.3.4 coded is there a way to force FS
> not to use uPNP so it does not request nat info from the router? I need
> it on in the router for other things. Thanks. Mario
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From babak.freeswitch at gmail.com  Tue Oct 12 23:37:03 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Wed, 13 Oct 2010 10:07:03 +0330
Subject: [Freeswitch-users] mod_managed socket binding problem
In-Reply-To: <D51C1FE5-C34E-4B04-95AF-1D84B6AAFCA0@freeswitch.org>
References: <AANLkTinqR-8GTQ7qAOM=u5TFEoDbX__qAt7p8V++D06c@mail.gmail.com>
	<D51C1FE5-C34E-4B04-95AF-1D84B6AAFCA0@freeswitch.org>
Message-ID: <AANLkTimn5wPsR18CNxXPV0kzG4Jm1ayQ3j1=QUpuUsvY@mail.gmail.com>

you r right. the problem is I can not find anyway to set shut down method in
my module load. I tried to use DomainUnload notification but it is invoked
after the new instance is loaded. I know in c it's like
SWITCH_MODULE_DEFINITION but how can I do the same in c# (mod_managed)
thanx
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From slim at thegreek.com  Wed Oct 13 03:23:22 2010
From: slim at thegreek.com (Jeroen C. van Gelderen)
Date: Wed, 13 Oct 2010 05:23:22 -0500
Subject: [Freeswitch-users] tone_detect vs. ignore_early_mediaANDmod_fifo
In-Reply-To: <AANLkTinun0CHg5+yjEkZuC=jFNMQUUvqhhqqA_GqWG0L@mail.gmail.com>
References: <429823D0454A4EB4AE179DF13803F9B9@mbnet.local><AANLkTim1B_oxbCAXNPTHNoZxh6vUfbUiWsyHzxG=xwif@mail.gmail.com><DAFF983A4D6A43A186D8662FE5876375@mbnet.local><515E6743DC69484A8E3514E807B24353@mbnet.local><D33D5C7C9BAE42CA96354582D25F8B21@mbnet.local><AANLkTimea0_Gs6jcc2ic=pYFaxFhKPAErgPnRbQnZva0@mail.gmail.com>
	<AANLkTinun0CHg5+yjEkZuC=jFNMQUUvqhhqqA_GqWG0L@mail.gmail.com>
Message-ID: <E6B113A909994D269D44050C25BD2A09@mbnet.local>


I was just going to respond to your previous mail: I have commented out the
pause/resume statements and this solved my problem.

I will try the latest git when I get back in office. Thank you.

Cheers,
-Slim
--
Jeroen C. "Slim" van Gelderen

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Tuesday, October 12, 2010 12:34
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] tone_detect vs.
ignore_early_mediaANDmod_fifo

try latest GIT

On Mon, Oct 11, 2010 at 4:00 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> This was done to stop recordings and other things on callers waiting in a
queue.
> We may have to change it to be configurable.
>
>
> On Mon, Oct 11, 2010 at 6:51 AM, Jeroen C. van Gelderen
> <slim at thegreek.com> wrote:
>>
>>
>> I have an inkling that the following commit made between 1.0.2 and 1.0.3
>> might have something to do with this:
>>
>>
>>
>> ?? * mod_fifo: pause media bugs while not in a bridge (r:11466,11490)
>>
>>
>>
>>
http://lists.freeswitch.org/pipermail/freeswitch-svn/2009-January/009566..ht
ml
>>
>>
http://lists.freeswitch.org/pipermail/freeswitch-svn/2009-January/009589..ht
ml
>>
>>
>>
>> Can anyone comment on the how and why? Is there any way to reconcile this
>> with use of tone_detect?
>>
>> Cheers,
>> -Slim
>> --
>> Jeroen C. "Slim" van Gelderen
>> Olympic Sports Data Services
>> Email: slim at thegreek.com
>> Phone: +1 876 953 6182 x128
>>
>> ________________________________
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Jeroen
>> C. van Gelderen
>> Sent: Monday, October 11, 2010 05:31
>> To: 'FreeSWITCH Users Help'
>> Subject: Re: [Freeswitch-users] tone_detect vs. ignore_early_media
>> ANDmod_fifo
>>
>>
>>
>>
>>
>> Hmm
 the plot thickens

>>
>>
>>
>> When I bridge my FXO port to the SIP-GSM gateway directly (i.e. without
>> using mod_fifo) I don?t seem to need monitor_early_media_fail. Using
>> tone_detect on the A leg works fine when ?ignore_early_media=true? is
used
>> on the B leg:
>>
>>
>>
>> The following dialplan excerpt WORKS (i.e. FXO hang-ups are detected at
all
>> stages by tone_detect):
>>
>>
>>
>> ??? <extension name="pan_125">
>>
>> ????? <condition field="${freetdm_span_name}"
>> expression="^span_fxo_helpdesk$">
>>
>> ??????? <!-- Detect busy tone from Panasonic -->
>>
>> ??????? <action application="tone_detect" data="busy-pan 350,440 r 0
hangup
>> undef 2"/>
>>
>>
>>
>> ??????? <action application="answer"/>
>>
>>
>>
>> ??????? <action application="set" data="continue_on_fail=true"/>
>>
>> ??????? <action application="set" data="instant_ringback=true"/>
>>
>> ??????? <action application="set"
data="ringback=%(1000,1500,440.0,480.0)"/>
>>
>>
>>
>> ??????? <action application="bridge"
>>
data="{ignore_early_media=true}sofia/internal/1??????76??@192.168.3.11:5060"
/>
>>
>> ??????? <action application="bridge"
>>
data="{ignore_early_media=true}sofia/internal/1??????77??@192.168.3.11:5060"
/>
>>
>> ????? </condition>
>>
>> ??? </extension>
>>
>>
>>
>> The problem seems to occur only when mod_fifo is added to the mix:
>>
>>
>>
>> ??? <extension name="test2">
>>
>> ????? <condition field="${freetdm_span_name}"
expression="^span_fxo_test$">
>>
>> ??????? <!-- Detect busy tone from Panasonic -->
>>
>> ??????? <action application="tone_detect" data="busy-pan 350,440 r 0
hangup
>> undef 2"/>
>>
>>
>>
>> ??????? <action application="answer"/>
>>
>>
>>
>> ??????? <action application="playback" data="thankyou.wav"/>
>>
>> ??????? <action application="set" data="fifo_music=$${hold_music}"/>
>>
>> ??????? <action application="fifo" data="fifo_helpdesk in"/>
>>
>> ????? </condition>
>>
>> ??? </extension>
>>
>>
>>
>> In all cases:
>>
>> -????????? tone_detect works BEFORE the call is handed to mod_fifo (i.e.
>> during playback)
>>
>> -????????? tone_detect works AFTER the call is established by mod_fifo
and
>> audio is being exchanged between A and B leg.
>>
>>
>>
>> IF the A leg (with tone_detect enabled) is handed to mod_fifo AND there
are
>> no agents in the fifo THEN tone_detect does not work. Mod_fifo simply
plays
>> MOH to the A leg perpetually.
>>
>>
>>
>> IF the A leg (with tone_detect enabled) is handed to mod_fifo AND
mod_fifo
>> places an outbound call with ?ignore_early_media=true? THEN tone_detect
does
>> not work on the A leg during the early media phase on leg B.
>>
>>
>>
>> IF the A leg (with tone_detect enabled) is handed to mod_fifo AND
mod_fifo
>> places an outbound call with ?ignore_early_media=false? THEN tone_detect
>> does work. I guess this is because audio is being exchanged between A and
B
>> legs.
>>
>>
>>
>> Any idea what would cause the tone_detect to be ?suspended? when mod_fifo
is
>> in the mix?
>>
>> Cheers,
>> -Slim
>> --
>> Jeroen C. "Slim" van Gelderen
>>
>> ________________________________
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Jeroen
>> C. van Gelderen
>> Sent: Monday, October 11, 2010 03:13
>> To: 'FreeSWITCH Users Help'
>> Subject: Re: [Freeswitch-users] tone_detect vs. ignore_early_media (per
>> leg?)
>>
>>
>>
>> Thank you for the quick response.
>>
>>
>>
>> It looks like monitor_early_media_fail should do what I need (thanks for
the
>> suggestion!) but I can?t seem to make it work. See below for my
uneducated
>> best guess why.
>>
>>
>>
>> In my case the failure conditions from the GSM side are handled
out-of-band
>> by SIP. That leaves only one failure condition I need to listen for in
early
>> media (the Panasonic far-end hang up on FXO) which can successfully be
>> detected with:
>>
>>
>>
>> ? <action application="tone_detect" data="busy-pan 350,440 r 0 hangup>
>>
>>
>>
>> This results in the following log entries:
>>
>>
>>
>> [DEBUG] switch_ivr_async.c:2072 Adding tone spec 350,440 index 0 hits 1
>>
>> [DEBUG] switch_core_media_bug.c:360 Attaching BUG to FreeTDM/4:1/
>>
>>
>>
>> Translating tone_detect to monitor_early_media_fail I tried the
following:
>>
>>
>>
>> ? <action application="set" data="result=${fifo_member(add fifo_helpdesk
>>
{fifo_member_wait=nowait,ignore_early_media=true,monitor_early_media_fail=bu
sy-pan:1:350+440}sofia/internal/${caller_id_number}@192.168.3.11:5060)}"/>
>>
>>
>>
>> This results in the following relevant log entries:
>>
>>
>>
>> [DEBUG] switch_channel.c:2444
(sofia/internal/1876??????@192.168.3.11:5060)
>> Callstate Change RINGING -> EARLY
>>
>> [DEBUG] switch_ivr_async.c:2072 Adding tone spec 350,440 index 0 hits 1
>>
>> [DEBUG] switch_core_media_bug.c:360 Attaching BUG to
>> sofia/internal/1??????7693 at 192.168.3.11:5060
>>
>>
>>
>> So Freeswitch is listening for the right tones (tone_spec present and
>> identical to the one used in tone_detect approach) but it isn?t detecting
>> them. The obvious difference is in the BUG attachment.
>>
>>
>>
>> Is it possible that BUG isn?t listening to the right (A) leg in the case
of
>> monitor_early_media_fail? Or is this too easy? J
>>
>> Cheers,
>> -Slim
>> --
>> Jeroen C. "Slim" van Gelderen
>>
>> ________________________________
>>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael
>> Collins
>> Sent: Sunday, October 10, 2010 19:01
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] tone_detect vs. ignore_early_media (per
>> leg?)
>>
>>
>>
>> Check out monitor_early_media_fail:
>>
http://wiki.freeswitch.org/wiki/Channel_Variables#monitor_early_media_fail
>>
>> It is a hybrid of ignoring early media and listening to early media for
>> various conditions. In your case you'll need to play around with it. In
your
>> case you need to figure out which early media scenarios count as a "fail"
>> and will cause processing to move on as if the call really did fail.
>>
>> Roll up your sleeves, you have some work to do. :)
>>
>> -MC
>>
>> On Sat, Oct 9, 2010 at 9:24 PM, Jeroen C. van Gelderen
<slim at thegreek.com>
>> wrote:
>>
>> Hi Guys,
>>
>> I have a legacy Panasonic PBX which does not support CPC/Disconnect
>> Supervision. Calls from this PBX are sent to Freeswitch by way of
>> DAHDI/FreeTDM FXO ports (on a Xorcom Astribank). I use tone_detect to
deal
>> with far-end hang-ups. This works fine:
>>
>> <!-- Detect busy tone from Panasonic -->
>> <action application="tone_detect" data="busy 350,440 r 0 hangup 4"/>
>>
>> Some calls from the Panasonic PBX are put in a FIFO and from there they
are
>> sent to a SIP-to-GSM bridge (GoIP) to be terminated on one of 3 GSM
phones
>> in a round-robin fashion.
>>
>> The GSM bridge requires me to use "ignore_early_media=true" otherwise the
>> caller will receive messages like "the number you are calling does not
>> answer". When I set "ignore_early_media=true" the FIFO correctly keeps
>> hunting for a GSM phone that is actually answered and will ignore phones
>> that are busy, no-answer or turned off. This too works fine.
>>
>> The problem occurs when the two are combined as follows:
>>
>> Panasonic PBX --FXO--> Freeswitch --SIP--> GSM-bridge
>>
>> If I enable ignore_early_media then tone_detect doesn't work UNTIL one of
>> the GSMs is answered. This is a problem when none of the GSMs are
answered
>> and the Panasonic PBX hangs up. The FXO port will stay "up" and the FIFO
>> will keep hunting until it times out.
>>
>> If I don't use "ignore_early_media=true" then tone_detect works fine but
>> then every telco message gets mistaken for an answered call and the
hunting
>> stops early.
>>
>> I tried changing this example line from my fifo.conf:
>>
>> <member timeout="30" simo="1" lag="5">
>>
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7701 at 192.1
>> 68.3.11:5060
>> </member>
>>
>> to
>>
>> <member timeout="30" simo="1" lag="5">
>>
{member_wait=nowait}[ignore_early_media=true]sofia/internal/1??????7701 at 192.
>> 168.3.11:5060
>> </member>
>>
>> in a vain attempt to ignore early media on the SIP leg only. This doesn't
>> seem to do anything however.
>>
>> Can anyone clue me in on what I'm missing? I've snipped the relevant
>> configuration bits below. I have the feeling I'm missing something
obvious.
>>
>> Cheers,
>> -Slim
>>
>> ----8<----8<----8<----8<----8<----
>>
>> <configuration name="fifo.conf" description="FIFO Configuration">
>> ?<settings>
>> ? ?<param name="delete-all-outbound-member-on-startup" value="false"/>
>> ?</settings>
>> ?<fifos>
>> ? ?<fifo name="cool_fifo@$${domain}" importance="0">
>>
>> ? ? ?<member timeout="30" simo="1" lag="5">
>>
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7701 at 192.1
>> 68.3.11:5060</member>
>>
>> ? ? ?<member timeout="30" simo="1" lag="5">
>>
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7693 at 192.1
>> 68.3.11:5060</member>
>>
>> ? ? ?<member timeout="30" simo="1" lag="5">
>>
{member_wait=nowait,ignore_early_media=true}sofia/internal/1??????7698 at 192.1
>> 68.3.11:5060</member>
>>
>> ? ?</fifo>
>> ?</fifos>
>> </configuration>
>>
>> ----8<----8<----8<----8<----8<----
>>
>> <include>
>> ?<context name="pbx_test">
>>
>> ? ?<extension name="test1">
>> ? ? ?<condition field="${freetdm_span_name}"
expression="^span_fxo_test$">
>> ? ? ? ?<action application="answer"/>
>>
>> ? ? ? ?<!-- Detect busy tone from Panasonic -->
>> ? ? ? ?<action application="tone_detect" data="busy 350,440 r 0 hangup
4"/>
>>
>> ? ? ? ?<action application="playback"
>> data="ivr/ivr-hold_connect_call.wav"/>
>>
>> ? ? ? ?<action application="fifo" data="cool_fifo@$${domain} in"/>
>>
>> ? ? ?</condition>
>> ? ?</extension>
>> ?</context>
>> </include>
>>
>> ----8<----8<----8<----8<----8<----
>>
>> Cheers,
>> -Slim
>> --
>> Jeroen C. "Slim" van Gelderen
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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
>



-- 
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

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



From slim at thegreek.com  Wed Oct 13 03:28:54 2010
From: slim at thegreek.com (Jeroen C. van Gelderen)
Date: Wed, 13 Oct 2010 05:28:54 -0500
Subject: [Freeswitch-users] skip/exit mod_fifo when no agents?
Message-ID: <2D766595C13E45688CE1E5C8888F3693@mbnet.local>

Hi,

Is there a proper way to drop callers out of mod_fifo when no agents are
logged in? When no agents are logged in I'd like to proceed in the dialplan
escalating the call to a backup number and/or backup fifo. Almost like I
need a nowait option for callers.

Cheers,
-Slim
--
Jeroen C. "Slim" van Gelderen




From andy at fabulous4.co.uk  Wed Oct 13 03:42:32 2010
From: andy at fabulous4.co.uk (Andy)
Date: Wed, 13 Oct 2010 11:42:32 +0100
Subject: [Freeswitch-users] record on demand for a conference call
Message-ID: <6DFF80D26E14467299C2B7789D1D82D6@D810>

Hi,
 
I would like to set up my conferencing system so that it's possible to
initiate the recording of the conference when the moderator presses a key on
their keypad.
 
Failing that I would like to know if there is any way of recording the
entire conference call from the moment the first caller arrives rather than
using the auto-record feature which only kicks in when 2 or more people are
connected.
 
Can anyone give me some idea how this can be done or point me to the right
wiki pages?
 
Many thanks for you help
Andy
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From xyangni at gmail.com  Wed Oct 13 05:33:32 2010
From: xyangni at gmail.com (xuyan yang)
Date: Wed, 13 Oct 2010 20:33:32 +0800
Subject: [Freeswitch-users] iLBC codec generating only noise.
In-Reply-To: <AANLkTimiSPgCzs31VjEp0x8g1qDe-kzppsDM7uig0YGU@mail.gmail.com>
References: <AANLkTinz4JEfyGnjU-4nneOFg=A0qXHjVw9xE=jp2qSo@mail.gmail.com>
	<AANLkTinvkxRaS+Wmso1vubuH3Pj4Mib96B32Bx+sqR7H@mail.gmail.com>
	<AANLkTimmMM8Ua9AP9BwXU0yDfP9FReK2e9w-iMRiMVFu@mail.gmail.com>
	<AANLkTimiSPgCzs31VjEp0x8g1qDe-kzppsDM7uig0YGU@mail.gmail.com>
Message-ID: <AANLkTikFDu0xC=VcXpzJ0_8i=NL-QfgBm5ms7nMftRzu@mail.gmail.com>

It seems to be a windows only problem.
All my test start from git clone, so it is clean. The win xp version do have
the described problem, but the ubuntu platform do not.


On Mon, Oct 11, 2010 at 10:59 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> you can't mix ptimes in your codec list, they will be filtered out.
> I can not reproduce any problems on latest GIT, have you done a complete
> build?
>
> On Sat, Oct 9, 2010 at 7:43 AM, xuyan yang <xyangni at gmail.com> wrote:
> > Hi Anthony,
> > Tried the latest, it works when both lag A and B are using iLBC. But
> there
> > are still some problems:
> > 1, when lag A use PCMU and lag B use iLBC or the reverse, the PCMU side
> hear
> > only noise even if it is muted.
> > 2, when added "iLBC at 30i" to codec list, it is ignored. traced the debug,
> > iLBC is not compared as a choice. So adding only "iLBC" should be used as
> > mitigation.
> >
> >
> >
> > On Sat, Oct 9, 2010 at 11:34 AM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> Try latest
> >>
> >> On Oct 8, 2010 2:28 PM, "xuyan yang" <xyangni at gmail.com> wrote:
> >> > Hi,
> >> >
> >> > I am trying to use iLBC codec with both eyebeam and some iphone
> client.
> >> > When
> >> > a ivr is called, the client can here system voice and make dtmf input.
> >> > but
> >> > the voice recorded from client's microphone is only noise.
> >> > The call between 2 clients also have such problem.
> >> >
> >> > Sometimes, FS may even got crashed with the following information:
> >> >
> >> > 2010-10-08 19:56:21.343750 [INFO] switch_rtp.c:2527 Auto Changing port
> >> > from
> >> > 10.20.132.244:18570 to 8
> >> > 2.132.139.197:19536
> >> > alloc: asked for negative size -2147483648
> >> >
> >> > in var.xml, I have added iLBC instead of iLBC at 30i, because iLBC at 30iis
> >> > ignored during codec negotiation.
> >> > <X-PRE-PROCESS cmd="set"
> >> > data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h
> >> > ,G722,speex at 8000h@20i,iLBC,PCMU,PCMA,GSM"/>
> >> >
> >> > The git version last week is used in my test. Is there anything wrong
> >> > with
> >> > my setup? Thanks.
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From michofr at hotmail.com  Wed Oct 13 00:46:18 2010
From: michofr at hotmail.com (micho fr)
Date: Wed, 13 Oct 2010 07:46:18 +0000
Subject: [Freeswitch-users] Use Freeswitch
Message-ID: <BAY147-w1466287814D00201AFE0A0BB550@phx.gbl>


Dear All,

I'm planning to use the freeswitch in my network but I have some doubt regarding some features and need your help please before taking my decision...
I need to use the freeswitch as a registrar server + Sip Proxy...I need to know the below:

1- Does freeswitch supports encrypted SIP inside UDP
2- Does freeswitch supports TCP
3- How much TCP connections can handle?

Regards
 		 	   		  
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From emd at hiredminds.com  Tue Oct 12 21:13:33 2010
From: emd at hiredminds.com (Erik M. Devane)
Date: Wed, 13 Oct 2010 13:13:33 +0900
Subject: [Freeswitch-users] Multiple Soundcard channels
In-Reply-To: <AANLkTi=8XEG-r=0aD3kchSvGcoFECGqJb31ZVgMQz1Yb@mail.gmail.com>
References: <4CB4FBE2.30903@mgtech.com>
	<AANLkTi=8XEG-r=0aD3kchSvGcoFECGqJb31ZVgMQz1Yb@mail.gmail.com>
Message-ID: <6E741949-743A-48B5-92EF-FA19D5966528@hiredminds.com>

Can I start multiple instances of mod-portaudio?

I would like to have multiple channels from my alsa-compliant soundcard but I can't find a way to run multiple channels in FreeSWITCH. 

I am connecting to a PABX and need to bridge calls.

Am I missing something obvious?

Thank you,

Erik

From peter.olsson at visionutveckling.se  Wed Oct 13 06:51:29 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 13 Oct 2010 15:51:29 +0200
Subject: [Freeswitch-users] Use Freeswitch
In-Reply-To: <BAY147-w1466287814D00201AFE0A0BB550@phx.gbl>
References: <BAY147-w1466287814D00201AFE0A0BB550@phx.gbl>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57E5E8E65E@cooper>

1.       No, as far as I know there is no such thing as TLS over UDP? Someone correct me if I'm wrong.

2.       Yes.

3.       Depends on your OS/hardware I guess - but FS itself is quite scalable.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r micho fr
Skickat: den 13 oktober 2010 09:46
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Use Freeswitch

Dear All,

I'm planning to use the freeswitch in my network but I have some doubt regarding some features and need your help please before taking my decision...
I need to use the freeswitch as a registrar server + Sip Proxy...I need to know the below:

1- Does freeswitch supports encrypted SIP inside UDP
2- Does freeswitch supports TCP
3- How much TCP connections can handle?

Regards
!DSPAM:4cb5b6b632931939610325!
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From mario_fs at mgtech.com  Wed Oct 13 07:17:14 2010
From: mario_fs at mgtech.com (Mario)
Date: Wed, 13 Oct 2010 07:17:14 -0700
Subject: [Freeswitch-users] Can uPNP be turned off?
In-Reply-To: <AANLkTikLaD2BWiiySGy0cSfAVA0q-GwodWrjF6BQ1U28@mail.gmail.com>
References: <4CB4FBE2.30903@mgtech.com>
	<AANLkTikLaD2BWiiySGy0cSfAVA0q-GwodWrjF6BQ1U28@mail.gmail.com>
Message-ID: <4CB5BF6A.308@mgtech.com>

Thanks, I saw that in the wiki and was going to try it next. But after
adding autonat:1.2.3.4 looks like last nights test is ok now, I am
hoping the problem is fixed. FS would work fine first 4-9 hours then
stopped answering incoming. Funny thing, no problem like this with the
SPA9000 for 2 years using 2 DSL connections. Thanks again.

On 10/12/10 19:28, Anthony Minessale wrote:
> Start fs with -nonat
> 
> On Oct 12, 2010 7:27 PM, "Mario" <mario_fs at mgtech.com
> <mailto:mario_fs at mgtech.com>> wrote:
>> I am using ext... autonat:1.2.3.4 in sip internal and external as per
>> wiki. The sofia status is ok for the profiles bu the nat_map shows the
>> wrong address because the router is sending the wrong wan IP in the pnp
>> request. Since I have autonat:1.2.3.4 coded is there a way to force FS
>> not to use uPNP so it does not request nat info from the router? I need
>> it on in the router for other things. Thanks. Mario
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From dujinfang at gmail.com  Wed Oct 13 07:20:42 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 13 Oct 2010 22:20:42 +0800
Subject: [Freeswitch-users] skip/exit mod_fifo when no agents?
In-Reply-To: <2D766595C13E45688CE1E5C8888F3693@mbnet.local>
References: <2D766595C13E45688CE1E5C8888F3693@mbnet.local>
Message-ID: <AANLkTim_b+xBK2nGM5EeOPriksu1W3_t1XhX4obwFs0K@mail.gmail.com>

check the nowait param

On Wed, Oct 13, 2010 at 6:28 PM, Jeroen C. van Gelderen
<slim at thegreek.com> wrote:
> Hi,
>
> Is there a proper way to drop callers out of mod_fifo when no agents are
> logged in? When no agents are logged in I'd like to proceed in the dialplan
> escalating the call to a backup number and/or backup fifo. Almost like I
> need a nowait option for callers.
>
> Cheers,
> -Slim
> --
> Jeroen C. "Slim" van Gelderen
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From dujinfang at gmail.com  Wed Oct 13 07:22:46 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 13 Oct 2010 22:22:46 +0800
Subject: [Freeswitch-users] record on demand for a conference call
In-Reply-To: <6DFF80D26E14467299C2B7789D1D82D6@D810>
References: <6DFF80D26E14467299C2B7789D1D82D6@D810>
Message-ID: <AANLkTimcHmXTkoyetBVjjVtQT7KXPO=p7RXQqQO0veZM@mail.gmail.com>

I'm not sure, but might be possible to use the sched_api or sched_app feature.

On Wed, Oct 13, 2010 at 6:42 PM, Andy <andy at fabulous4.co.uk> wrote:
> Hi,
>
> I would like to set up my conferencing system so that it's possible to
> initiate the recording of the conference when?the moderator presses a key on
> their keypad.
>
> Failing that I would like to know if there is any way of recording the
> entire conference call from the moment the first caller arrives rather than
> using the auto-record feature which only kicks in when 2 or more people are
> connected.
>
> Can anyone give me some idea how this can be done or point me to the right
> wiki pages?
>
> Many thanks for you help
> Andy
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From dujinfang at gmail.com  Wed Oct 13 07:31:50 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 13 Oct 2010 22:31:50 +0800
Subject: [Freeswitch-users] iLBC codec generating only noise.
In-Reply-To: <AANLkTikFDu0xC=VcXpzJ0_8i=NL-QfgBm5ms7nMftRzu@mail.gmail.com>
References: <AANLkTinz4JEfyGnjU-4nneOFg=A0qXHjVw9xE=jp2qSo@mail.gmail.com>
	<AANLkTinvkxRaS+Wmso1vubuH3Pj4Mib96B32Bx+sqR7H@mail.gmail.com>
	<AANLkTimmMM8Ua9AP9BwXU0yDfP9FReK2e9w-iMRiMVFu@mail.gmail.com>
	<AANLkTimiSPgCzs31VjEp0x8g1qDe-kzppsDM7uig0YGU@mail.gmail.com>
	<AANLkTikFDu0xC=VcXpzJ0_8i=NL-QfgBm5ms7nMftRzu@mail.gmail.com>
Message-ID: <AANLkTi=6Esu59uFuRMvxd-RQ6B_fiSAkXS_Qx4Zu-sfF@mail.gmail.com>

I also experienced similar problem a few days ago, haven't try the latest git.

It's a client based on FSComm, when use iLBC at 30i talk to a FS server,
I hear garbage. It's only happend on windows. The same code works on
Mac without problem.

I give up iLBC on windows then.

On Wed, Oct 13, 2010 at 8:33 PM, xuyan yang <xyangni at gmail.com> wrote:
> It seems to be a windows only problem.
> All my test start from git clone, so it is clean. The win xp version do have
> the described problem, but the ubuntu platform do not.
>
> On Mon, Oct 11, 2010 at 10:59 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> you can't mix ptimes in your codec list, they will be filtered out.
>> I can not reproduce any problems on latest GIT, have you done a complete
>> build?
>>
>> On Sat, Oct 9, 2010 at 7:43 AM, xuyan yang <xyangni at gmail.com> wrote:
>> > Hi?Anthony,
>> > Tried the latest, it works when both lag A and B are using iLBC. But
>> > there
>> > are still some problems:
>> > 1, when lag A use PCMU and lag B use iLBC or the reverse, the PCMU side
>> > hear
>> > only noise even if it is muted.
>> > 2, when added "iLBC at 30i" to codec list, it is ignored. traced the debug,
>> > iLBC is not compared as a choice. So adding only "iLBC" should be used
>> > as
>> > mitigation.
>> >
>> >
>> >
>> > On Sat, Oct 9, 2010 at 11:34 AM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> Try latest
>> >>
>> >> On Oct 8, 2010 2:28 PM, "xuyan yang" <xyangni at gmail.com> wrote:
>> >> > Hi,
>> >> >
>> >> > I am trying to use iLBC codec with both eyebeam and some iphone
>> >> > client.
>> >> > When
>> >> > a ivr is called, the client can here system voice and make dtmf
>> >> > input.
>> >> > but
>> >> > the voice recorded from client's microphone is only noise.
>> >> > The call between 2 clients also have such problem.
>> >> >
>> >> > Sometimes, FS may even got crashed with the following information:
>> >> >
>> >> > 2010-10-08 19:56:21.343750 [INFO] switch_rtp.c:2527 Auto Changing
>> >> > port
>> >> > from
>> >> > 10.20.132.244:18570 to 8
>> >> > 2.132.139.197:19536
>> >> > alloc: asked for negative size -2147483648
>> >> >
>> >> > in var.xml, I have added iLBC instead of iLBC at 30i, because iLBC at 30i
>> >> > is
>> >> > ignored during codec negotiation.
>> >> > <X-PRE-PROCESS cmd="set"
>> >> > data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h
>> >> > ,G722,speex at 8000h@20i,iLBC,PCMU,PCMA,GSM"/>
>> >> >
>> >> > The git version last week is used in my test. Is there anything wrong
>> >> > with
>> >> > my setup? Thanks.
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From msc at freeswitch.org  Wed Oct 13 07:58:35 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 13 Oct 2010 07:58:35 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today - Moises Silva
	From Sangoma Speaking
Message-ID: <AANLkTin9f5ku11Nhivs5=KQe4rDa4bUr2FTM7tLQpS6k@mail.gmail.com>

Hey all!

We have a good agenda for today:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_10_13

Moises will be in to talk about FreeTDM and where we stand. If you have
FreeTDM questions please join and Moises will answer them. We also have a
few new FreeSWITCH features to discuss.

Talk to you in a few hours!

-Michael
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From msc at freeswitch.org  Wed Oct 13 08:00:18 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 13 Oct 2010 08:00:18 -0700
Subject: [Freeswitch-users] Use Freeswitch
In-Reply-To: <BAY147-w1466287814D00201AFE0A0BB550@phx.gbl>
References: <BAY147-w1466287814D00201AFE0A0BB550@phx.gbl>
Message-ID: <AANLkTimWCok2E5wfAN_WZCNLP6MHZtwedhL_hzO6QOGj@mail.gmail.com>

FreeSWITCH is not a proxy, it is a B2BUA. If you need a true SIP proxy then
you want OpenSIPS or Kamailio.
-MC

On Wed, Oct 13, 2010 at 12:46 AM, micho fr <michofr at hotmail.com> wrote:

>  Dear All,
>
> I'm planning to use the freeswitch in my network but I have some doubt
> regarding some features and need your help please before taking my
> decision...
> I need to use the freeswitch as a registrar server + Sip Proxy...I need to
> know the below:
>
> 1- Does freeswitch supports encrypted SIP inside UDP
> 2- Does freeswitch supports TCP
> 3- How much TCP connections can handle?
>
> Regards
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From john at 247-talk.co.uk  Wed Oct 13 05:14:42 2010
From: john at 247-talk.co.uk (John Carpenter)
Date: Wed, 13 Oct 2010 13:14:42 +0100
Subject: [Freeswitch-users] Mod_h323 problem
Message-ID: <1286972082.1495.19.camel@Zaphod.gateway.2wire.net>

Hi, I am trying to bridge an incoming h323 call to an external SIP
provider, I am using latest git release. If I use G729 codec I get
message

"Unsupported ptime of 2 on  write Audio codec G.729{sw}  for connection
[0xb4c0ee10]"

and call fails to connect. If I use ulaw codec call fails to connect
with

"Write PDU fail: no control channel"

If I make a straight SIP to SIP call though same provider all work ok. I
have posted log of call in http://pastebin.freeswitch.org/14216 because
it is rather large.

my h323.conf.xml file looks like this

<configuration name="h323.conf" description="H323 Endpoints">
 <settings>
   <param name="trace-level" value="10"/>
   <param name="context" value="public"/>
   <param name="dialplan" value="XML"/>
   <param name="codec-prefs" value="PCMA,PCMU,GSM,G729"/>
   <param name="use-rtp-timer" value="true"/> <!-- enable RTP timer -
should always be enabled -->
   <param name="rtp-timer-name" value="soft"/> <!-- Timer name, soft is
default -->
   <!-- <param name="ptime-override-value" value="20"/> --> <!--
Override negotiated ptime value with this value -->
   <param name="gk-address" value=""/>    <!-- empty to disable, "*" to
search LAN -->
   <param name="gk-identifer" value=""/>  <!-- optional name of gk -->
   <param name="gk-interface" value=""/>  <!-- mandatory listener
interface name -->
   <param name="gk-retry" value="30"/>  <!-- optional GK register retry
timer -->
   <param name="faststart" value="true"/>  <!-- optional -->
   <param name="h245tunneling" value="true"/>  <!-- optional -->
   <param name="h245insetup" value="true"/>  <!-- optional -->
   <param name="jitter-size" value="60"/>  <!-- optional -->
   <param name="progress-indication" value="8"/> <!-- optional - PI
value in progress message-->
   <param name="alerting-indication" value="8"/> <!-- optional - PI
value in alerting message-->
   <param name="endpoint-name" value="fs"/>
   <param name="fax-old-asn" value="true"/>
 </settings>
 <listeners>
   <listener name="default">
     <param name="h323-ip" value="$${local_ip_v4}"/>
     <param name="h323-port" value="1720"/>
   </listener>
 </listeners>
</configuration>

And this is the log of the xml_curl dialplan that is executed

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="dialplan" description="php dialplan">
<context name="public">
<extension name="normal">
<condition field="destination_number" expression="^(\d+)$">
<action application="set" data="bypass_media=false"/>
<action application="set" data="proxy_media=true"/>
<action application="bridge" data="sofia/gateway/arbinet-o/$1"/>
</condition>
</extension>
</context>
</section>
</document>

Any help will be greatly appreciated 

regards, John Carpenter
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From john at 247-talk.co.uk  Wed Oct 13 07:59:42 2010
From: john at 247-talk.co.uk (John Carpenter)
Date: Wed, 13 Oct 2010 15:59:42 +0100
Subject: [Freeswitch-users]  H323 to Sip conversion problem
Message-ID: <1286981982.2305.4.camel@Zaphod>

Hi, I am trying to bridge an incoming h323 call to an external SIP
provider, I am using latest git release. If I use G729 codec I get
message

"Unsupported ptime of 2 on  write Audio codec G.729{sw}  for connection
[0xb4c0ee10]"

and call fails to connect. If I use ulaw codec call fails to connect
with

"Write PDU fail: no control channel"

If I make a straight SIP to SIP call though same provider all work ok. I
have posted log of call in http://pastebin.freeswitch.org/14216 because
it is rather large.
This is my first venture into using the mod_h323 module and maybe I am
doing something stupid but have read all docs and seem to have hit a
brick wall on this.

my h323.conf.xml file looks like this

<configuration name="h323.conf" description="H323 Endpoints">
<settings>
   <param name="trace-level" value="10"/>
   <param name="context" value="public"/>
   <param name="dialplan" value="XML"/>
   <param name="codec-prefs" value="PCMA,PCMU,GSM,G729"/>
   <param name="use-rtp-timer" value="true"/> <!-- enable RTP timer -
should always be enabled -->
   <param name="rtp-timer-name" value="soft"/> <!-- Timer name, soft is
default -->
   <!-- <param name="ptime-override-value" value="20"/> --> <!--
Override negotiated ptime value with this value -->
   <param name="gk-address" value=""/>    <!-- empty to disable, "*" to
search LAN -->
   <param name="gk-identifer" value=""/>  <!-- optional name of gk -->
   <param name="gk-interface" value=""/>  <!-- mandatory listener
interface name -->
   <param name="gk-retry" value="30"/>  <!-- optional GK register retry
timer -->
   <param name="faststart" value="true"/>  <!-- optional -->
   <param name="h245tunneling" value="true"/>  <!-- optional -->
   <param name="h245insetup" value="true"/>  <!-- optional -->
   <param name="jitter-size" value="60"/>  <!-- optional -->
   <param name="progress-indication" value="8"/> <!-- optional - PI
value in progress message-->
   <param name="alerting-indication" value="8"/> <!-- optional - PI
value in alerting message-->
   <param name="endpoint-name" value="fs"/>
   <param name="fax-old-asn" value="true"/>
</settings>
<listeners>
   <listener name="default">
     <param name="h323-ip" value="$${local_ip_v4}"/>
     <param name="h323-port" value="1720"/>
   </listener>
</listeners>
</configuration>

And this is the log of the xml_curl dialplan that is executed

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="dialplan" description="php dialplan">
<context name="public">
<extension name="normal">
<condition field="destination_number" expression="^(\d+)$">
<action application="set" data="bypass_media=false"/>
<action application="set" data="proxy_media=true"/>
<action application="bridge" data="sofia/gateway/arbinet-o/$1"/>
</condition>
</extension>
</context>
</section>
</document>

Any help will be greatly appreciated 

regards, John Carpenter
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From jrichey at itltd.net  Tue Oct 12 17:51:07 2010
From: jrichey at itltd.net (JRichey)
Date: Tue, 12 Oct 2010 17:51:07 -0700
Subject: [Freeswitch-users] Can uPNP be turned off?
Message-ID: <6ECAF1527329364583AB525CF34ABF950832197B@ms.kallback.com>

Starting FreeSwitch with "-nonat" (e.g. /usr/local/freeswitch/bin/freeswitch
-nc -nonat) will disable UPnP.

-Justin


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of Mario
Sent: Tuesday, October 12, 2010 5:23 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Can uPNP be turned off?


I am using ext... autonat:1.2.3.4 in sip internal and external as per
wiki. The sofia status is ok for the profiles bu the nat_map shows the
wrong address because the router is sending the wrong wan IP in the pnp
request. Since I have autonat:1.2.3.4 coded is there a way to force FS
not to use uPNP so it does not request nat info from the router? I need
it on in the router for other things. Thanks. Mario

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




From michofr at hotmail.com  Wed Oct 13 07:27:38 2010
From: michofr at hotmail.com (micho fr)
Date: Wed, 13 Oct 2010 14:27:38 +0000
Subject: [Freeswitch-users] Use Freeswitch
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57E5E8E65E@cooper>
References: <BAY147-w1466287814D00201AFE0A0BB550@phx.gbl>,
	<549CFEF87AEDE841A38E9D15EAB4C04C57E5E8E65E@cooper>
Message-ID: <BAY147-w824A7255BFD3D31448B27BB550@phx.gbl>


Sorry , it's not TLS over UDP...It's TLs over TCP...Does freeswitch support it?

From: peter.olsson at visionutveckling.se
To: freeswitch-users at lists.freeswitch.org
Date: Wed, 13 Oct 2010 15:51:29 +0200
Subject: Re: [Freeswitch-users] Use Freeswitch

















1.      
No, as far as I know there is
no such thing as TLS over UDP? Someone correct me if I?m wrong.

2.      
Yes.

3.      
Depends on your OS/hardware I
guess ? but FS itself is quite scalable.

 

/Peter

 

 





Fr?n:
freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r micho fr

Skickat: den 13 oktober 2010 09:46

Till: freeswitch-users at lists.freeswitch.org

?mne: [Freeswitch-users] Use Freeswitch





 

Dear
All,



I'm planning to use the freeswitch in my network but I have some doubt
regarding some features and need your help please before taking my decision...

I need to use the freeswitch as a registrar server + Sip Proxy...I need to know
the below:



1- Does freeswitch supports encrypted SIP inside UDP

2- Does freeswitch supports TCP

3- How much TCP connections can handle?



Regards

!DSPAM:4cb5b6b632931939610325! 







_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 		 	   		  
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From mwlucas at blackhelicopters.org  Wed Oct 13 06:48:55 2010
From: mwlucas at blackhelicopters.org (Michael W. Lucas)
Date: Wed, 13 Oct 2010 09:48:55 -0400
Subject: [Freeswitch-users] problem with freeswitch and Cisco 7960
Message-ID: <20101013134855.GA26190@bewilderbeast.blackhelicopters.org>

Hi,

I'm using Monday's FreeSWITCH build, with a config originally loaded
19 August 2010.  I'm working my way through the FS 1.0.6 book.

I have two phones, one Zoiper soft phone and one Cisco 7960.  Both
register fine, and the Cisco can call Zoiper.  Zoiper cannot call the
Cisco, however.

The Cisco phone sends its register requests from UDP port 50790.  When
FreeSWITCH sends an INVITE to the Cisco, it tries to contact
UDP/50790, and the phone returns an ICMP "destination unreachable."

This same phone is also registered to an Asterisk box.  When I call
that number, asterisk connects to UDP 5060 on this phone.  The phone
replies from a high-numbered port to 5060.

It seems I need some option to tell FreeSWITCH to send the INVITE to
5060?  Or am I missing something?

Thanks,
==ml

-- 
Michael W. Lucas 	mwlucas at BlackHelicopters.org
http://www.MichaelWLucas.com/, http://blather.MichaelWLucas.com/
New book available: Network Flow Analysis
http://www.networkflowanalysis.com/


From anthony.minessale at gmail.com  Wed Oct 13 08:13:08 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Oct 2010 10:13:08 -0500
Subject: [Freeswitch-users] skip/exit mod_fifo when no agents?
In-Reply-To: <2D766595C13E45688CE1E5C8888F3693@mbnet.local>
References: <2D766595C13E45688CE1E5C8888F3693@mbnet.local>
Message-ID: <AANLkTiky0DTwgDZt-6ndfO+ojM9ntUTijucaRTx+1Acp@mail.gmail.com>

That's not a bad idea but we don't have it implemented.
I'll think about it.


On Wed, Oct 13, 2010 at 5:28 AM, Jeroen C. van Gelderen
<slim at thegreek.com> wrote:
> Hi,
>
> Is there a proper way to drop callers out of mod_fifo when no agents are
> logged in? When no agents are logged in I'd like to proceed in the dialplan
> escalating the call to a backup number and/or backup fifo. Almost like I
> need a nowait option for callers.
>
> Cheers,
> -Slim
> --
> Jeroen C. "Slim" van Gelderen
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From Russell.Mosemann at cune.org  Wed Oct 13 08:21:29 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Wed, 13 Oct 2010 15:21:29 -0000
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today - Moises
	Silva	From Sangoma Speaking
In-Reply-To: <AANLkTin9f5ku11Nhivs5=KQe4rDa4bUr2FTM7tLQpS6k@mail.gmail.com>
Message-ID: <20101013152129.BCF223BB1D2@cuneorg-email.cune.pri>

Michael Collins <msc at freeswitch.org> said:

> Hey all!
> 
> We have a good agenda for today:
> 
> http://wiki.freeswitch.org/wiki/FS_weekly_2010_10_13

Are conferences archived? I didn't see anything obvious on the agenda
page. Unfortunately, I have something scheduled every Wednesday at that
time, and it would nice to hear the discussions.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From mario_fs at mgtech.com  Wed Oct 13 08:28:43 2010
From: mario_fs at mgtech.com (Mario)
Date: Wed, 13 Oct 2010 08:28:43 -0700
Subject: [Freeswitch-users] Can uPNP be turned off?
In-Reply-To: <6ECAF1527329364583AB525CF34ABF950832197B@ms.kallback.com>
References: <6ECAF1527329364583AB525CF34ABF950832197B@ms.kallback.com>
Message-ID: <4CB5D02B.6030802@mgtech.com>

I saw -nonat but was wondering if I could turn off just the uPNP part
since I had autonat:1.2.3.4 if that makes any sense. It seemed to fix
the problem though so I may not need to turn off uPNP. Thanks.

On 10/12/2010 05:51 PM, JRichey wrote:
> Starting FreeSwitch with "-nonat" (e.g. /usr/local/freeswitch/bin/freeswitch
> -nc -nonat) will disable UPnP.
> 
> -Justin
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of Mario
> Sent: Tuesday, October 12, 2010 5:23 PM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] Can uPNP be turned off?
> 
> 
> I am using ext... autonat:1.2.3.4 in sip internal and external as per
> wiki. The sofia status is ok for the profiles bu the nat_map shows the
> wrong address because the router is sending the wrong wan IP in the pnp
> request. Since I have autonat:1.2.3.4 coded is there a way to force FS
> not to use uPNP so it does not request nat info from the router? I need
> it on in the router for other things. Thanks. Mario
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
*Mario*


From rupa at rupa.com  Wed Oct 13 08:38:45 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 13 Oct 2010 10:38:45 -0500
Subject: [Freeswitch-users] Can uPNP be turned off?
In-Reply-To: <4CB5BF6A.308@mgtech.com>
References: <4CB4FBE2.30903@mgtech.com>
	<AANLkTikLaD2BWiiySGy0cSfAVA0q-GwodWrjF6BQ1U28@mail.gmail.com>
	<4CB5BF6A.308@mgtech.com>
Message-ID: <AANLkTikEb_yZ9TqUasquHD7ZUO8Ap9gGR0TShN0wOqea@mail.gmail.com>

Mario,

Does your router have the ability to show current NAT mappings that are
setup via UPNP?  If so, can you verify that it's (router's) table matches
the mappings that FS thinks are setup (nat_map status)?  Freeswitch is
supposed to periodically refresh the nat mappings but maybe there is some
weird issue with your particular router.

There are also some test programs in libs/miniupnpc which can query your
router's UPNP table if the router itself can't show it.

On Wed, Oct 13, 2010 at 9:17 AM, Mario <mario_fs at mgtech.com> wrote:

> Thanks, I saw that in the wiki and was going to try it next. But after
> adding autonat:1.2.3.4 looks like last nights test is ok now, I am
> hoping the problem is fixed. FS would work fine first 4-9 hours then
> stopped answering incoming. Funny thing, no problem like this with the
> SPA9000 for 2 years using 2 DSL connections. Thanks again.
>
>
-- 
-Rupa
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From kris at kriskinc.com  Wed Oct 13 08:59:50 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 13 Oct 2010 11:59:50 -0400
Subject: [Freeswitch-users] problem with freeswitch and Cisco 7960
In-Reply-To: <20101013134855.GA26190@bewilderbeast.blackhelicopters.org>
References: <20101013134855.GA26190@bewilderbeast.blackhelicopters.org>
Message-ID: <AANLkTin8o0d3Ki2zz0Syhb=cCs8wVK7FQxAmot=kyw2_@mail.gmail.com>

Michael,

  It looks like something strange is happening to the SIP traffic
between the Cisco and FreeSWITCH.  You shouldn't have to force
FreeSWITCH to send traffic to a registered user on a specific port,
the Contact: header and/or FreeSWITCH NAT handling take care of that
for you.

   Can you show us a siptrace and console output with "sofia profile
internal siptrace on" and upload it to pastebin.freeswitch.org?

On Wed, Oct 13, 2010 at 9:48 AM, Michael W. Lucas
<mwlucas at blackhelicopters.org> wrote:
> Hi,
>
> I'm using Monday's FreeSWITCH build, with a config originally loaded
> 19 August 2010. ?I'm working my way through the FS 1.0.6 book.
>
> I have two phones, one Zoiper soft phone and one Cisco 7960. ?Both
> register fine, and the Cisco can call Zoiper. ?Zoiper cannot call the
> Cisco, however.
>
> The Cisco phone sends its register requests from UDP port 50790. ?When
> FreeSWITCH sends an INVITE to the Cisco, it tries to contact
> UDP/50790, and the phone returns an ICMP "destination unreachable."
>
> This same phone is also registered to an Asterisk box. ?When I call
> that number, asterisk connects to UDP 5060 on this phone. ?The phone
> replies from a high-numbered port to 5060.
>
> It seems I need some option to tell FreeSWITCH to send the INVITE to
> 5060? ?Or am I missing something?
>
> Thanks,
> ==ml
>
> --
> Michael W. Lucas ? ? ? ?mwlucas at BlackHelicopters.org
> http://www.MichaelWLucas.com/, http://blather.MichaelWLucas.com/
> New book available: Network Flow Analysis
> http://www.networkflowanalysis.com/
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From pjintheusa at gmail.com  Wed Oct 13 09:13:11 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Wed, 13 Oct 2010 12:13:11 -0400
Subject: [Freeswitch-users] Use Freeswitch
In-Reply-To: <BAY147-w824A7255BFD3D31448B27BB550@phx.gbl>
References: <BAY147-w1466287814D00201AFE0A0BB550@phx.gbl>
	<549CFEF87AEDE841A38E9D15EAB4C04C57E5E8E65E@cooper>
	<BAY147-w824A7255BFD3D31448B27BB550@phx.gbl>
Message-ID: <AANLkTi=b1vzRXHH4UTQjsRJC_oiQOW+7ELerQgL2+NGE@mail.gmail.com>

I think it does:

http://wiki.freeswitch.org/wiki/Tls

On Wed, Oct 13, 2010 at 10:27 AM, micho fr <michofr at hotmail.com> wrote:

>  Sorry , it's not TLS over UDP...It's TLs over TCP...Does freeswitch
> support it?
>
> ------------------------------
> From: peter.olsson at visionutveckling.se
> To: freeswitch-users at lists.freeswitch.org
> Date: Wed, 13 Oct 2010 15:51:29 +0200
> Subject: Re: [Freeswitch-users] Use Freeswitch
>
>
>  1.       No, as far as I know there is no such thing as TLS over UDP?
> Someone correct me if I?m wrong.
>
> 2.       Yes.
>
> 3.       Depends on your OS/hardware I guess ? but FS itself is quite
> scalable.
>
>
>
> /Peter
>
>
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *micho fr
> *Skickat:* den 13 oktober 2010 09:46
> *Till:* freeswitch-users at lists.freeswitch.org
> *?mne:* [Freeswitch-users] Use Freeswitch
>
>
>
> Dear All,
>
> I'm planning to use the freeswitch in my network but I have some doubt
> regarding some features and need your help please before taking my
> decision...
> I need to use the freeswitch as a registrar server + Sip Proxy...I need to
> know the below:
>
> 1- Does freeswitch supports encrypted SIP inside UDP
> 2- Does freeswitch supports TCP
> 3- How much TCP connections can handle?
>
> Regards
> !DSPAM:4cb5b6b632931939610325!
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Wed Oct 13 09:24:05 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Oct 2010 12:24:05 -0400
Subject: [Freeswitch-users] BUG BUG -- mod_spidermonkey or
	mod_spidermonkey or mod_curl --- BUG BUG???
In-Reply-To: <AANLkTi=AzzT6SFtesR8RsJO9q-BKCVYJUB8KjJXrbhHK@mail.gmail.com>
References: <AANLkTinng4=jMXJLXdAMZEJSn2V_s2MqRy02C+_0wS40@mail.gmail.com>
	<AANLkTi=AzzT6SFtesR8RsJO9q-BKCVYJUB8KjJXrbhHK@mail.gmail.com>
Message-ID: <BE0BC93F-0A75-47FC-BDCB-1EC26B154F20@jerris.com>

In the future, for more information on how to file bugs:

http://wiki.freeswitch.org/wiki/Reporting_Bugs


On Oct 12, 2010, at 7:04 PM, Shamun toha md wrote:

> Never mind its solved. It was receiving 4 length instead of 2. 

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From jeff at jefflenk.com  Wed Oct 13 09:49:50 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 13 Oct 2010 09:49:50 -0700 (PDT)
Subject: [Freeswitch-users] iLBC codec generating only noise.
In-Reply-To: <AANLkTi=6Esu59uFuRMvxd-RQ6B_fiSAkXS_Qx4Zu-sfF@mail.gmail.com>
References: <AANLkTinz4JEfyGnjU-4nneOFg=A0qXHjVw9xE=jp2qSo@mail.gmail.com>
	<AANLkTinvkxRaS+Wmso1vubuH3Pj4Mib96B32Bx+sqR7H@mail.gmail.com>
	<AANLkTimmMM8Ua9AP9BwXU0yDfP9FReK2e9w-iMRiMVFu@mail.gmail.com>
	<AANLkTimiSPgCzs31VjEp0x8g1qDe-kzppsDM7uig0YGU@mail.gmail.com>
	<AANLkTikFDu0xC=VcXpzJ0_8i=NL-QfgBm5ms7nMftRzu@mail.gmail.com>
	<AANLkTi=6Esu59uFuRMvxd-RQ6B_fiSAkXS_Qx4Zu-sfF@mail.gmail.com>
Message-ID: <1286988590331-5631742.post@n2.nabble.com>


Please open a bug report with the specific build environment and details of
how to reproduce the problem on windows. - Thanks.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/iLBC-codec-generating-only-noise-tp5616210p5631742.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mario_fs at mgtech.com  Wed Oct 13 09:56:08 2010
From: mario_fs at mgtech.com (Mario)
Date: Wed, 13 Oct 2010 09:56:08 -0700
Subject: [Freeswitch-users] Can uPNP be turned off?
In-Reply-To: <AANLkTikEb_yZ9TqUasquHD7ZUO8Ap9gGR0TShN0wOqea@mail.gmail.com>
References: <4CB4FBE2.30903@mgtech.com>	<AANLkTikLaD2BWiiySGy0cSfAVA0q-GwodWrjF6BQ1U28@mail.gmail.com>	<4CB5BF6A.308@mgtech.com>
	<AANLkTikEb_yZ9TqUasquHD7ZUO8Ap9gGR0TShN0wOqea@mail.gmail.com>
Message-ID: <4CB5E4A8.9020505@mgtech.com>

The router (Linksys RV042) has a problem that caused me grief with FS: I
use dual wan, 1 static and 1 dynamic DSL. It turned out uPNP only works
for the dynamic ip. If I turn off dynamic then uPNP does not allow FS to
get the nat info for the static line. But if I have both wans active/on
and sip traffic goes out the static uPNP reports the dynamic as the
public address! This messes up the ITSP and looks like what caused FS to
stop incoming calls after several hours.

What drove me nut is: I did not think this was involved because my
Linksys SPA9000/400 SIP PBX has no trouble with the same dual wan setup.
Thanks a lot for answering but until I figure out how to dual wan FS or
replace the RV042 with something better the autonat:1.2.3.4 looks like
it fixed the problem, need more hours to test. Thanks again. Mario

On 10/13/2010 08:38 AM, Rupa Schomaker wrote:
> Mario,
> 
> Does your router have the ability to show current NAT mappings that are
> setup via UPNP?  If so, can you verify that it's (router's) table
> matches the mappings that FS thinks are setup (nat_map status)?
>  Freeswitch is supposed to periodically refresh the nat mappings but
> maybe there is some weird issue with your particular router.
> 
> There are also some test programs in libs/miniupnpc which can query your
> router's UPNP table if the router itself can't show it.
> 
> On Wed, Oct 13, 2010 at 9:17 AM, Mario <mario_fs at mgtech.com
> <mailto:mario_fs at mgtech.com>> wrote:
> 
>     Thanks, I saw that in the wiki and was going to try it next. But after
>     adding autonat:1.2.3.4 looks like last nights test is ok now, I am
>     hoping the problem is fixed. FS would work fine first 4-9 hours then
>     stopped answering incoming. Funny thing, no problem like this with the
>     SPA9000 for 2 years using 2 DSL connections. Thanks again.
> 
> 
> -- 
> -Rupa
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
*Mario*


From jeff at jefflenk.com  Wed Oct 13 09:58:27 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 13 Oct 2010 09:58:27 -0700 (PDT)
Subject: [Freeswitch-users] mod_managed socket binding problem
In-Reply-To: <AANLkTimn5wPsR18CNxXPV0kzG4Jm1ayQ3j1=QUpuUsvY@mail.gmail.com>
References: <AANLkTinqR-8GTQ7qAOM=u5TFEoDbX__qAt7p8V++D06c@mail.gmail.com>
	<D51C1FE5-C34E-4B04-95AF-1D84B6AAFCA0@freeswitch.org>
	<AANLkTimn5wPsR18CNxXPV0kzG4Jm1ayQ3j1=QUpuUsvY@mail.gmail.com>
Message-ID: <1286989107863-5631782.post@n2.nabble.com>


You should be able to implement dispose in your plugin and do any
housekeeping there.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-managed-socket-binding-problem-tp5627356p5631782.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Wed Oct 13 10:47:34 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 13 Oct 2010 10:47:34 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today - Moises
 Silva From Sangoma Speaking
In-Reply-To: <20101013152129.BCF223BB1D2@cuneorg-email.cune.pri>
References: <AANLkTin9f5ku11Nhivs5=KQe4rDa4bUr2FTM7tLQpS6k@mail.gmail.com>
	<20101013152129.BCF223BB1D2@cuneorg-email.cune.pri>
Message-ID: <AANLkTinSue_RiF9A8Vv+1SJCKsiXWSqgHyv=c7zNFJfJ@mail.gmail.com>

We haven't been archiving them in general, only when there are presentations
that are specifically for learning. I'll talk to the guys about archiving
the calls.
-MC

On Wed, Oct 13, 2010 at 8:21 AM, <Russell.Mosemann at cune.org> wrote:

> Michael Collins <msc at freeswitch.org> said:
>
> > Hey all!
> >
> > We have a good agenda for today:
> >
> > http://wiki.freeswitch.org/wiki/FS_weekly_2010_10_13
>
> Are conferences archived? I didn't see anything obvious on the agenda
> page. Unfortunately, I have something scheduled every Wednesday at that
> time, and it would nice to hear the discussions.
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Oct 13 12:05:57 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 13 Oct 2010 12:05:57 -0700
Subject: [Freeswitch-users] record on demand for a conference call
In-Reply-To: <6DFF80D26E14467299C2B7789D1D82D6@D810>
References: <6DFF80D26E14467299C2B7789D1D82D6@D810>
Message-ID: <AANLkTimuurW-3SJX1+JB=rzbgmq1mGd12hQEvzmfD4A6@mail.gmail.com>

Just curious - what value is there to recording the call when only one
person is there?
-MC

On Wed, Oct 13, 2010 at 3:42 AM, Andy <andy at fabulous4.co.uk> wrote:

>  Hi,
>
> I would like to set up my conferencing system so that it's possible to
> initiate the recording of the conference when the moderator presses a key on
> their keypad.
>
> Failing that I would like to know if there is any way of recording the
> entire conference call from the moment the first caller arrives rather than
> using the auto-record feature which only kicks in when 2 or more people are
> connected.
>
> Can anyone give me some idea how this can be done or point me to the right
> wiki pages?
>
> Many thanks for you help
> Andy
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Wed Oct 13 13:28:06 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Oct 2010 16:28:06 -0400
Subject: [Freeswitch-users] Voicemail refresh event
In-Reply-To: <4C8EBD45.4000502@communicatefreely.net>
References: <4C8EBD45.4000502@communicatefreely.net>
Message-ID: <EB35595C-4F9F-48F9-9728-F2A7DA215F18@jerris.com>

The only place this should be necessary is deleting a message and inserting a message.  There are api commands in freeswitch designed to do this and trigger all the appropriate actions.  If you are doing these from the web ui, you should use the freeswitch api commands to accomplish these tasks instead of directly changing the database.

Mike

On Sep 13, 2010, at 8:09 PM, Tim St. Pierre wrote:

> Hello,
> 
> I'm using ODBC with voicemail, and several other core functions.  My ultimate goal is to use this in
> an HA cluster, but there are also some places where this data gets looked at outside of Freeswitch.
> 
> I'm building a visual voicemail application for Aastra phones using the XML interface.  I can query
> the freeswitch database to see the voicemail_prefs and voicemail_msgs tables, listing each message,
> paths to audio, etc.  Whenever I modify this data, Freeswitch eventually notices and updates the MWI
> indicator on the phone to match what's in the database.
> 
> Is there a way to send an event to Freeswitch that will cause it to poll the database and refresh
> the status of a voice mail box?
> 
> Is it a bad idea to manipulate this data outside of freeswitch?  I don't think I'm doing anything
> that wouldn't take place if the alternate FS box changed this data in an HA situation.



From mike at jerris.com  Wed Oct 13 13:30:18 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 Oct 2010 16:30:18 -0400
Subject: [Freeswitch-users] no bridge_uuid variable in xml cdr
In-Reply-To: <AANLkTimrVcLjkof7Y9s8JzuvbL7d7SHaYOM=ZZg0M-4L@mail.gmail.com>
References: <AANLkTimrVcLjkof7Y9s8JzuvbL7d7SHaYOM=ZZg0M-4L@mail.gmail.com>
Message-ID: <6F66FEF1-BB83-410A-9B5D-7A62D1ADA1C5@jerris.com>

look at the full set of vars, there should be a last_bridge_uuid.


On Sep 14, 2010, at 2:36 AM, babak yakhchali wrote:

> Hi
> I'm using the following simple extension:
> <extension name="babak-test">		
> 		<condition field="destination_number" expression="^(223|224)$">		
> 		<action application="set" data="call_timeout=8"/-->
>         <action application="bridge" data="sofia/internal/$1%${domain}"/>
> </condition>
> 
> to test my aspx handler for mod_xml_cdr. it seems no bridge_uuid and bridge_hangup_cause exist on the xml_cdr. should I add them manually or set something to add them to channel variables?
> thanx
> <04174a82-5525-4635-8590-0729c3624209_14-09-10-10-02-51.xml>_______________________________________________



From justlikeef at gmail.com  Wed Oct 13 09:44:40 2010
From: justlikeef at gmail.com (Rob Hutton)
Date: Wed, 13 Oct 2010 12:44:40 -0400
Subject: [Freeswitch-users] Problems making a receiveing calls with
	mod_jingling
Message-ID: <201010131244.41086.justlikeef@gmail.com>

I am trying to get mod_dingaling working, and I am sure I am doing something stupid, but I can't figure out what it is...

Relevant parts of call log:

Dialplan: sofia/sipinterface_1/1001 at 192.168.1.200 Regex (PASS) [dingaling_1_pattern_1] destination_number(94045550100) =~ /^91{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$/ break=on-false                                                                                                                                    
Dialplan: sofia/sipinterface_1/1001 at 192.168.1.200 Action set(prepend=1)                                                                                    
Dialplan: sofia/sipinterface_1/1001 at 192.168.1.200 Action set(hangup_after_bridge=true)                                                                     
Dialplan: sofia/sipinterface_1/1001 at 192.168.1.200 Action bridge(dingaling/GoogleTalk/+${prepend}4045550100 at voice.google.com)                               
2010-10-13 12:15:41.427591 [DEBUG] switch_core_state_machine.c:119 (sofia/sipinterface_1/1001 at 192.168.1.200) State Change CS_ROUTING -> CS_EXECUTE         
...
EXECUTE sofia/sipinterface_1/1001 at 192.168.1.200 hash(insert/192.168.1.200-spymap/1001/87e31e76-56ff-47f2-b210-a316c3191a44)                                
EXECUTE sofia/sipinterface_1/1001 at 192.168.1.200 hash(insert/192.168.1.200-last_dial/1001/94045550100)                                                      
EXECUTE sofia/sipinterface_1/1001 at 192.168.1.200 hash(insert/192.168.1.200-last_dial/global/87e31e76-56ff-47f2-b210-a316c3191a44)                           
EXECUTE sofia/sipinterface_1/1001 at 192.168.1.200 set(RFC2822_DATE=Wed, 13 Oct 2010 12:15:41 -0400)                                                          
2010-10-13 12:15:41.431854 [DEBUG] mod_dptools.c:1024 sofia/sipinterface_1/1001 at 192.168.1.200 SET [RFC2822_DATE]=[Wed, 13 Oct 2010 12:15:41 -0400]         
EXECUTE sofia/sipinterface_1/1001 at 192.168.1.200 set(prepend=1)                                                                                             
2010-10-13 12:15:41.431854 [DEBUG] mod_dptools.c:1024 sofia/sipinterface_1/1001 at 192.168.1.200 SET [prepend]=[1]                                            
EXECUTE sofia/sipinterface_1/1001 at 192.168.1.200 set(hangup_after_bridge=true)                                                                              
2010-10-13 12:15:41.431854 [DEBUG] mod_dptools.c:1024 sofia/sipinterface_1/1001 at 192.168.1.200 SET [hangup_after_bridge]=[true]                             
EXECUTE sofia/sipinterface_1/1001 at 192.168.1.200 bridge(dingaling/GoogleTalk/+14045550100 at voice.google.com)                                                 
2010-10-13 12:15:41.432855 [DEBUG] mod_dingaling.c:1727 Unknown Profile!                                                                                   
2010-10-13 12:15:41.432855 [DEBUG] mod_dingaling.c:704 Terminate called from line 1728 state=CS_NEW                                                        
2010-10-13 12:15:41.432855 [NOTICE] mod_dingaling.c:715 Close Channel N/A [CS_NEW]                                                                         
2010-10-13 12:15:41.432855 [DEBUG] switch_core_state_machine.c:430 () Running State Change CS_DESTROY                                                      
2010-10-13 12:15:41.432855 [DEBUG] switch_core_state_machine.c:440 (N/A) State DESTROY                                                                     
2010-10-13 12:15:41.432855 [DEBUG] switch_core_state_machine.c:440 (N/A) State DESTROY going to sleep                                                      
2010-10-13 12:15:41.432855 [ERR] switch_ivr_originate.c:2605 Cannot create outgoing channel of type [dingaling] cause: [DESTINATION_OUT_OF_ORDER]          
2010-10-13 12:15:41.432855 [DEBUG] switch_ivr_originate.c:3413 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]                            
2010-10-13 12:15:41.432855 [INFO] mod_dptools.c:2575 Originate Failed.  Cause: DESTINATION_OUT_OF_ORDER                                                    
2010-10-13 12:15:41.432855 [DEBUG] switch_channel.c:2445 (sofia/sipinterface_1/1001 at 192.168.1.200) Callstate Change RINGING -> HANGUP                      
2010-10-13 12:15:41.432855 [NOTICE] mod_dptools.c:2638 Hangup sofia/sipinterface_1/1001 at 192.168.1.200 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]              
2010-10-13 12:15:41.432855 [DEBUG] switch_channel.c:2461 Send signal sofia/sipinterface_1/1001 at 192.168.1.200 [KILL]                                        

I think the interesting message is "Unknown Profile!"

Here is the dialplan that is being matched:

    <extension name="dingaling_1_pattern_1" continue="true">
      <condition field="destination_number" expression="^91{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$" bluebox="pattern_1">
        <action application="set" bluebox="prepend" data="prepend=1"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="bridge" data="dingaling/GoogleTalk/+${prepend}$1 at voice.google.com"/>
      </condition>
    </extension>

Here is the profile:

  <profile name="dingaling_1">
    <param name="name" value="GoogleTalk"/>
    <param name="login" value="justlikeef at gmail.com/gtalk"/>
    <param name="password" value="!Cl0r0x1.."/>
    <param name="server" value="talk.google.com"/>
    <param name="dialplan" value="XML"/>
    <param name="message" value=""/>
    <param name="rtp-ip" value="$${bind_server_ip}"/>
    <param name="auto-login" value="true"/>
    <param name="auto-reply" value=""/>
    <param name="sasl" value="plain"/> 
    <param name="tls" value="true"/>   
    <param name="use-rtp-timer" value="true"/>
    <param name="vad" value="both"/>
    <param name="candidate-acl" value="wan.auto"/>
    <param name="local-network-acl" value="localnet.auto"/>
    <param name="context" value="context_1"/>
    <param name="exten" value="1001"/>
  </profile>

Complete config and log in pastebin...

-- 
Thanks,
Rob


From mwlucas at blackhelicopters.org  Wed Oct 13 10:14:22 2010
From: mwlucas at blackhelicopters.org (Michael W. Lucas)
Date: Wed, 13 Oct 2010 13:14:22 -0400
Subject: [Freeswitch-users] problem with freeswitch and Cisco 7960
In-Reply-To: <AANLkTin8o0d3Ki2zz0Syhb=cCs8wVK7FQxAmot=kyw2_@mail.gmail.com>
References: <20101013134855.GA26190@bewilderbeast.blackhelicopters.org>
	<AANLkTin8o0d3Ki2zz0Syhb=cCs8wVK7FQxAmot=kyw2_@mail.gmail.com>
Message-ID: <20101013171422.GA27259@bewilderbeast.blackhelicopters.org>

On Wed, Oct 13, 2010 at 11:59:50AM -0400, Kristian Kielhofner wrote:
> Michael,
> 
>   It looks like something strange is happening to the SIP traffic
> between the Cisco and FreeSWITCH.  You shouldn't have to force
> FreeSWITCH to send traffic to a registered user on a specific port,
> the Contact: header and/or FreeSWITCH NAT handling take care of that
> for you.

I thought FreeSWITCH handled all those things automatically, that's
one reason I wanted to use it.

There's no NAT or firewall between the phone and FreeSWITCH, though.
They are on different networks, but only because the VM server farm is
across my T1.

>    Can you show us a siptrace and console output with "sofia profile
> internal siptrace on" and upload it to pastebin.freeswitch.org?

Done, as user mwlucas.

I appreciate any help you can offer.

Thanks,
==ml

> 
> On Wed, Oct 13, 2010 at 9:48 AM, Michael W. Lucas
> <mwlucas at blackhelicopters.org> wrote:
> > Hi,
> >
> > I'm using Monday's FreeSWITCH build, with a config originally loaded
> > 19 August 2010. ?I'm working my way through the FS 1.0.6 book.
> >
> > I have two phones, one Zoiper soft phone and one Cisco 7960. ?Both
> > register fine, and the Cisco can call Zoiper. ?Zoiper cannot call the
> > Cisco, however.
> >
> > The Cisco phone sends its register requests from UDP port 50790. ?When
> > FreeSWITCH sends an INVITE to the Cisco, it tries to contact
> > UDP/50790, and the phone returns an ICMP "destination unreachable."
> >
> > This same phone is also registered to an Asterisk box. ?When I call
> > that number, asterisk connects to UDP 5060 on this phone. ?The phone
> > replies from a high-numbered port to 5060.
> >
> > It seems I need some option to tell FreeSWITCH to send the INVITE to
> > 5060? ?Or am I missing something?
> >
> > Thanks,
> > ==ml
> >
> > --
> > Michael W. Lucas ? ? ? ?mwlucas at BlackHelicopters.org
> > http://www.MichaelWLucas.com/, http://blather.MichaelWLucas.com/
> > New book available: Network Flow Analysis
> > http://www.networkflowanalysis.com/
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> 
> -- 
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Michael W. Lucas 	mwlucas at BlackHelicopters.org
http://www.MichaelWLucas.com/, http://blather.MichaelWLucas.com/
New book available: Network Flow Analysis
http://www.networkflowanalysis.com/


From emd at hiredminds.com  Wed Oct 13 13:48:11 2010
From: emd at hiredminds.com (Erik M. Devane)
Date: Thu, 14 Oct 2010 05:48:11 +0900
Subject: [Freeswitch-users] Multiple Soundcard Channels
Message-ID: <4159C148-46DA-40EA-98F5-652CB19B37F9@hiredminds.com>

[Repost - apologies to all for my earlier unintentional hijack.]


Can I start multiple instances of mod-portaudio?

I would like to have multiple channels from my alsa-compliant soundcard but I can't find a way to run multiple channels in FreeSWITCH. 

I am connecting to a PABX and need to bridge calls.

Am I missing something obvious?

Thank you,

Erik
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From msc at freeswitch.org  Wed Oct 13 14:51:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 13 Oct 2010 14:51:07 -0700
Subject: [Freeswitch-users] Problems making a receiveing calls with
	mod_jingling
In-Reply-To: <201010131244.41086.justlikeef@gmail.com>
References: <201010131244.41086.justlikeef@gmail.com>
Message-ID: <AANLkTim60-09hvL1=SBrb8R86_1qqPXE2-+42ZSxZ9Ja@mail.gmail.com>

> Here is the profile:
>
>  <profile name="dingaling_1">
>
I believe that this should be:
<profile type="client">

-MC
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From william.suffill at gmail.com  Wed Oct 13 14:52:33 2010
From: william.suffill at gmail.com (William Suffill)
Date: Wed, 13 Oct 2010 17:52:33 -0400
Subject: [Freeswitch-users] problem with freeswitch and Cisco 7960
In-Reply-To: <20101013171422.GA27259@bewilderbeast.blackhelicopters.org>
References: <20101013134855.GA26190@bewilderbeast.blackhelicopters.org>
	<AANLkTin8o0d3Ki2zz0Syhb=cCs8wVK7FQxAmot=kyw2_@mail.gmail.com>
	<20101013171422.GA27259@bewilderbeast.blackhelicopters.org>
Message-ID: <AANLkTi=aJALO8g2S1shSO4VvTr2N0aMcAbP2Lxue9e87@mail.gmail.com>

http://pastebin.freeswitch.org/14221 for Michael's paste.
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From tayeb.meftah at gmail.com  Thu Oct 14 14:20:26 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Thu, 14 Oct 2010 23:20:26 +0200
Subject: [Freeswitch-users] Fwd: Fwd: [Rates] Please tell Meftah Tayeb to
 stop harass and damage my company
Message-ID: <4CB7741A.4060202@gmail.com>

ban diegoviola
piss of shit of the world



-------- Message original --------
Sujet: 	Fwd: [Rates] Please tell Meftah Tayeb to stop harass and damage 
my company
Date : 	Wed, 13 Oct 2010 18:13:09 -0400
De : 	Yaro Donchenko <yaro at ovetel.com>
Pour : 	Tayeb Meftah <tayeb.meftah at gmail.com>





---------- Forwarded message ----------
From: *Diego Viola* <diego.viola at gmail.com <mailto:diego.viola at gmail.com>>
Date: Wed, Oct 13, 2010 at 5:46 PM
Subject: [Rates] Please tell Meftah Tayeb to stop harass and damage my 
company
To: rates <rates at ovetel.com <mailto:rates at ovetel.com>>, 
we-sales at ovetel.com <mailto:we-sales at ovetel.com>, ee-sales at ovetel.com 
<mailto:ee-sales at ovetel.com>, cis-sales at ovetel.com 
<mailto:cis-sales at ovetel.com>, mena-sales at ovetel.com 
<mailto:mena-sales at ovetel.com>, africa-sales at ovetel.com 
<mailto:africa-sales at ovetel.com>, Tony Volkov <tony at ovetel.com 
<mailto:tony at ovetel.com>>


Hello OVETEL,

This person that works for your company, Tayeb Meftah, it's an
homosexual freak that harass me every day in the Messenger, and I
believe he has serious mental issues.

I believe he is trashing the image of your company. Please ask him to
stop bothering me and tell stupid things about me and my company.

This freak is trashing your image and of your company, so if you are
interested to have him in your company, it's your problem.

But I just want him to stop bothering me and stop saying bullshit
about me and my company.

For you to see that this is all true, I put at your disposal the
conversations that I had with the freak of your company. When he told
me that he was the owner of OVETEL and when he told me that he told
you bad things about me and my company?s image.

This homosexual freak of your company is obsessed to write me and ask
me things for free. Please take matters into this subject, and if you
wish to have this person in your company it?s your problem.

If this freak continues to harass me and damage the name of my company
I will have to take judicial actions against him so please take some
control about this.

Best Regards,
Diego Viola
Representative of Bridgecom LLC




---------- Forwarded message ----------
From: tayeb at ovetel.com <mailto:tayeb at ovetel.com> <tayeb at ovetel.com 
<mailto:tayeb at ovetel.com>>
Date: Wed, Oct 13, 2010 at 6:24 PM
Subject: Chat with tayeb at ovetel.com <mailto:tayeb at ovetel.com>
To: diego.viola at gmail.com <mailto:diego.viola at gmail.com>

5:47 PM tayeb.meftah: ovetel don't speak with you anymore just don't 
wait for it
6:18 PM me: what did you tell them?
6:19 PM tayeb.meftah: shut up and go away from me
  me: ok
  tayeb.meftah: ecause you are montaly infected
  and you have a very high late
6:20 PM and you lie in adition to this
  me: ?
  you are an idiot for doing this
  tayeb.meftah: and you?
  go fuck yourself



11:48 AM tayeb.meftah: http://www.ovetel.com
  my company
11:49 AM me: nice
  are you the owner of this company?
  tayeb.meftah: me and other one
Notice: This mail is covered by the Electronics Communications Privacy 
Act, 18 U.S.C. 2510-2521 and is legally privileged. This message, 
together with any attachments, is intended only for the addressee. It 
may contain information which is legally privileged, confidential and 
exempt from disclosure. If you are not the intended recipient, you are 
hereby notified that any disclosure, copying, distribution, use or any 
action or reliance on this communication is strictly prohibited. If you 
have received this e-mail in error, please notify the sender immediately 
by telephone +1-212-401-0707 or by return e-mail to the original sender 
and delete and permanently erase this message and all attachments of any 
and all originals and copies.




-- 
Best Regards
*Yaro Donchenko*
/VP of Business Development/

Direct: +1-845-475-9347
Mobile: +1-917-267-9276
MSN: yaro at ovetel.com <mailto:yaro at ovetel.com>
Yahoo: yaroovetel

OVETEL USA: +1-212-401-0707
OVETEL UK: +44-20-34110486
OVETEL Ukraine: +380-44-3607309
OVETEL Jordan: +962-6-2508905
OVETEL South Africa: +27-11-4613345

Organize a meeting with me: http://tungle.me/yaro

Follow us on Twitter for the latest price updates: http://twitter.com/ovetel

Notice: This mail is covered by the Electronics Communications Privacy Act, 18 U.S.C. 2510-2521 and is legally privileged. This message, together with any attachments, is intended only for the addressee. It may contain information which is legally privileged, confidential and exempt from disclosure. If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution, use or any action or reliance on this communication is strictly prohibited. If you have received this e-mail in error, please notify the sender immediately by telephone +1-212-401-0707 or by return e-mail to the original sender and delete and permanently erase this message and all attachments of any and all originals and copies.



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From anthony.minessale at gmail.com  Wed Oct 13 15:36:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Oct 2010 17:36:45 -0500
Subject: [Freeswitch-users] Fwd: Fwd: [Rates] Please tell Meftah Tayeb
 to stop harass and damage my company
In-Reply-To: <4CB7741A.4060202@gmail.com>
References: <4CB7741A.4060202@gmail.com>
Message-ID: <AANLkTi=GRHtVtkJYpHStM-YPBPUCmi1gvCg+DspStfNO@mail.gmail.com>

you are now both moderated from this mailing list......


On Thu, Oct 14, 2010 at 4:20 PM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:

>  ban diegoviola
> piss of shit of the world
>
>
>
> -------- Message original --------  Sujet: Fwd: [Rates] Please tell Meftah
> Tayeb to stop harass and damage my company  Date : Wed, 13 Oct 2010
> 18:13:09 -0400  De : Yaro Donchenko <yaro at ovetel.com> <yaro at ovetel.com>   :
> Tayeb Meftah <tayeb.meftah at gmail.com> <tayeb.meftah at gmail.com>
>
>
>
> ---------- Forwarded message ----------
> From: Diego Viola <diego.viola at gmail.com>
> Date: Wed, Oct 13, 2010 at 5:46 PM
> Subject: [Rates] Please tell Meftah Tayeb to stop harass and damage my
> company
> To: rates <rates at ovetel.com>, we-sales at ovetel.com, ee-sales at ovetel.com,
> cis-sales at ovetel.com, mena-sales at ovetel.com, africa-sales at ovetel.com, Tony
> Volkov <tony at ovetel.com>
>
>
> Hello OVETEL,
>
> This person that works for your company, Tayeb Meftah, it's an
> homosexual freak that harass me every day in the Messenger, and I
> believe he has serious mental issues.
>
> I believe he is trashing the image of your company. Please ask him to
> stop bothering me and tell stupid things about me and my company.
>
> This freak is trashing your image and of your company, so if you are
> interested to have him in your company, it's your problem.
>
> But I just want him to stop bothering me and stop saying bullshit
> about me and my company.
>
> For you to see that this is all true, I put at your disposal the
> conversations that I had with the freak of your company. When he told
> me that he was the owner of OVETEL and when he told me that he told
> you bad things about me and my company?s image.
>
> This homosexual freak of your company is obsessed to write me and ask
> me things for free. Please take matters into this subject, and if you
> wish to have this person in your company it?s your problem.
>
> If this freak continues to harass me and damage the name of my company
> I will have to take judicial actions against him so please take some
> control about this.
>
> Best Regards,
> Diego Viola
> Representative of Bridgecom LLC
>
>
>
>
> ---------- Forwarded message ----------
> From: tayeb at ovetel.com <tayeb at ovetel.com>
> Date: Wed, Oct 13, 2010 at 6:24 PM
> Subject: Chat with tayeb at ovetel.com
> To: diego.viola at gmail.com
>
> 5:47 PM tayeb.meftah: ovetel don't speak with you anymore just don't wait
> for it
> 6:18 PM me: what did you tell them?
> 6:19 PM tayeb.meftah: shut up and go away from me
>  me: ok
>  tayeb.meftah: ecause you are montaly infected
>  and you have a very high late
> 6:20 PM and you lie in adition to this
>  me: ?
>  you are an idiot for doing this
>  tayeb.meftah: and you?
>  go fuck yourself
>
>
>
> 11:48 AM tayeb.meftah: http://www.ovetel.com
>  my company
> 11:49 AM me: nice
>  are you the owner of this company?
>  tayeb.meftah: me and other one
> Notice: This mail is covered by the Electronics Communications Privacy Act,
> 18 U.S.C. 2510-2521 and is legally privileged. This message, together with
> any attachments, is intended only for the addressee. It may contain
> information which is legally privileged, confidential and exempt from
> disclosure. If you are not the intended recipient, you are hereby notified
> that any disclosure, copying, distribution, use or any action or reliance on
> this communication is strictly prohibited. If you have received this e-mail
> in error, please notify the sender immediately by telephone +1-212-401-0707
> or by return e-mail to the original sender and delete and permanently erase
> this message and all attachments of any and all originals and copies.
>
>
>
>
> --
> Best Regards
> *Yaro Donchenko*
> *VP of Business Development*
>
> Direct: +1-845-475-9347
> Mobile: +1-917-267-9276
> MSN: yaro at ovetel.com
> Yahoo: yaroovetel
>
> OVETEL USA: +1-212-401-0707
> OVETEL UK: +44-20-34110486
> OVETEL Ukraine: +380-44-3607309
> OVETEL Jordan: +962-6-2508905
> OVETEL South Africa: +27-11-4613345
>
> Organize a meeting with me: http://tungle.me/yaro
>
> Follow us on Twitter for the latest price updates:
> http://twitter.com/ovetel
>
> Notice: This mail is covered by the Electronics Communications Privacy Act, 18 U.S.C. 2510-2521 and is legally privileged. This message, together with any attachments, is intended only for the addressee. It may contain information which is legally privileged, confidential and exempt from disclosure. If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution, use or any action or reliance on this communication is strictly prohibited. If you have received this e-mail in error, please notify the sender immediately by telephone +1-212-401-0707 or by return e-mail to the original sender and delete and permanently erase this message and all attachments of any and all originals and copies.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From riedinger at sns.eu  Wed Oct 13 15:38:32 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Thu, 14 Oct 2010 00:38:32 +0200
Subject: [Freeswitch-users] H323 to Sip conversion problem
In-Reply-To: <1286981982.2305.4.camel@Zaphod>
References: <1286981982.2305.4.camel@Zaphod>
Message-ID: <4CB634E8.9050404@sns.eu>

  Hi John,

a ptime of 2 seems to be much too less and is a strange value, the 
default value is 20. Maybe you should try to force its usage. I'm not 
sure, how this can be done, maybe by using
<param name="ptime-override-value" value="20"/> or by specifying 
G729 at 20i for the codec.

BR
    Jan

Am 13.10.2010 16:59, schrieb John Carpenter:
> Hi, I am trying to bridge an incoming h323 call to an external SIP 
> provider, I am using latest git release. If I use G729 codec I get message
>
> "Unsupported ptime of 2 on  write Audio codec G.729{sw}  for 
> connection [0xb4c0ee10]"
>
> and call fails to connect. If I use ulaw codec call fails to connect with
>
> "Write PDU fail: no control channel"
>
> If I make a straight SIP to SIP call though same provider all work ok. 
> I have posted log of call in http://pastebin.freeswitch.org/14216 
> because it is rather large.
> This is my first venture into using the mod_h323 module and maybe I am 
> doing something stupid but have read all docs and seem to have hit a 
> brick wall on this.
>
> my h323.conf.xml file looks like this
>
> <configuration name="h323.conf" description="H323 Endpoints">
> <settings>
> <param name="trace-level" value="10"/>
> <param name="context" value="public"/>
> <param name="dialplan" value="XML"/>
> <param name="codec-prefs" value="PCMA,PCMU,GSM,G729"/>
> <param name="use-rtp-timer" value="true"/> <!-- enable RTP timer - 
> should always be enabled -->
> <param name="rtp-timer-name" value="soft"/> <!-- Timer name, soft is 
> default -->
> <!-- <param name="ptime-override-value" value="20"/> --> <!-- Override 
> negotiated ptime value with this value -->
> <param name="gk-address" value=""/> <!-- empty to disable, "*" to 
> search LAN -->
> <param name="gk-identifer" value=""/> <!-- optional name of gk -->
> <param name="gk-interface" value=""/> <!-- mandatory listener 
> interface name -->
> <param name="gk-retry" value="30"/> <!-- optional GK register retry 
> timer -->
> <param name="faststart" value="true"/> <!-- optional -->
> <param name="h245tunneling" value="true"/> <!-- optional -->
> <param name="h245insetup" value="true"/> <!-- optional -->
> <param name="jitter-size" value="60"/> <!-- optional -->
> <param name="progress-indication" value="8"/> <!-- optional - PI value 
> in progress message-->
> <param name="alerting-indication" value="8"/> <!-- optional - PI value 
> in alerting message-->
> <param name="endpoint-name" value="fs"/>
> <param name="fax-old-asn" value="true"/>
> </settings>
> <listeners>
> <listener name="default">
> <param name="h323-ip" value="$${local_ip_v4}"/>
> <param name="h323-port" value="1720"/>
> </listener>
> </listeners>
> </configuration>
>
> And this is the log of the xml_curl dialplan that is executed
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="dialplan" description="php dialplan">
> <context name="public">
> <extension name="normal">
> <condition field="destination_number" expression="^(\d+)$">
> <action application="set" data="bypass_media=false"/>
> <action application="set" data="proxy_media=true"/>
> <action application="bridge" data="sofia/gateway/arbinet-o/$1"/>
> </condition>
> </extension>
> </context>
> </section>
> </document>
>
> Any help will be greatly appreciated
>
> regards, John Carpenter
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                         E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973

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From kris at kriskinc.com  Wed Oct 13 16:57:24 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 13 Oct 2010 19:57:24 -0400
Subject: [Freeswitch-users] problem with freeswitch and Cisco 7960
In-Reply-To: <20101013171422.GA27259@bewilderbeast.blackhelicopters.org>
References: <20101013134855.GA26190@bewilderbeast.blackhelicopters.org>
	<AANLkTin8o0d3Ki2zz0Syhb=cCs8wVK7FQxAmot=kyw2_@mail.gmail.com>
	<20101013171422.GA27259@bewilderbeast.blackhelicopters.org>
Message-ID: <AANLkTi=qXdrG2rbPtv2xsSLMzBVE4diw9krCqoiWSTB2@mail.gmail.com>

Michael,

  Do you have a sip trace that includes the REGISTER sequence from the Cisco?

On Wed, Oct 13, 2010 at 1:14 PM, Michael W. Lucas
<mwlucas at blackhelicopters.org> wrote:
> On Wed, Oct 13, 2010 at 11:59:50AM -0400, Kristian Kielhofner wrote:
>> Michael,
>>
>> ? It looks like something strange is happening to the SIP traffic
>> between the Cisco and FreeSWITCH. ?You shouldn't have to force
>> FreeSWITCH to send traffic to a registered user on a specific port,
>> the Contact: header and/or FreeSWITCH NAT handling take care of that
>> for you.
>
> I thought FreeSWITCH handled all those things automatically, that's
> one reason I wanted to use it.
>
> There's no NAT or firewall between the phone and FreeSWITCH, though.
> They are on different networks, but only because the VM server farm is
> across my T1.
>
>> ? ?Can you show us a siptrace and console output with "sofia profile
>> internal siptrace on" and upload it to pastebin.freeswitch.org?
>
> Done, as user mwlucas.
>
> I appreciate any help you can offer.
>
> Thanks,
> ==ml
>
>>
>> On Wed, Oct 13, 2010 at 9:48 AM, Michael W. Lucas
>> <mwlucas at blackhelicopters.org> wrote:
>> > Hi,
>> >
>> > I'm using Monday's FreeSWITCH build, with a config originally loaded
>> > 19 August 2010. ?I'm working my way through the FS 1.0.6 book.
>> >
>> > I have two phones, one Zoiper soft phone and one Cisco 7960. ?Both
>> > register fine, and the Cisco can call Zoiper. ?Zoiper cannot call the
>> > Cisco, however.
>> >
>> > The Cisco phone sends its register requests from UDP port 50790. ?When
>> > FreeSWITCH sends an INVITE to the Cisco, it tries to contact
>> > UDP/50790, and the phone returns an ICMP "destination unreachable."
>> >
>> > This same phone is also registered to an Asterisk box. ?When I call
>> > that number, asterisk connects to UDP 5060 on this phone. ?The phone
>> > replies from a high-numbered port to 5060.
>> >
>> > It seems I need some option to tell FreeSWITCH to send the INVITE to
>> > 5060? ?Or am I missing something?
>> >
>> > Thanks,
>> > ==ml
>> >
>> > --
>> > Michael W. Lucas ? ? ? ?mwlucas at BlackHelicopters.org
>> > http://www.MichaelWLucas.com/, http://blather.MichaelWLucas.com/
>> > New book available: Network Flow Analysis
>> > http://www.networkflowanalysis.com/
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Kristian Kielhofner
>> http://www.astlinux.org
>> http://blog.krisk.org
>> http://www.star2star.com
>> http://www.submityoursip.com
>> http://www.voalte.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> --
> Michael W. Lucas ? ? ? ?mwlucas at BlackHelicopters.org
> http://www.MichaelWLucas.com/, http://blather.MichaelWLucas.com/
> New book available: Network Flow Analysis
> http://www.networkflowanalysis.com/
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From msc at freeswitch.org  Wed Oct 13 17:13:08 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 13 Oct 2010 17:13:08 -0700
Subject: [Freeswitch-users] fifo help
In-Reply-To: <AANLkTinv11QYE0UKnOcJE=v8ZADgX3q6AosyXAcCF1jS@mail.gmail.com>
References: <AANLkTinv11QYE0UKnOcJE=v8ZADgX3q6AosyXAcCF1jS@mail.gmail.com>
Message-ID: <AANLkTim9bafNjioV0iVGSMbW1t7q7mZvstKQziWS74Nh@mail.gmail.com>

Kyle,

I updated the agent login/logout example on the wiki:
http://wiki.freeswitch.org/wiki/Mod_fifo#Simple_On-hook_Agent_Login.2FLogout_Example

If you add group_confirm_key and group_confirm_file to the member dialstring
it will work. (I just tested on latest git no problems.)

Also, you are trying to add a fifo member every time a caller is put into
the queue. Adding and removing fifo member (aka "agents") is totally
separate from putting callers into the fifo queue. You'll see in the wiki
that there are 3 different extensions: one for an agent logging in, one for
an agent logging out, and one for putting a caller into the fifo.

Give it a whirl and let us know how it works for you.

-MC

P.S. -  use the text on the wiki and not the file in SVN contrib - I still
need to update the contrib repo...

On Tue, Oct 12, 2010 at 11:19 AM, Kyle Haefner
<Kyle.Haefner at colostate.edu>wrote:

> Hi All,
>
> I'm pretty new to freeswitch :)
>
> I'm trying to set up the following scenario:
>
> 1. A caller calls in is greeted by a message that says, "Please hold while
> your party is located"
> 2. The caller is placed on hold
> 3. An outbound call sequentially rings cell phones through a gateway
> 4. If a cellphone answers the "agent" must push a digit (to verify they are
> not voice mail)
> 5. If a digit is pushed the agent is connected to the caller
>
> I'm trying to implement this using mod_fifo, however when I add the caller
> to the fifo, processing of that extension stops and I can't ever call and
> test the cellphone to add them as a member of the fifo.
>
> <context name="custom">
>   <extension name="FIFO-Call">
>     <condition field="destination_number" expression="^(100)$">
> -       <action application="answer"/>
>         <action application="playback"
> data="/usr/local/freeswitch/sounds/en/us/callie/misc/noc_intro.wav"/>
>         <action application="fifo" data="noc_fifo in nowait"/>
>         <action application="sleep" data="300"/>
>         <action application="set"
> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>         <action application="set"
> data="effective_caller_id_name=${outbound_caller_id_name}"/>
>         <action application="set" data="ringback=${us-ring}"/>
>         <action application="set" data="call_timeout=60"/>
>         <action application="set"
> data="group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/misc/prove_youre_human.wav"/>
>         <action application="set" data="group_confirm_key=4"/>
>         <action application="set" data="fifo_member(add noc_fifo
> {fifo_member_wait=nowait}sofia/custom_dialplan/89883757 at pstn-gateway
> :5060)"/>
>     </condition>
>   </extension>
> </context>
>
>
> Any help would be greatly appreciated!
>
> Kyle
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Oct 13 17:36:26 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Oct 2010 19:36:26 -0500
Subject: [Freeswitch-users] 3-way calling fails with git
In-Reply-To: <AANLkTi==jVtq3AosUD=pYTjCq3O-k7dtvVPbxB3fnaa4@mail.gmail.com>
References: <AANLkTi==jVtq3AosUD=pYTjCq3O-k7dtvVPbxB3fnaa4@mail.gmail.com>
Message-ID: <47F33E8E-5269-419C-A5C7-1B5598B60392@freeswitch.org>

Can you give me your config to see exactly what is going on.

/b

On Oct 12, 2010, at 2:25 AM, Woody Dickson wrote:

> Hi,
> 
> I am having problem with using the att_xfer app after upgrading to the latest git.  After the callee presses a meta_app key and bridged a 3rd party in, the callee got hung up immediately leaving the 3rd part and the original caller talking to each other only.
> 
> I was able to get 3-way call to work in the previous release of fs.  Does anyone know if this is a bug or a bad config on my end?
> 
> Thanks,
> Woody
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Wed Oct 13 18:31:16 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Oct 2010 20:31:16 -0500
Subject: [Freeswitch-users] 3-way calling fails with git
In-Reply-To: <47F33E8E-5269-419C-A5C7-1B5598B60392@freeswitch.org>
References: <AANLkTi==jVtq3AosUD=pYTjCq3O-k7dtvVPbxB3fnaa4@mail.gmail.com>
	<47F33E8E-5269-419C-A5C7-1B5598B60392@freeswitch.org>
Message-ID: <AANLkTikH8p2LGGMToXEdhoRLYVXyH-pWVxXHSfcV2C0x@mail.gmail.com>

got it, try latest

On Wed, Oct 13, 2010 at 7:36 PM, Brian West <brian at freeswitch.org> wrote:
> Can you give me your config to see exactly what is going on.
>
> /b
>
> On Oct 12, 2010, at 2:25 AM, Woody Dickson wrote:
>
>> Hi,
>>
>> I am having problem with using the att_xfer app after upgrading to the latest git. ?After the callee presses a meta_app key and bridged a 3rd party in, the callee got hung up immediately leaving the 3rd part and the original caller talking to each other only.
>>
>> I was able to get 3-way call to work in the previous release of fs. ?Does anyone know if this is a bug or a bad config on my end?
>>
>> Thanks,
>> Woody
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From justlikeef at gmail.com  Wed Oct 13 15:07:45 2010
From: justlikeef at gmail.com (Rob Hutton)
Date: Wed, 13 Oct 2010 18:07:45 -0400
Subject: [Freeswitch-users] Problems making a receiveing calls with
	mod_jingling
In-Reply-To: <AANLkTim60-09hvL1=SBrb8R86_1qqPXE2-+42ZSxZ9Ja@mail.gmail.com>
References: <201010131244.41086.justlikeef@gmail.com>
	<AANLkTim60-09hvL1=SBrb8R86_1qqPXE2-+42ZSxZ9Ja@mail.gmail.com>
Message-ID: <201010131807.46035.justlikeef@gmail.com>

Caught and changed that, but it does the same thing.  I also changed the profile name to gtalk in case there was a case sensitivity issue or something, but same result.

I am told that the wiki is not quite correct either, but with everyone away from their development PCs, it will be the weekend before the corrections are available.

-- 
Thanks,
Rob
On Wednesday 13 October 2010 17:51:07 Michael Collins wrote:
> > Here is the profile:
> >
> >  <profile name="dingaling_1">
> >
> I believe that this should be:
> <profile type="client">
> 
> -MC
> 
I


From tear152 at hotmail.com  Wed Oct 13 18:35:55 2010
From: tear152 at hotmail.com (=?big5?B?pr8gQ2hpYW5nIKbcpKQgQ2hpaC1DaHVuZyAgIFd5Ymll?=)
Date: Thu, 14 Oct 2010 09:35:55 +0800
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>
Message-ID: <SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>



 




Hello
 
I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X-Lite to call
 
my question is how can I call between these servers? 
 
if I can , how shoould I set in *.xml? or X-Lite?
 
thank you for your attension
 
 
Best Regards
Gary
 		 	   		  
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From xyangni at gmail.com  Wed Oct 13 19:55:49 2010
From: xyangni at gmail.com (xuyan yang)
Date: Thu, 14 Oct 2010 10:55:49 +0800
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>
	<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>
Message-ID: <AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>

I think you can reg at server A and than call B as a normal SIP call to
external domain.

2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>

>
>
>
> Hello
>
> I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X-Lite to
> call
>
> my question is how can I call between these servers?
>
> if I can , how shoould I set in *.xml? or X-Lite?
>
> thank you for your attension
>
>
> Best Regards
> Gary
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tear152 at hotmail.com  Wed Oct 13 20:44:59 2010
From: tear152 at hotmail.com (=?big5?B?pr8gQ2hpYW5nIKbcpKQgQ2hpaC1DaHVuZyAgIFd5Ymll?=)
Date: Thu, 14 Oct 2010 11:44:59 +0800
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>,
	<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>,
	<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>
Message-ID: <SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>


I tried it 
but not working
and show errors below
2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate Failed.  Cause: USER_NOT_REGISTERED
 
and for example
 I use SIP server 60.248.175.38  user id=1007
and I call to 1007 at 60.248.175.37    that will show these errors
 
how should set more?
 
thank you for your attension
 
Best Regards
Gary
 


Date: Thu, 14 Oct 2010 10:55:49 +0800
From: xyangni at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite

I think you can reg at server A and than call B as a normal SIP call to external domain.


2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>



 

Hello
 
I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X-Lite to call
 
my question is how can I call between these servers? 
 
if I can , how shoould I set in *.xml? or X-Lite?
 
thank you for your attension
 
 
Best Regards
Gary

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



_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 		 	   		  
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From curriegrad2004 at gmail.com  Wed Oct 13 22:17:13 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Wed, 13 Oct 2010 22:17:13 -0700
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>
	<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>
	<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>
	<SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>
Message-ID: <AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>

You can create a SIP trunk between the two servers so calls can be
routed back and forth from the servers you've specified.

2010/10/13 ? Chiang ?? Chih-Chung   Wybie <tear152 at hotmail.com>:
> I tried it
> but not working
> and show errors below
> 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493 Cannot create
> outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493 Cannot create
> outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate Failed.
> Cause: USER_NOT_REGISTERED
>
> and for example
> ?I use SIP server 60.248.175.38? user id=1007
> and I call to 1007 at 60.248.175.37??? that will show these errors
>
> how should set more?
>
> thank you for your attension
>
> Best Regards
> Gary
>
> ________________________________
> Date: Thu, 14 Oct 2010 10:55:49 +0800
> From: xyangni at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> I think you can reg at server A and than call B as a normal SIP call to
> external domain.
>
> 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
>
>
>
> Hello
>
> I set two SIP servers ?60.248.175.37 & 60.248.175.38 and use X-Lite to call
>
> my question is how can I call between these servers?
>
> if I can , how shoould I set in *.xml? or X-Lite?
>
> thank you for your attension
>
>
> Best Regards
> Gary
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From curriegrad2004 at gmail.com  Wed Oct 13 22:19:35 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Wed, 13 Oct 2010 22:19:35 -0700
Subject: [Freeswitch-users] Fwd: Fwd: [Rates] Please tell Meftah Tayeb
 to stop harass and damage my company
In-Reply-To: <AANLkTi=GRHtVtkJYpHStM-YPBPUCmi1gvCg+DspStfNO@mail.gmail.com>
References: <4CB7741A.4060202@gmail.com>
	<AANLkTi=GRHtVtkJYpHStM-YPBPUCmi1gvCg+DspStfNO@mail.gmail.com>
Message-ID: <AANLkTimrHiTU=v73=1tC8dKuhey2eA-HZtRg=BBVsgOW@mail.gmail.com>

This is not the place where you throw your personal vendetta against
each other. We're here to share ideas and help users who are having
issues with FreeSwitch.

On Wed, Oct 13, 2010 at 3:36 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
>
> you are now both moderated from this mailing list......
>
> On Thu, Oct 14, 2010 at 4:20 PM, Meftah Tayeb <tayeb.meftah at gmail.com> wrote:
>>
>> ban diegoviola
>> piss of shit of the world
>>
>>
>>
>> -------- Message original --------
>> Sujet: Fwd: [Rates] Please tell Meftah Tayeb to stop harass and damage my company
>> Date?: Wed, 13 Oct 2010 18:13:09 -0400
>> De?: Yaro Donchenko <yaro at ovetel.com>
>> ?: Tayeb Meftah <tayeb.meftah at gmail.com>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Diego Viola <diego.viola at gmail.com>
>> Date: Wed, Oct 13, 2010 at 5:46 PM
>> Subject: [Rates] Please tell Meftah Tayeb to stop harass and damage my company
>> To: rates <rates at ovetel.com>, we-sales at ovetel.com, ee-sales at ovetel.com, cis-sales at ovetel.com, mena-sales at ovetel.com, africa-sales at ovetel.com, Tony Volkov <tony at ovetel.com>
>>
>>
>> Hello OVETEL,
>>
>> This person that works for your company, Tayeb Meftah, it's an
>> homosexual freak that harass me every day in the Messenger, and I
>> believe he has serious mental issues.
>>
>> I believe he is trashing the image of your company. Please ask him to
>> stop bothering me and tell stupid things about me and my company.
>>
>> This freak is trashing your image and of your company, so if you are
>> interested to have him in your company, it's your problem.
>>
>> But I just want him to stop bothering me and stop saying bullshit
>> about me and my company.
>>
>> For you to see that this is all true, I put at your disposal the
>> conversations that I had with the freak of your company. When he told
>> me that he was the owner of OVETEL and when he told me that he told
>> you bad things about me and my company?s image.
>>
>> This homosexual freak of your company is obsessed to write me and ask
>> me things for free. Please take matters into this subject, and if you
>> wish to have this person in your company it?s your problem.
>>
>> If this freak continues to harass me and damage the name of my company
>> I will have to take judicial actions against him so please take some
>> control about this.
>>
>> Best Regards,
>> Diego Viola
>> Representative of Bridgecom LLC
>>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: tayeb at ovetel.com <tayeb at ovetel.com>
>> Date: Wed, Oct 13, 2010 at 6:24 PM
>> Subject: Chat with tayeb at ovetel.com
>> To: diego.viola at gmail.com
>>
>> 5:47 PM tayeb.meftah: ovetel don't speak with you anymore just don't wait for it
>> 6:18 PM me: what did you tell them?
>> 6:19 PM tayeb.meftah: shut up and go away from me
>> ?me: ok
>> ?tayeb.meftah: ecause you are montaly infected
>> ?and you have a very high late
>> 6:20 PM and you lie in adition to this
>> ?me: ?
>> ?you are an idiot for doing this
>> ?tayeb.meftah: and you?
>> ?go fuck yourself
>>
>>
>>
>> 11:48 AM tayeb.meftah: http://www.ovetel.com
>> ?my company
>> 11:49 AM me: nice
>> ?are you the owner of this company?
>> ?tayeb.meftah: me and other one
>> Notice: This mail is covered by the Electronics Communications Privacy Act, 18 U.S.C. 2510-2521 and is legally privileged. This message, together with any attachments, is intended only for the addressee. It may contain information which is legally privileged, confidential and exempt from disclosure. If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution, use or any action or reliance on this communication is strictly prohibited. If you have received this e-mail in error, please notify the sender immediately by telephone +1-212-401-0707 or by return e-mail to the original sender and delete and permanently erase this message and all attachments of any and all originals and copies.
>>
>>
>>
>>
>> --
>> Best Regards
>> Yaro Donchenko
>> VP of Business Development
>>
>> Direct: +1-845-475-9347
>> Mobile: +1-917-267-9276
>> MSN: yaro at ovetel.com
>> Yahoo: yaroovetel
>>
>> OVETEL USA: +1-212-401-0707
>> OVETEL UK: +44-20-34110486
>> OVETEL Ukraine: +380-44-3607309
>> OVETEL Jordan: +962-6-2508905
>> OVETEL South Africa: +27-11-4613345
>>
>> Organize a meeting with me: http://tungle.me/yaro
>>
>> Follow us on Twitter for the latest price updates: http://twitter.com/ovetel
>>
>> Notice: This mail is covered by the Electronics Communications Privacy Act, 18 U.S.C. 2510-2521 and is legally privileged. This message, together with any attachments, is intended only for the addressee. It may contain information which is legally privileged, confidential and exempt from disclosure. If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution, use or any action or reliance on this communication is strictly prohibited. If you have received this e-mail in error, please notify the sender immediately by telephone +1-212-401-0707 or by return e-mail to the original sender and delete and permanently erase this message and all attachments of any and all originals and copies.
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From curriegrad2004 at gmail.com  Wed Oct 13 22:28:42 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Wed, 13 Oct 2010 22:28:42 -0700
Subject: [Freeswitch-users] Problems making a receiveing calls with
	mod_jingling
In-Reply-To: <201010131807.46035.justlikeef@gmail.com>
References: <201010131244.41086.justlikeef@gmail.com>
	<AANLkTim60-09hvL1=SBrb8R86_1qqPXE2-+42ZSxZ9Ja@mail.gmail.com>
	<201010131807.46035.justlikeef@gmail.com>
Message-ID: <AANLkTi=K65YxFQAXudWiPTSAUSH2Xzjd4ks5_coSJ5K_@mail.gmail.com>

This is my config inside the jingle_profiles:
 <profile type="client">
*    <param name="name" value="gtalk[Example]"/>*
   <param name="login" value="[EXAMPLE]@gmail.com/talk"/>
   <param name="password" value="[REDACTED]"/>
   <param name="dialplan" value="XML"/>
   <param name="context" value="public"/>
   <param name="message" value="Online"/>
   <param name="rtp-ip" value="$${bind_server_ip}"/>
   <!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
   <param name="auto-login" value="true"/>
   <!-- SASL "plain" or "md5" -->
   <param name="sasl" value="plain"/>
   <!-- if the server where the jabber is hosted is not the same as the one
in the jid -->
   <param name="server" value="talk.google.com"/>
   <!-- Enable TLS or not -->
   <param name="tls" value="true"/>
   <!-- disable to trade async for more calls -->
   <param name="use-rtp-timer" value="true"/>
   <!-- default extension (if one cannot be determined) -->
   <param name="exten" value="[EXTENSION]"/>
   <!-- VAD choose one -->
   <!-- <param name="vad" value="in"/> -->
   <!-- <param name="vad" value="out"/> -->
   <param name="vad" value="both"/>
   <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
   <param name="candidate-acl" value="wan.auto"/>
   <param name="local-network-acl" value="localnet.auto"/>
 </profile>

And dialplan to follow:

 <extension name="gvoice_out">
   <condition field="destination_number" expression="^4664(\d{10})$">
*      <action application="bridge" data="dingaling/gtalk[EXAMPLE]/+1$
1 at voice.google.com"/>*
   </condition>
 </extension>

Pay special attention to the bolded areas of the configuration example I've
provided. Configuring Google talk to work with FreeSwitch is quite
straightforward with my configuration example I've provided above. I've ran
to this same configuration ambiguity when they first completed this feature
anyways, so I hope my configuration example does help you into resolving
that problem

On Wed, Oct 13, 2010 at 3:07 PM, Rob Hutton <justlikeef at gmail.com> wrote:
> Caught and changed that, but it does the same thing.  I also changed the
profile name to gtalk in case there was a case sensitivity issue or
something, but same result.
>
> I am told that the wiki is not quite correct either, but with everyone
away from their development PCs, it will be the weekend before the
corrections are available.
>
> --
> Thanks,
> Rob
> On Wednesday 13 October 2010 17:51:07 Michael Collins wrote:
>> > Here is the profile:
>> >
>> >  <profile name="dingaling_1">
>> >
>> I believe that this should be:
>> <profile type="client">
>>
>> -MC
>>
> I
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From curriegrad2004 at gmail.com  Wed Oct 13 22:34:15 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Wed, 13 Oct 2010 22:34:15 -0700
Subject: [Freeswitch-users] devel-bootstrap.sh - what is it for?
Message-ID: <AANLkTimtho9URq4hJnw=7Kh-vj2nN26QSL1WTn=q+Dac@mail.gmail.com>

Today when I was pulling the latest revision of freeswitch via git and
I noticed git created a new file called devel-bootstrap.sh. I'm
wondering what this new devel-bootstrap script does and how different
is it from the bootstrap.sh script we're used to when we build
freeswitch from sources.


From kdjakovic at hotmail.com  Thu Oct 14 00:54:25 2010
From: kdjakovic at hotmail.com (katarina djakovic)
Date: Thu, 14 Oct 2010 00:54:25 -0700
Subject: [Freeswitch-users] ACL and Digest authentication problem
In-Reply-To: <AANLkTinWQ65Ah_Vy6QDt=GC60g28eukv01M1Yz9BSocO@mail.gmail.com>
References: <SNT137-w5155113C52492E566CEC53B3540@phx.gbl>,
	<AANLkTinWQ65Ah_Vy6QDt=GC60g28eukv01M1Yz9BSocO@mail.gmail.com>
Message-ID: <SNT137-w9F4268DF625DD71A3C514B3560@phx.gbl>


Dear Ognjen,
 
thanks a lot. As you are saying the FS default behavour is  such when <param name="apply-register-acl" value=.../> is set in a sip profile, then, Register doesn't fall back to Digest authentication (in case when caller does not belong to the acl list).
 
So, to acomplish what we wanted we configured 2 sip profiles, one to handle ACL registrations/calls and another to handle Digest authentication registrations/calls and solved our problem.
 
Thanks again,
Katarina
 
 


Date: Tue, 12 Oct 2010 17:15:57 +0200
From: oseslija at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] ACL and Digest authentication problem


Hello Katarina, 

I can answer your questions in (I believe) our mother tongue.


On Tue, Oct 12, 2010 at 3:12 PM, katarina djakovic <kdjakovic at hotmail.com> wrote:


Dear FreeSwitch users,

we need some help about ACL and Digest authenication.  

This is what we want:

1) We want certain users to be authenticated through ACL (certain IP addresses) including both Register and Invite messages. In other words, we want those users to be granted access to our FS withouth having to authenticate with username and password when registering or calling.
2) On the other hand, if users don't fall into our ACL list (registering/calling from other IP addresses) we want them to authenticate normally throught Digest authentication (username/password).



2) je FreeSWITCH-ov default konfiguracija.
 

We tried to configure FS for our needs, but we didn't acomplished what we wanted. Namely, now, for any users that do not belong to the ACL list our FS will reject their registration and will NOT fall back to Digest authentication. In other words, our FS will let all users that fall into ACL list register and call without authenticating --- but all others will be rejected on the attempt to register (debug trace says: sofia_reg.c IP YY.YY.YY.YY Rejected by register acl "domains") and will not let them fall back to Digest authentication.



Ako se koristi register acl FS ne koristi fallback na Digest. Ovo ne vazi za INVITE-e gde to radi.

 

These are our settings:

    a) acl.conf.xml: 
        <configuration name="acl.conf" description="Network Lists">
          <network-lists>

          <!--
        This will traverse the directory adding all users 
         with the cidr= tag to this ACL, when this ACL matches
        the users variables and params apply as if they 
        digest authenticated.
          -->
          <list name="domains" default="deny">
            <node type="allow" domain="$${domain}"/>
            <node type="allow" domain="XX.XX.XX.XX/32"/>
      
          </list>

          </network-lists>
        </configuration>

b) sip profile:

   <param name="apply-inbound-acl" value="domains"/>
   <param name="apply-register-acl" value="domains"/>
   <param name="auth-calls" value="true"/>

c) users that fall into ACL will have a cidr parameter set aproprietelly <user id="2000" mailbox="2000" cidr="XX.XX.XX.XX/32">

Other users, that we want to be authenticated through Digest authentication will not have anything related to ACL in their user profiles in the Directory.

2) On the other hand, if we remove the <param name="apply-register-acl" value="domains"/> from the sip profile, then users that do not belong to the ACL list will register normally and when calling - their calls (Invite) will fall back to digest authentication (here is the debug: "sofia.c:5847 IP YY.YY.YY.YY Rejected by acl "domains". Falling back to Digest auth.).

That is fine with us - but then we have a different problem, then the users from the ACL list will be asked to register by username/password credentials, i.e. their registration will have to authenticated and that is not what we wanted.


We are mistaging somewhere. Hopefully what I wrote makes sense and maybe someone could help us configure the system to fit our needs.



Kao sto sam rekao ovo je podrazumevana opcija.

 



 

Many thanks in advance,
Katarina 


Regards,
Ognjen

irc #freeswitch: sekil
 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 		 	   		  
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From tear152 at hotmail.com  Thu Oct 14 00:55:52 2010
From: tear152 at hotmail.com (=?big5?B?pr8gQ2hpYW5nIKbcpKQgQ2hpaC1DaHVuZyAgIFd5Ymll?=)
Date: Thu, 14 Oct 2010 15:55:52 +0800
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>,
	<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>,
	<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>,
	<SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>,
	<AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>
Message-ID: <SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>


I tried Connect Two FreeSWITCH Boxes  http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
and my configuration :
 
60.248.175.37
 
(autoload_configs/act.conf.xml)
<list name="domains" default="deny">
      <node type="allow" cidr="60.248.175.38/28"/>
     <!-- <node type="allow" domain="$${domain}"/>-->
</list>

(dialplan/default.xml)
 
<extension name="Dial to 38">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="bridge" data="sofia/internal/$1 at 60.248.175.38"/>
      </condition>
 </extension>
 
(dialplan/public.xml)

<extension name="Calls from 38">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="transfer" data="$1 XML default"/>
      </condition>
 </extension>
 
 
 
60.248.175.38
 
(autoload_configs/act.conf.xml)
<list name="domains" default="deny">
      <node type="allow" cidr="60.248.175.37/28"/>
     <!-- <node type="allow" domain="$${domain}"/>-->
</list>

(dialplan/default.xml)
 
<extension name="Dial to 37">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="bridge" data="sofia/internal/$1 at 60.248.175.37"/>
      </condition>
 </extension>
 
(dialplan/public.xml)

<extension name="Calls from 37">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="transfer" data="$1 XML default"/>
      </condition>
 </extension>
 
when I use 1007 at 38 calls to 1007 at 37
it won't show errors but still not work
 
how should I change something ?
 
Thank you for your attension
 
Best Regards
Gary
 
 
 
 
 
> Date: Wed, 13 Oct 2010 22:17:13 -0700
> From: curriegrad2004 at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> 
> You can create a SIP trunk between the two servers so calls can be
> routed back and forth from the servers you've specified.
> 
> 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
> > I tried it
> > but not working
> > and show errors below
> > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493 Cannot create
> > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493 Cannot create
> > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate Failed.
> > Cause: USER_NOT_REGISTERED
> >
> > and for example
> >  I use SIP server 60.248.175.38  user id=1007
> > and I call to 1007 at 60.248.175.37    that will show these errors
> >
> > how should set more?
> >
> > thank you for your attension
> >
> > Best Regards
> > Gary
> >
> > ________________________________
> > Date: Thu, 14 Oct 2010 10:55:49 +0800
> > From: xyangni at gmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >
> > I think you can reg at server A and than call B as a normal SIP call to
> > external domain.
> >
> > 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
> >
> >
> >
> > Hello
> >
> > I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X-Lite to call
> >
> > my question is how can I call between these servers?
> >
> > if I can , how shoould I set in *.xml? or X-Lite?
> >
> > thank you for your attension
> >
> >
> > Best Regards
> > Gary
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________ FreeSWITCH-users mailing
> > list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
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From david.ponzone at ipeva.fr  Thu Oct 14 01:16:28 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 14 Oct 2010 10:16:28 +0200
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>,
	<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>,
	<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>,
	<SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>,
	<AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>
	<SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>
Message-ID: <C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>

I think we would need a complete log from the target FS, in order to  
provide you with any efficient help.

Thank

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 14/10/2010 ? 09:55, ? Chiang ?? Chih-Chung Wybie a ?crit :

> I tried Connect Two FreeSWITCH Boxes  http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
> and my configuration :
>
> 60.248.175.37
>
> (autoload_configs/act.conf.xml)
> <list name="domains" default="deny">
>       <node type="allow" cidr="60.248.175.38/28"/>
>      <!-- <node type="allow" domain="$${domain}"/>-->
> </list>
>
> (dialplan/default.xml)
>
> <extension name="Dial to 38">
>       <condition field="destination_number" expression="^([0-4]0[01] 
> [0-9])$">
>         <action application="bridge" data="sofia/internal/$1 at 60.248.175.38 
> "/>
>       </condition>
>  </extension>
>
> (dialplan/public.xml)
>
> <extension name="Calls from 38">
>       <condition field="destination_number" expression="^([0-4]0[01] 
> [0-9])$">
>         <action application="transfer" data="$1 XML default"/>
>       </condition>
>  </extension>
>
>
>
> 60.248.175.38
>
> (autoload_configs/act.conf.xml)
> <list name="domains" default="deny">
>       <node type="allow" cidr="60.248.175.37/28"/>
>      <!-- <node type="allow" domain="$${domain}"/>-->
> </list>
>
> (dialplan/default.xml)
>
> <extension name="Dial to 37">
>       <condition field="destination_number" expression="^([0-4]0[01] 
> [0-9])$">
>         <action application="bridge" data="sofia/internal/$1 at 60.248.175.37 
> "/>
>       </condition>
>  </extension>
>
> (dialplan/public.xml)
>
> <extension name="Calls from 37">
>       <condition field="destination_number" expression="^([0-4]0[01] 
> [0-9])$">
>         <action application="transfer" data="$1 XML default"/>
>       </condition>
>  </extension>
>
> when I use 1007 at 38 calls to 1007 at 37
> it won't show errors but still not work
>
> how should I change something ?
>
> Thank you for your attension
>
> Best Regards
> Gary
>
>
>
>
>
> > Date: Wed, 13 Oct 2010 22:17:13 -0700
> > From: curriegrad2004 at gmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X- 
> Lite
> >
> > You can create a SIP trunk between the two servers so calls can be
> > routed back and forth from the servers you've specified.
> >
> > 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
> > > I tried it
> > > but not working
> > > and show errors below
> > > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493  
> Cannot create
> > > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> > > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493  
> Cannot create
> > > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> > > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate  
> Failed.
> > > Cause: USER_NOT_REGISTERED
> > >
> > > and for example
> > >  I use SIP server 60.248.175.38  user id=1007
> > > and I call to 1007 at 60.248.175.37    that will show these errors
> > >
> > > how should set more?
> > >
> > > thank you for your attension
> > >
> > > Best Regards
> > > Gary
> > >
> > > ________________________________
> > > Date: Thu, 14 Oct 2010 10:55:49 +0800
> > > From: xyangni at gmail.com
> > > To: freeswitch-users at lists.freeswitch.org
> > > Subject: Re: [Freeswitch-users] some questions of freeswitch & X- 
> Lite
> > >
> > > I think you can reg at server A and than call B as a normal SIP  
> call to
> > > external domain.
> > >
> > > 2010/10/14 ? Chiang ?? Chih-Chung Wybie  
> <tear152 at hotmail.com>
> > >
> > >
> > >
> > > Hello
> > >
> > > I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X- 
> Lite to call
> > >
> > > my question is how can I call between these servers?
> > >
> > > if I can , how shoould I set in *.xml? or X-Lite?
> > >
> > > thank you for your attension
> > >
> > >
> > > Best Regards
> > > Gary
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> > >
> > > _______________________________________________ FreeSWITCH-users  
> mailing
> > > list FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From babak.freeswitch at gmail.com  Thu Oct 14 01:40:20 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Thu, 14 Oct 2010 12:10:20 +0330
Subject: [Freeswitch-users] mod_managed socket binding problem
In-Reply-To: <1286989107863-5631782.post@n2.nabble.com>
References: <AANLkTinqR-8GTQ7qAOM=u5TFEoDbX__qAt7p8V++D06c@mail.gmail.com>
	<D51C1FE5-C34E-4B04-95AF-1D84B6AAFCA0@freeswitch.org>
	<AANLkTimn5wPsR18CNxXPV0kzG4Jm1ayQ3j1=QUpuUsvY@mail.gmail.com>
	<1286989107863-5631782.post@n2.nabble.com>
Message-ID: <AANLkTik3CBc=3Quf6P=1apd9-9MWD=h=0kJKvNt4umyH@mail.gmail.com>

The problem is all of this functions (Dispose, Destructor ...) are called
after the new instance of module is created, so on the creation of the new
instance because still the old one is listening on for example port 9999 the
new one can not bind to the port. I've found
switch_loadable_module_function_table_t which seems to be where I should add
my shut down routine but I can not find a way to do so
thanx
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From mnhassan at usa.net  Thu Oct 14 01:49:43 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Thu, 14 Oct 2010 14:49:43 +0600
Subject: [Freeswitch-users] ACL and Digest authentication problem
In-Reply-To: <SNT137-w9F4268DF625DD71A3C514B3560@phx.gbl>
References: <SNT137-w5155113C52492E566CEC53B3540@phx.gbl>
	<AANLkTinWQ65Ah_Vy6QDt=GC60g28eukv01M1Yz9BSocO@mail.gmail.com>
	<SNT137-w9F4268DF625DD71A3C514B3560@phx.gbl>
Message-ID: <AANLkTimdd__QNNOU-SGXCoXHrD9Gmgu=7-2QXw7=APgo@mail.gmail.com>

Another nice way to do that on the same profile is to not have any ACL
defined at all, instead put a "cidr=" parameter in the directory entry
for a user.

Regards
HASSAN


On 2010-10-14, katarina djakovic <kdjakovic at hotmail.com> wrote:
>
> Dear Ognjen,
>
> thanks a lot. As you are saying the FS default behavour is  such when <param
> name="apply-register-acl" value=.../> is set in a sip profile, then,
> Register doesn't fall back to Digest authentication (in case when caller
> does not belong to the acl list).
>
> So, to acomplish what we wanted we configured 2 sip profiles, one to handle
> ACL registrations/calls and another to handle Digest authentication
> registrations/calls and solved our problem.
>
> Thanks again,
> Katarina
>
>
>
>
> Date: Tue, 12 Oct 2010 17:15:57 +0200
> From: oseslija at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] ACL and Digest authentication problem
>
>
> Hello Katarina,
>
> I can answer your questions in (I believe) our mother tongue.
>
>
> On Tue, Oct 12, 2010 at 3:12 PM, katarina djakovic <kdjakovic at hotmail.com>
> wrote:
>
>
> Dear FreeSwitch users,
>
> we need some help about ACL and Digest authenication.
>
> This is what we want:
>
> 1) We want certain users to be authenticated through ACL (certain IP
> addresses) including both Register and Invite messages. In other words, we
> want those users to be granted access to our FS withouth having to
> authenticate with username and password when registering or calling.
> 2) On the other hand, if users don't fall into our ACL list
> (registering/calling from other IP addresses) we want them to authenticate
> normally throught Digest authentication (username/password).
>
>
>
> 2) je FreeSWITCH-ov default konfiguracija.
>
>
> We tried to configure FS for our needs, but we didn't acomplished what we
> wanted. Namely, now, for any users that do not belong to the ACL list our FS
> will reject their registration and will NOT fall back to Digest
> authentication. In other words, our FS will let all users that fall into ACL
> list register and call without authenticating --- but all others will be
> rejected on the attempt to register (debug trace says: sofia_reg.c IP
> YY.YY.YY.YY Rejected by register acl "domains") and will not let them fall
> back to Digest authentication.
>
>
>
> Ako se koristi register acl FS ne koristi fallback na Digest. Ovo ne vazi za
> INVITE-e gde to radi.
>
>
>
> These are our settings:
>
>     a) acl.conf.xml:
>         <configuration name="acl.conf" description="Network Lists">
>           <network-lists>
>
>           <!--
>         This will traverse the directory adding all users
>          with the cidr= tag to this ACL, when this ACL matches
>         the users variables and params apply as if they
>         digest authenticated.
>           -->
>           <list name="domains" default="deny">
>             <node type="allow" domain="$${domain}"/>
>             <node type="allow" domain="XX.XX.XX.XX/32"/>
>
>           </list>
>
>           </network-lists>
>         </configuration>
>
> b) sip profile:
>
>    <param name="apply-inbound-acl" value="domains"/>
>    <param name="apply-register-acl" value="domains"/>
>    <param name="auth-calls" value="true"/>
>
> c) users that fall into ACL will have a cidr parameter set aproprietelly
> <user id="2000" mailbox="2000" cidr="XX.XX.XX.XX/32">
>
> Other users, that we want to be authenticated through Digest authentication
> will not have anything related to ACL in their user profiles in the
> Directory.
>
> 2) On the other hand, if we remove the <param name="apply-register-acl"
> value="domains"/> from the sip profile, then users that do not belong to the
> ACL list will register normally and when calling - their calls (Invite) will
> fall back to digest authentication (here is the debug: "sofia.c:5847 IP
> YY.YY.YY.YY Rejected by acl "domains". Falling back to Digest auth.).
>
> That is fine with us - but then we have a different problem, then the users
> from the ACL list will be asked to register by username/password
> credentials, i.e. their registration will have to authenticated and that is
> not what we wanted.
>
>
> We are mistaging somewhere. Hopefully what I wrote makes sense and maybe
> someone could help us configure the system to fit our needs.
>
>
>
> Kao sto sam rekao ovo je podrazumevana opcija.
>
>
>
>
>
>
>
> Many thanks in advance,
> Katarina
>
>
> Regards,
> Ognjen
>
> irc #freeswitch: sekil
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 		 	   		

-- 
Sent from my mobile device


From mgg at giagnocavo.net  Thu Oct 14 02:03:22 2010
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Thu, 14 Oct 2010 05:03:22 -0400
Subject: [Freeswitch-users] mod_managed socket binding problem
In-Reply-To: <AANLkTik3CBc=3Quf6P=1apd9-9MWD=h=0kJKvNt4umyH@mail.gmail.com>
References: <AANLkTinqR-8GTQ7qAOM=u5TFEoDbX__qAt7p8V++D06c@mail.gmail.com>
	<D51C1FE5-C34E-4B04-95AF-1D84B6AAFCA0@freeswitch.org>
	<AANLkTimn5wPsR18CNxXPV0kzG4Jm1ayQ3j1=QUpuUsvY@mail.gmail.com>
	<1286989107863-5631782.post@n2.nabble.com>
	<AANLkTik3CBc=3Quf6P=1apd9-9MWD=h=0kJKvNt4umyH@mail.gmail.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C670367D5CA8F@mse17be1.mse17.exchange.ms>

You're correct, it's not possible to work around this very easily via mod_managed. The issue is that there can be one or more outstanding calls (sessions or API calls) to your module, so the loading code loads up the new version, installs it for new calls, then waits for the reference count to hit zero before unloading the old one. And it doesn't have to be just one - as long as there's something using a module, that version will stay in memory regardless of how many newer versions are loaded. I'm open to suggestions on how mod_managed could do the loading differently. The native FreeSWITCH functions won't be of any help, as FS is unaware of the internal mod_managed module loading.

As far as sorting out your situation right now: What I'd do is on module load, get a named mutex. If you get ownership, then you're the active version and can take over resources (socket bindings) and so on. Open a named pipe (System.IO.Pipes) and listen for messages. The next version of the module to load also gets the same named mutex, but it won't acquire ownership. So, it sends a message down the named pipe. The receiver of the message knows it's life is up, so it closes its ports and releases the mutex. The new version is waiting on this mutex. When it acquires the mutex, then it's the active version and opens all resources, and so on.

That's not a very complete solution. For instance, 4 copies of the module could be in memory, and the mutex wait is not guaranteed to be fair. So a starting module should have some way of reading shared memory to determine if the current active version is newer than itself, and if so, not attempt to take ownership. Fortunately, I think mod_managed won't start loading a new version until it's finished loading a previous one. So, it's probably easy to sort out any race issues there.

-Michael


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of babak yakhchali
Sent: Thursday, October 14, 2010 2:40 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_managed socket binding problem

The problem is all of this functions (Dispose, Destructor ...) are called after the new instance of module is created, so on the creation of the new instance because still the old one is listening on for example port 9999 the new one can not bind to the port. I've found switch_loadable_module_function_table_t which seems to be where I should add my shut down routine but I can not find a way to do so
thanx
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From tear152 at hotmail.com  Thu Oct 14 02:53:26 2010
From: tear152 at hotmail.com (=?gb2312?B?va0gQ2hpYW5nINbB1tAgQ2hpaC1DaHVuZyAgIFd5Ymll?=)
Date: Thu, 14 Oct 2010 17:53:26 +0800
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>, ,
	<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>, ,
	<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>, ,
	<SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>, ,
	<AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>,
	<SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>,
	<C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>
Message-ID: <SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>


I tried use 1013 at 38 calls to 1019 at 37
 
show log below
 
2010-10-14 17:50:41.249876 [NOTICE] switch_channel.c:675 New Channel sofia/internal/1013 at 60.248.175.38 [875fdeef-b94e-41fb-a621-ea005bbaedbd]
2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing 1013->1019 in context public
2010-10-14 17:50:41.263809 [NOTICE] switch_ivr.c:1450 Transfer sofia/internal/1013 at 60.248.175.38 to XML[1019 at default]
2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing 1013->1019 in context default
2010-10-14 17:50:41.283904 [INFO] switch_core_session.c:1764 Sending early media
2010-10-14 17:50:41.285023 [NOTICE] mod_sofia.c:1994 Pre-Answer sofia/internal/1013 at 60.248.175.38!

seems not work
 
Thank you for you attension
Best Regards
 
Gary
 

 


From: david.ponzone at ipeva.fr
To: freeswitch-users at lists.freeswitch.org
Date: Thu, 14 Oct 2010 10:16:28 +0200
Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite

I think we would need a complete log from the target FS, in order to provide you with any efficient help.


Thank





David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34


Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr  -   www.ipeva-studio.com



Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 14/10/2010 ? 09:55, ? Chiang ?? Chih-Chung Wybie a ?crit :

I tried Connect Two FreeSWITCH Boxes  http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
and my configuration :
 
60.248.175.37
 
(autoload_configs/act.conf.xml)
<list name="domains" default="deny">
      <node type="allow" cidr="60.248.175.38/28"/>
     <!-- <node type="allow" domain="$${domain}"/>-->
</list>

(dialplan/default.xml)
 
<extension name="Dial to 38">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="bridge" data="sofia/internal/$1 at 60.248.175.38"/>
      </condition>
 </extension>
 
(dialplan/public.xml)

<extension name="Calls from 38">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="transfer" data="$1 XML default"/>
      </condition>
 </extension>
 
 
 
60.248.175.38
 
(autoload_configs/act.conf.xml)
<list name="domains" default="deny">
      <node type="allow" cidr="60.248.175.37/28"/>
     <!-- <node type="allow" domain="$${domain}"/>-->
</list>

(dialplan/default.xml)
 
<extension name="Dial to 37">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="bridge" data="sofia/internal/$1 at 60.248.175.37"/>
      </condition>
 </extension>
 
(dialplan/public.xml)

<extension name="Calls from 37">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="transfer" data="$1 XML default"/>
      </condition>
 </extension>
 
when I use 1007 at 38 calls to 1007 at 37
it won't show errors but still not work
 
how should I change something ?
 
Thank you for your attension
 
Best Regards
Gary
 
 
 
 
 
> Date: Wed, 13 Oct 2010 22:17:13 -0700
> From: curriegrad2004 at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> 
> You can create a SIP trunk between the two servers so calls can be
> routed back and forth from the servers you've specified.
> 
> 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
> > I tried it
> > but not working
> > and show errors below
> > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493 Cannot create
> > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493 Cannot create
> > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate Failed.
> > Cause: USER_NOT_REGISTERED
> >
> > and for example
> >  I use SIP server 60.248.175.38  user id=1007
> > and I call to 1007 at 60.248.175.37    that will show these errors
> >
> > how should set more?
> >
> > thank you for your attension
> >
> > Best Regards
> > Gary
> >
> > ________________________________
> > Date: Thu, 14 Oct 2010 10:55:49 +0800
> > From: xyangni at gmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >
> > I think you can reg at server A and than call B as a normal SIP call to
> > external domain.
> >
> > 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
> >
> >
> >
> > Hello
> >
> > I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X-Lite to call
> >
> > my question is how can I call between these servers?
> >
> > if I can , how shoould I set in *.xml? or X-Lite?
> >
> > thank you for your attension
> >
> >
> > Best Regards
> > Gary
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________ FreeSWITCH-users mailing
> > list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 		 	   		  
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From david.ponzone at ipeva.fr  Thu Oct 14 03:20:56 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 14 Oct 2010 12:20:56 +0200
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>, ,
	<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>, ,
	<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>, ,
	<SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>, ,
	<AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>,
	<SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>,
	<C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>
	<SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>
Message-ID: <6A8D7D0E-4DF5-4F35-8A00-F52FECC4F826@ipeva.fr>

Don't be shy on the logs, send the full ones!

in fs_cli:
fsctl loglevel debug
/log 7

and then make a call

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 14/10/2010 ? 11:53, ? Chiang ?? Chih-Chung Wybie a ?crit :

> I tried use 1013 at 38 calls to 1019 at 37
>
> show log below
>
> 2010-10-14 17:50:41.249876 [NOTICE] switch_channel.c:675 New Channel sofia/internal/1013 at 60.248.175.38 
> [875fdeef-b94e-41fb-a621-ea005bbaedbd]
> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing  
> 1013->1019 in context public
> 2010-10-14 17:50:41.263809 [NOTICE] switch_ivr.c:1450 Transfer sofia/internal/1013 at 60.248.175.38 
>  to XML[1019 at default]
> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing  
> 1013->1019 in context default
> 2010-10-14 17:50:41.283904 [INFO] switch_core_session.c:1764 Sending  
> early media
> 2010-10-14 17:50:41.285023 [NOTICE] mod_sofia.c:1994 Pre-Answer sofia/internal/1013 at 60.248.175.38 
> !
>
> seems not work
>
> Thank you for you attension
> Best Regards
>
> Gary
>
>
>
> From: david.ponzone at ipeva.fr
> To: freeswitch-users at lists.freeswitch.org
> Date: Thu, 14 Oct 2010 10:16:28 +0200
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> I think we would need a complete log from the target FS, in order to  
> provide you with any efficient help.
>
> Thank
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
> message, merci de le d?truire imm?diatement et d'avertir  
> l'exp?diteur.
>
>
>
>
> Le 14/10/2010 ? 09:55, ? Chiang ?? Chih-Chung Wybie a ?crit :
>
> I tried Connect Two FreeSWITCH Boxes  http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
> and my configuration :
>
> 60.248.175.37
>
> (autoload_configs/act.conf.xml)
> <list name="domains" default="deny">
>       <node type="allow" cidr="60.248.175.38/28"/>
>      <!-- <node type="allow" domain="$${domain}"/>-->
> </list>
>
> (dialplan/default.xml)
>
> <extension name="Dial to 38">
>       <condition field="destination_number" expression="^([0-4]0[01] 
> [0-9])$">
>         <action application="bridge" data="sofia/internal/$1 at 60.248.175.38 
> "/>
>       </condition>
>  </extension>
>
> (dialplan/public.xml)
>
> <extension name="Calls from 38">
>       <condition field="destination_number" expression="^([0-4]0[01] 
> [0-9])$">
>         <action application="transfer" data="$1 XML default"/>
>       </condition>
>  </extension>
>
>
>
> 60.248.175.38
>
> (autoload_configs/act.conf.xml)
> <list name="domains" default="deny">
>       <node type="allow" cidr="60.248.175.37/28"/>
>      <!-- <node type="allow" domain="$${domain}"/>-->
> </list>
>
> (dialplan/default.xml)
>
> <extension name="Dial to 37">
>       <condition field="destination_number" expression="^([0-4]0[01] 
> [0-9])$">
>         <action application="bridge" data="sofia/internal/$1 at 60.248.175.37 
> "/>
>       </condition>
>  </extension>
>
> (dialplan/public.xml)
>
> <extension name="Calls from 37">
>       <condition field="destination_number" expression="^([0-4]0[01] 
> [0-9])$">
>         <action application="transfer" data="$1 XML default"/>
>       </condition>
>  </extension>
>
> when I use 1007 at 38 calls to 1007 at 37
> it won't show errors but still not work
>
> how should I change something ?
>
> Thank you for your attension
>
> Best Regards
> Gary
>
>
>
>
>
> > Date: Wed, 13 Oct 2010 22:17:13 -0700
> > From: curriegrad2004 at gmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X- 
> Lite
> >
> > You can create a SIP trunk between the two servers so calls can be
> > routed back and forth from the servers you've specified.
> >
> > 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
> > > I tried it
> > > but not working
> > > and show errors below
> > > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493  
> Cannot create
> > > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> > > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493  
> Cannot create
> > > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> > > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate  
> Failed.
> > > Cause: USER_NOT_REGISTERED
> > >
> > > and for example
> > >  I use SIP server 60.248.175.38  user id=1007
> > > and I call to 1007 at 60.248.175.37    that will show these errors
> > >
> > > how should set more?
> > >
> > > thank you for your attension
> > >
> > > Best Regards
> > > Gary
> > >
> > > ________________________________
> > > Date: Thu, 14 Oct 2010 10:55:49 +0800
> > > From: xyangni at gmail.com
> > > To: freeswitch-users at lists.freeswitch.org
> > > Subject: Re: [Freeswitch-users] some questions of freeswitch & X- 
> Lite
> > >
> > > I think you can reg at server A and than call B as a normal SIP  
> call to
> > > external domain.
> > >
> > > 2010/10/14 ? Chiang ?? Chih-Chung Wybie  
> <tear152 at hotmail.com>
> > >
> > >
> > >
> > > Hello
> > >
> > > I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X- 
> Lite to call
> > >
> > > my question is how can I call between these servers?
> > >
> > > if I can , how shoould I set in *.xml? or X-Lite?
> > >
> > > thank you for your attension
> > >
> > >
> > > Best Regards
> > > Gary
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> > >
> > > _______________________________________________ FreeSWITCH-users  
> mailing
> > > list FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________ FreeSWITCH-users  
> mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
>  http:// 
> www.freeswitch.org_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From babak.freeswitch at gmail.com  Thu Oct 14 04:02:47 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Thu, 14 Oct 2010 14:32:47 +0330
Subject: [Freeswitch-users] mod_managed socket binding problem
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C670367D5CA8F@mse17be1.mse17.exchange.ms>
References: <AANLkTinqR-8GTQ7qAOM=u5TFEoDbX__qAt7p8V++D06c@mail.gmail.com>
	<D51C1FE5-C34E-4B04-95AF-1D84B6AAFCA0@freeswitch.org>
	<AANLkTimn5wPsR18CNxXPV0kzG4Jm1ayQ3j1=QUpuUsvY@mail.gmail.com>
	<1286989107863-5631782.post@n2.nabble.com>
	<AANLkTik3CBc=3Quf6P=1apd9-9MWD=h=0kJKvNt4umyH@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C670367D5CA8F@mse17be1.mse17.exchange.ms>
Message-ID: <AANLkTinLn6-h=Wy=xoRK9CQm0a6W5srjCB5TTPrfEX+j@mail.gmail.com>

Thank u very very much Micheal.
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From william.suffill at gmail.com  Thu Oct 14 07:40:09 2010
From: william.suffill at gmail.com (William Suffill)
Date: Thu, 14 Oct 2010 10:40:09 -0400
Subject: [Freeswitch-users] devel-bootstrap.sh - what is it for?
In-Reply-To: <AANLkTimtho9URq4hJnw=7Kh-vj2nN26QSL1WTn=q+Dac@mail.gmail.com>
References: <AANLkTimtho9URq4hJnw=7Kh-vj2nN26QSL1WTn=q+Dac@mail.gmail.com>
Message-ID: <AANLkTikmhehS4BKEuNPAKRQjhuCXCWPyZhfMCdWUFcC1@mail.gmail.com>

the devel-bootstrap sets gdb compile flags then runs bootstrap -j and
./configure
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From brian at freeswitch.org  Thu Oct 14 07:47:17 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 14 Oct 2010 09:47:17 -0500
Subject: [Freeswitch-users] CCM Interop
Message-ID: <2DBA14F9-8ABC-4717-BDEB-EAEC11375179@freeswitch.org>

We are looking for a SIP trunk from Cisco Call Manager pointed at one of our Dev boxes so we can do interop. to ensure it works with FreeSWITCH.  

If you have one can you please find me on IRC bkw_ or email me.

Thanks you.

/b



From davidwaf at gmail.com  Thu Oct 14 07:51:13 2010
From: davidwaf at gmail.com (David Wafula)
Date: Thu, 14 Oct 2010 16:51:13 +0200
Subject: [Freeswitch-users] SIP Registration Failing
In-Reply-To: <AANLkTimcMg0Q_29vh7bHF9NKu2jXwEZQhrEnLM8HcPwX@mail.gmail.com>
References: <AANLkTi=1OQ4uUS8Ecb-NaoM6AT3--fgGquqb_nvvJzdV@mail.gmail.com>
	<B8B73B88-95A0-44A7-A921-DB620F33AA43@ipeva.fr>
	<AANLkTimcMg0Q_29vh7bHF9NKu2jXwEZQhrEnLM8HcPwX@mail.gmail.com>
Message-ID: <AANLkTikmspNb+gbgxktn-=67miNefsh24nYEury7EK35@mail.gmail.com>

On Mon, Oct 11, 2010 at 9:49 PM, David Wafula <davidwaf at gmail.com> wrote:

>
>>
> i shall certainly post it here as soon as i get back into the network. So
> it is possible the problem could be with my network configuration .
>


Just to confirm that now i got the java softphone (
http://sourceforge.net/projects/peers/) working nicely with Freeswitch,on my
LAN. Next, going to try to make calls from outside the LAN and see how it
goes.  The problems i was experiencing had all to do my network
configurations (laptop is both wired and on wireless, the softphone had
issues deciding which of the IPs to use, so i switched off wireless).

-- 
David Wafula
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From mkellem at vontoo.com  Thu Oct 14 07:56:11 2010
From: mkellem at vontoo.com (Marc Kellem)
Date: Thu, 14 Oct 2010 10:56:11 -0400
Subject: [Freeswitch-users] mod_cepstral 5.1 known issues?
Message-ID: <AANLkTik8KZdxAnUUfv0nKF6EkTHe559PtC2bF74eng11@mail.gmail.com>

The mod_cepstral wiki page says "Warning: the suggested version to use
is 4.x since there are known issues with 5.1 (which is closed source)".
 What exactly are the known issues?

http://wiki.freeswitch.org/wiki/Mod_cepstral

Thanks,
Marc Kellem
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From null at invalid.name  Thu Oct 14 08:21:47 2010
From: null at invalid.name (Dan Lane)
Date: Thu, 14 Oct 2010 16:21:47 +0100
Subject: [Freeswitch-users] Speex VBR and VAD
Message-ID: <AANLkTi=qqrBNL3d+_tcEfpL55vd+o=kQgS2pJaTnH4QB@mail.gmail.com>

Hi,

Is there any way to tell Freeswitch to enable VBR and VAD on Speex
channels as it appears to be disabled by default and I can't find any
mention of it in the wiki or previous mailing list posts.

The client is sending the following SDP:

a=rtpmap:111 speex/16000/1
a=fmtp:111 vbr=on
a=rtpmap:110 speex/8000/1
a=fmtp:110 vbr=on
a=rtpmap:0 GSM/22050/1
a=rtpmap:0 GSM/11025/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:113 iLBC/8000/1
a=fmtp:113 mode=30
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-11
a=sendrecv


And receiving back the following without vbr=on which disables it in the client:

a=rtpmap:111 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20



Cheers,
Dan


From anthony.minessale at gmail.com  Thu Oct 14 08:26:32 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Oct 2010 10:26:32 -0500
Subject: [Freeswitch-users] mod_cepstral 5.1 known issues?
In-Reply-To: <AANLkTik8KZdxAnUUfv0nKF6EkTHe559PtC2bF74eng11@mail.gmail.com>
References: <AANLkTik8KZdxAnUUfv0nKF6EkTHe559PtC2bF74eng11@mail.gmail.com>
Message-ID: <AANLkTi=MvWh14MQ9wPEboXZmMgAtURAY5zBoobn32Pkq@mail.gmail.com>

random deadlocks in the cepstral engine blocking all the speech
generation calls.


On Thu, Oct 14, 2010 at 9:56 AM, Marc Kellem <mkellem at vontoo.com> wrote:
> The mod_cepstral wiki page says "Warning: the suggested version to use
> is?4.x?since there are known issues with?5.1?(which is closed source)".
> ?What exactly are the known issues?
> http://wiki.freeswitch.org/wiki/Mod_cepstral
> Thanks,
> Marc Kellem
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From mkellem at vontoo.com  Thu Oct 14 08:58:07 2010
From: mkellem at vontoo.com (Marc Kellem)
Date: Thu, 14 Oct 2010 11:58:07 -0400
Subject: [Freeswitch-users] mod_cepstral 5.1 known issues?
In-Reply-To: <AANLkTi=MvWh14MQ9wPEboXZmMgAtURAY5zBoobn32Pkq@mail.gmail.com>
References: <AANLkTik8KZdxAnUUfv0nKF6EkTHe559PtC2bF74eng11@mail.gmail.com>
	<AANLkTi=MvWh14MQ9wPEboXZmMgAtURAY5zBoobn32Pkq@mail.gmail.com>
Message-ID: <AANLkTimj=SwX7d1sot+NhOxMgD4F-PN=bM3hxrh3bF3O@mail.gmail.com>

Is anyone using mod_unimrcp to access Cepstral 5.1 voices in a UniMRCP
server?

On Thu, Oct 14, 2010 at 11:26 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> random deadlocks in the cepstral engine blocking all the speech
> generation calls.
>
>
> On Thu, Oct 14, 2010 at 9:56 AM, Marc Kellem <mkellem at vontoo.com> wrote:
> > The mod_cepstral wiki page says "Warning: the suggested version to use
> > is 4.x since there are known issues with 5.1 (which is closed source)".
> >  What exactly are the known issues?
> > http://wiki.freeswitch.org/wiki/Mod_cepstral
> > Thanks,
> > Marc Kellem
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From john at 247-talk.co.uk  Thu Oct 14 05:04:27 2010
From: john at 247-talk.co.uk (John Carpenter)
Date: Thu, 14 Oct 2010 13:04:27 +0100
Subject: [Freeswitch-users] H323 to Sip conversion problem
In-Reply-To: <4CB634E8.9050404@sns.eu>
References: <1286981982.2305.4.camel@Zaphod>  <4CB634E8.9050404@sns.eu>
Message-ID: <1287057867.3836.2.camel@Zaphod>

Hi Jan, yep setting ptime-override-value to 20 cured the problem,
thanks.

regards, John

On Thu, 2010-10-14 at 00:38 +0200, Jan Riedinger wrote:

> Hi John,
> 
> a ptime of 2 seems to be much too less and is a strange value, the
> default value is 20. Maybe you should try to force its usage. I'm not
> sure, how this can be done, maybe by using 
> <param name="ptime-override-value" value="20"/> or by specifying
> G729 at 20i for the codec.
> 
> BR
>    Jan
> 
> Am 13.10.2010 16:59, schrieb John Carpenter: 
> 
> > Hi, I am trying to bridge an incoming h323 call to an external SIP
> > provider, I am using latest git release. If I use G729 codec I get
> > message
> > 
> > "Unsupported ptime of 2 on  write Audio codec G.729{sw}  for
> > connection [0xb4c0ee10]"
> > 
> > and call fails to connect. If I use ulaw codec call fails to connect
> > with
> > 
> > "Write PDU fail: no control channel"
> > 
> > If I make a straight SIP to SIP call though same provider all work
> > ok. I have posted log of call in
> > http://pastebin.freeswitch.org/14216 because it is rather large.
> > This is my first venture into using the mod_h323 module and maybe I
> > am doing something stupid but have read all docs and seem to have
> > hit a brick wall on this.
> > 
> > my h323.conf.xml file looks like this
> > 
> > <configuration name="h323.conf" description="H323 Endpoints">
> > <settings>
> >    <param name="trace-level" value="10"/>
> >    <param name="context" value="public"/>
> >    <param name="dialplan" value="XML"/>
> >    <param name="codec-prefs" value="PCMA,PCMU,GSM,G729"/>
> >    <param name="use-rtp-timer" value="true"/> <!-- enable RTP timer
> > - should always be enabled -->
> >    <param name="rtp-timer-name" value="soft"/> <!-- Timer name, soft
> > is default -->
> >    <!-- <param name="ptime-override-value" value="20"/> --> <!--
> > Override negotiated ptime value with this value -->
> >    <param name="gk-address" value=""/>    <!-- empty to disable, "*"
> > to search LAN -->
> >    <param name="gk-identifer" value=""/>  <!-- optional name of gk
> > -->
> >    <param name="gk-interface" value=""/>  <!-- mandatory listener
> > interface name -->
> >    <param name="gk-retry" value="30"/>  <!-- optional GK register
> > retry timer -->
> >    <param name="faststart" value="true"/>  <!-- optional -->
> >    <param name="h245tunneling" value="true"/>  <!-- optional -->
> >    <param name="h245insetup" value="true"/>  <!-- optional -->
> >    <param name="jitter-size" value="60"/>  <!-- optional -->
> >    <param name="progress-indication" value="8"/> <!-- optional - PI
> > value in progress message-->
> >    <param name="alerting-indication" value="8"/> <!-- optional - PI
> > value in alerting message-->
> >    <param name="endpoint-name" value="fs"/>
> >    <param name="fax-old-asn" value="true"/>
> > </settings>
> > <listeners>
> >    <listener name="default">
> >      <param name="h323-ip" value="$${local_ip_v4}"/>
> >      <param name="h323-port" value="1720"/>
> >    </listener>
> > </listeners>
> > </configuration>
> > 
> > And this is the log of the xml_curl dialplan that is executed
> > 
> > <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> > <document type="freeswitch/xml">
> > <section name="dialplan" description="php dialplan">
> > <context name="public">
> > <extension name="normal">
> > <condition field="destination_number" expression="^(\d+)$">
> > <action application="set" data="bypass_media=false"/>
> > <action application="set" data="proxy_media=true"/>
> > <action application="bridge" data="sofia/gateway/arbinet-o/$1"/>
> > </condition>
> > </extension>
> > </context>
> > </section>
> > </document>
> > 
> > Any help will be greatly appreciated 
> > 
> > regards, John Carpenter 
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> 
> -- 
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                         E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
> 
> AG Charlottenburg - HRB 71973
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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From vedran.zeljeznak at gmail.com  Thu Oct 14 02:28:00 2010
From: vedran.zeljeznak at gmail.com (Vedran Zeljeznak)
Date: Thu, 14 Oct 2010 11:28:00 +0200
Subject: [Freeswitch-users] Java_ESL_Client - unhandled exception caught
Message-ID: <AANLkTik4J3bbgSGjF-pkUtx6Jxcewxk+P_5sf2RowEUS@mail.gmail.com>

hello,

i've found a bug in Java_ESL_Client and the info on the website
(http://wiki.freeswitch.org/wiki/Java_ESL_Client#Issues) said it
should be reported here so i'm sending it.

It looks like sometimes FS sends malformed URL encoded values through
Event Socket, probably because value is not properly terminated
(probably on my version of FS only - FreeSWITCH Version 1.0.6
(hacked-20100420T130612Z) - official version).

exception trace follows:

#### TRACE / LOG info
14.10.2010. 11:16:29 org.jboss.netty.channel.SimpleChannelUpstreamHandler
WARNING: EXCEPTION, please implement
org.freeswitch.esl.client.inbound.InboundClientHandler.exceptionCaught()
for proper handling.
java.lang.IllegalArgumentException: URLDecoder: Illegal hex characters
in escape (%) pattern - For input string: "\5"
	at java.net.URLDecoder.decode(URLDecoder.java:173)
	at org.freeswitch.esl.client.transport.event.EslEvent.parsePlainBody(EslEvent.java:187)
	at org.freeswitch.esl.client.transport.event.EslEvent.<init>(EslEvent.java:73)
	at org.freeswitch.esl.client.transport.event.EslEvent.<init>(EslEvent.java:62)
	at org.freeswitch.esl.client.internal.AbstractEslClientHandler.messageReceived(AbstractEslClientHandler.java:80)
	at org.freeswitch.esl.client.internal.debug.ChannelEventRunnable.run(ChannelEventRunnable.java:76)
	at org.jboss.netty.handler.execution.MemoryAwareThreadPoolExecutor$MemoryAwareRunnable.run(MemoryAwareThreadPoolExecutor.java:541)
	at java.util.concurrent.ThreadPoolExecutor$Worker.runTask(ThreadPoolExecutor.java:886)
	at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:908)
	at java.lang.Thread.run(Thread.java:619)
#### TRACE / LOG info


---
Vedran Zeljeznak


From curriegrad2004 at gmail.com  Thu Oct 14 11:18:31 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Thu, 14 Oct 2010 11:18:31 -0700
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>
	<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>
	<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>
	<SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>
	<AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>
	<SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>
	<C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>
	<SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>
Message-ID: <AANLkTimsBQtco2B-s6uXbkrQ7WoiQ0SzF8N5=wSskOML@mail.gmail.com>

Dialplan's regex doesn't look right to me. You'll need to fix that
before you do anything else.

I'd suggest this regex instead of the one you provided:
^[0-4]0(10[1-9][0-9])$

instead of this one
^([0-4]0[01][0-9])$

If you can't get the grasp of regular expressions, I'd highly suggest
you to pick up a book and read up on it. Knowing regular expressions
is a great skill to have ;)

2010/10/14 ? Chiang ?? Chih-Chung   Wybie <tear152 at hotmail.com>:
> I tried use 1013 at 38 calls to 1019 at 37
>
> show log below
>
> 2010-10-14 17:50:41.249876 [NOTICE] switch_channel.c:675 New Channel
> sofia/internal/1013 at 60.248.175.38 [875fdeef-b94e-41fb-a621-ea005bbaedbd]
> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
> 1013->1019 in context public
> 2010-10-14 17:50:41.263809 [NOTICE] switch_ivr.c:1450 Transfer
> sofia/internal/1013 at 60.248.175.38 to XML[1019 at default]
> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
> 1013->1019 in context default
> 2010-10-14 17:50:41.283904 [INFO] switch_core_session.c:1764 Sending early
> media
> 2010-10-14 17:50:41.285023 [NOTICE] mod_sofia.c:1994 Pre-Answer
> sofia/internal/1013 at 60.248.175.38!
>
> seems not work
>
> Thank you for you attension
> Best Regards
>
> Gary
>
>
>
> ________________________________
> From: david.ponzone at ipeva.fr
> To: freeswitch-users at lists.freeswitch.org
> Date: Thu, 14 Oct 2010 10:16:28 +0200
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> I think we would need a complete log from the target FS, in order to provide
> you with any efficient help.
> Thank
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 14/10/2010 ? 09:55, ? Chiang ?? Chih-Chung Wybie a ?crit :
>
> I tried Connect Two FreeSWITCH
> Boxes??http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
> and my configuration :
>
> 60.248.175.37
>
> (autoload_configs/act.conf.xml)
> <list name="domains" default="deny">
> ????? <node type="allow" cidr="60.248.175.38/28"/>
> ???? <!-- <node type="allow" domain="$${domain}"/>-->
> </list>
>
> (dialplan/default.xml)
>
> <extension name="Dial to 38">
> ????? <condition field="destination_number"
> expression="^([0-4]0[01][0-9])$">
> ??????? <action application="bridge"
> data="sofia/internal/$1 at 60.248.175.38"/>
> ????? </condition>
> ?</extension>
>
> (dialplan/public.xml)
>
> <extension name="Calls from 38">
> ????? <condition field="destination_number"
> expression="^([0-4]0[01][0-9])$">
> ??????? <action application="transfer" data="$1 XML default"/>
> ????? </condition>
> ?</extension>
>
>
>
> 60.248.175.38
>
> (autoload_configs/act.conf.xml)
> <list name="domains" default="deny">
> ????? <node type="allow" cidr="60.248.175.37/28"/>
> ???? <!-- <node type="allow" domain="$${domain}"/>-->
> </list>
>
> (dialplan/default.xml)
>
> <extension name="Dial to 37">
> ????? <condition field="destination_number"
> expression="^([0-4]0[01][0-9])$">
> ??????? <action application="bridge"
> data="sofia/internal/$1 at 60.248.175.37"/>
> ????? </condition>
> ?</extension>
>
> (dialplan/public.xml)
>
> <extension name="Calls from 37">
> ????? <condition field="destination_number"
> expression="^([0-4]0[01][0-9])$">
> ??????? <action application="transfer" data="$1 XML default"/>
> ????? </condition>
> ?</extension>
>
> when I use?1007 at 38?calls to?1007 at 37
> it won't show errors but still not work
>
> how should I change something ?
>
> Thank you for your attension
>
> Best Regards
> Gary
>
>
>
>
>
>> Date: Wed, 13 Oct 2010 22:17:13 -0700
>> From:?curriegrad2004 at gmail.com
>> To:?freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>>
>> You can create a SIP trunk between the two servers so calls can be
>> routed back and forth from the servers you've specified.
>>
>> 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
>> > I tried it
>> > but not working
>> > and show errors below
>> > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493 Cannot
>> > create
>> > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
>> > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493 Cannot
>> > create
>> > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
>> > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate Failed.
>> > Cause: USER_NOT_REGISTERED
>> >
>> > and for example
>> > ?I use SIP server 60.248.175.38? user id=1007
>> > and I call to?1007 at 60.248.175.37??? that will show these errors
>> >
>> > how should set more?
>> >
>> > thank you for your attension
>> >
>> > Best Regards
>> > Gary
>> >
>> > ________________________________
>> > Date: Thu, 14 Oct 2010 10:55:49 +0800
>> > From:?xyangni at gmail.com
>> > To:?freeswitch-users at lists.freeswitch.org
>> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>> >
>> > I think you can reg at server A and than call B as a normal SIP call to
>> > external domain.
>> >
>> > 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
>> >
>> >
>> >
>> > Hello
>> >
>> > I set two SIP servers ?60.248.175.37 & 60.248.175.38 and use X-Lite to
>> > call
>> >
>> > my question is how can I call between these servers?
>> >
>> > if I can , how shoould I set in *.xml? or X-Lite?
>> >
>> > thank you for your attension
>> >
>> >
>> > Best Regards
>> > Gary
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> >?FreeSWITCH-users at lists.freeswitch.org
>> >?http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >?http://www.freeswitch.org
>> >
>> >
>> >
>> > _______________________________________________ FreeSWITCH-users mailing
>> > list?FreeSWITCH-users at lists.freeswitch.org
>> >?http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >?http://www.freeswitch.org
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> >?FreeSWITCH-users at lists.freeswitch.org
>> >?http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >?http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>?FreeSWITCH-users at lists.freeswitch.org
>>?http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>?http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Thu Oct 14 11:27:31 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Oct 2010 11:27:31 -0700
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <AANLkTimsBQtco2B-s6uXbkrQ7WoiQ0SzF8N5=wSskOML@mail.gmail.com>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>
	<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>
	<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>
	<SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>
	<AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>
	<SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>
	<C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>
	<SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>
	<AANLkTimsBQtco2B-s6uXbkrQ7WoiQ0SzF8N5=wSskOML@mail.gmail.com>
Message-ID: <AANLkTikTSTtaN1cd14U+G8yKA1u=A3dCgtKgSrEkYDb-@mail.gmail.com>

FYI, we have several resources to help
http://wiki.freeswitch.org/wiki/Regular_Expression
http://bit.ly/aijtAC

:)

-MC

On Thu, Oct 14, 2010 at 11:18 AM, Jeffrey Leung <curriegrad2004 at gmail.com>wrote:

> Dialplan's regex doesn't look right to me. You'll need to fix that
> before you do anything else.
>
> I'd suggest this regex instead of the one you provided:
> ^[0-4]0(10[1-9][0-9])$
>
> instead of this one
> ^([0-4]0[01][0-9])$
>
> If you can't get the grasp of regular expressions, I'd highly suggest
> you to pick up a book and read up on it. Knowing regular expressions
> is a great skill to have ;)
>
> 2010/10/14 ? Chiang ?? Chih-Chung   Wybie <tear152 at hotmail.com>:
> > I tried use 1013 at 38 calls to 1019 at 37
> >
> > show log below
> >
> > 2010-10-14 17:50:41.249876 [NOTICE] switch_channel.c:675 New Channel
> > sofia/internal/1013 at 60.248.175.38 [875fdeef-b94e-41fb-a621-ea005bbaedbd]
> > 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
> > 1013->1019 in context public
> > 2010-10-14 17:50:41.263809 [NOTICE] switch_ivr.c:1450 Transfer
> > sofia/internal/1013 at 60.248.175.38 to XML[1019 at default]
> > 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
> > 1013->1019 in context default
> > 2010-10-14 17:50:41.283904 [INFO] switch_core_session.c:1764 Sending
> early
> > media
> > 2010-10-14 17:50:41.285023 [NOTICE] mod_sofia.c:1994 Pre-Answer
> > sofia/internal/1013 at 60.248.175.38!
> >
> > seems not work
> >
> > Thank you for you attension
> > Best Regards
> >
> > Gary
> >
> >
> >
> > ________________________________
> > From: david.ponzone at ipeva.fr
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Thu, 14 Oct 2010 10:16:28 +0200
> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >
> > I think we would need a complete log from the target FS, in order to
> provide
> > you with any efficient help.
> > Thank
> > David Ponzone  Direction Technique
> > email: david.ponzone at ipeva.fr
> > tel:      01 74 03 18 97
> > gsm:   06 66 98 76 34
> > Service Client IPeva
> > tel:      0811 46 26 26
> > www.ipeva.fr  -   www.ipeva-studio.com
> > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> > l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion
> > non autoris?e est interdite. Tout message ?lectronique est susceptible
> > d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il
> > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> >
> >
> >
> > Le 14/10/2010 ? 09:55, ? Chiang ?? Chih-Chung Wybie a ?crit :
> >
> > I tried Connect Two FreeSWITCH
> > Boxes  http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
> > and my configuration :
> >
> > 60.248.175.37
> >
> > (autoload_configs/act.conf.xml)
> > <list name="domains" default="deny">
> >       <node type="allow" cidr="60.248.175.38/28"/>
> >      <!-- <node type="allow" domain="$${domain}"/>-->
> > </list>
> >
> > (dialplan/default.xml)
> >
> > <extension name="Dial to 38">
> >       <condition field="destination_number"
> > expression="^([0-4]0[01][0-9])$">
> >         <action application="bridge"
> > data="sofia/internal/$1 at 60.248.175.38"/>
> >       </condition>
> >  </extension>
> >
> > (dialplan/public.xml)
> >
> > <extension name="Calls from 38">
> >       <condition field="destination_number"
> > expression="^([0-4]0[01][0-9])$">
> >         <action application="transfer" data="$1 XML default"/>
> >       </condition>
> >  </extension>
> >
> >
> >
> > 60.248.175.38
> >
> > (autoload_configs/act.conf.xml)
> > <list name="domains" default="deny">
> >       <node type="allow" cidr="60.248.175.37/28"/>
> >      <!-- <node type="allow" domain="$${domain}"/>-->
> > </list>
> >
> > (dialplan/default.xml)
> >
> > <extension name="Dial to 37">
> >       <condition field="destination_number"
> > expression="^([0-4]0[01][0-9])$">
> >         <action application="bridge"
> > data="sofia/internal/$1 at 60.248.175.37"/>
> >       </condition>
> >  </extension>
> >
> > (dialplan/public.xml)
> >
> > <extension name="Calls from 37">
> >       <condition field="destination_number"
> > expression="^([0-4]0[01][0-9])$">
> >         <action application="transfer" data="$1 XML default"/>
> >       </condition>
> >  </extension>
> >
> > when I use 1007 at 38 calls to 1007 at 37
> > it won't show errors but still not work
> >
> > how should I change something ?
> >
> > Thank you for your attension
> >
> > Best Regards
> > Gary
> >
> >
> >
> >
> >
> >> Date: Wed, 13 Oct 2010 22:17:13 -0700
> >> From: curriegrad2004 at gmail.com
> >> To: freeswitch-users at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >>
> >> You can create a SIP trunk between the two servers so calls can be
> >> routed back and forth from the servers you've specified.
> >>
> >> 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
> >> > I tried it
> >> > but not working
> >> > and show errors below
> >> > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493 Cannot
> >> > create
> >> > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> >> > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493 Cannot
> >> > create
> >> > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> >> > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate Failed.
> >> > Cause: USER_NOT_REGISTERED
> >> >
> >> > and for example
> >> >  I use SIP server 60.248.175.38  user id=1007
> >> > and I call to 1007 at 60.248.175.37    that will show these errors
> >> >
> >> > how should set more?
> >> >
> >> > thank you for your attension
> >> >
> >> > Best Regards
> >> > Gary
> >> >
> >> > ________________________________
> >> > Date: Thu, 14 Oct 2010 10:55:49 +0800
> >> > From: xyangni at gmail.com
> >> > To: freeswitch-users at lists.freeswitch.org
> >> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >> >
> >> > I think you can reg at server A and than call B as a normal SIP call
> to
> >> > external domain.
> >> >
> >> > 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
> >> >
> >> >
> >> >
> >> > Hello
> >> >
> >> > I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X-Lite to
> >> > call
> >> >
> >> > my question is how can I call between these servers?
> >> >
> >> > if I can , how shoould I set in *.xml? or X-Lite?
> >> >
> >> > thank you for your attension
> >> >
> >> >
> >> > Best Regards
> >> > Gary
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >> >
> >> > _______________________________________________ FreeSWITCH-users
> mailing
> >> > list FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________ FreeSWITCH-users mailing
> > list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Thu Oct 14 11:34:16 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 14 Oct 2010 20:34:16 +0200
Subject: [Freeswitch-users] Java_ESL_Client - unhandled exception caught
In-Reply-To: <AANLkTik4J3bbgSGjF-pkUtx6Jxcewxk+P_5sf2RowEUS@mail.gmail.com>
References: <AANLkTik4J3bbgSGjF-pkUtx6Jxcewxk+P_5sf2RowEUS@mail.gmail.com>
Message-ID: <AANLkTimbCv6bwRmSqFq+88jt568-pOVr3HtqobMsf093@mail.gmail.com>

On Thu, Oct 14, 2010 at 11:28 AM, Vedran Zeljeznak <
vedran.zeljeznak at gmail.com> wrote:

> hello,
>
> i've found a bug in Java_ESL_Client and the info on the website
> (http://wiki.freeswitch.org/wiki/Java_ESL_Client#Issues) said it
> should be reported here so i'm sending it.
>
> It looks like sometimes FS sends malformed URL encoded values through
> Event Socket, probably because value is not properly terminated
> (probably on my version of FS only - FreeSWITCH Version 1.0.6
> (hacked-20100420T130612Z) - official version).
>

ya, quite old... can you try to reproduce it on the latest git ?
im not having this issues at all.
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From Holger.Esser at Convergys.com  Thu Oct 14 12:39:19 2010
From: Holger.Esser at Convergys.com (Esser, Holger)
Date: Thu, 14 Oct 2010 14:39:19 -0500
Subject: [Freeswitch-users] DTMF off time
In-Reply-To: <AANLkTimbCv6bwRmSqFq+88jt568-pOVr3HtqobMsf093@mail.gmail.com>
References: <AANLkTik4J3bbgSGjF-pkUtx6Jxcewxk+P_5sf2RowEUS@mail.gmail.com>
	<AANLkTimbCv6bwRmSqFq+88jt568-pOVr3HtqobMsf093@mail.gmail.com>
Message-ID: <0FF607C13B7F7A42B5B80DA9EB204C9109D44BDCBE@srv-ex01-dal.intervoice.int>

Hi,

Is there a way to configure the DTMF duration with gentones. Specifically is there a way to have the off time to be set as the on time per DTMF.

Thanks,
Holger

________________________________
This e-mail transmission may contain information that is proprietary, privileged and/or confidential and is intended exclusively for the person(s) to whom it is addressed. Any use, copying, retention or disclosure by any person other than the intended recipient or the intended recipient's designees is strictly prohibited. If you are the intended recipient, you must treat the information in confidence and in accordance with all laws related to the privacy and confidentiality of such information. If you are not the intended recipient or their designee, please notify the sender immediately by return e-mail and delete all copies of this email, including all attachments.

Intervoice, Inc. 17811 Waterview Parkway Dallas, TX 75252 USA

Intervoice Limited, Registered in England and Wales with number 2601740, 50 Park Road, Gatley, Cheshire, SK8 4HZ. VAT Number: 560421375

Intervoice GmbH, Hagenauer Stra?e 55, 65203 Wiesbaden, Sitz der Gesellschaft: Wiesbaden, Handelsregister: HRB 8186 (Amtsgericht Wiesbaden), Gesch?ftsf?hrer: Wayne Barclay, Steffen Selbmann
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From anthony.minessale at gmail.com  Thu Oct 14 13:36:41 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Oct 2010 15:36:41 -0500
Subject: [Freeswitch-users] DTMF off time
In-Reply-To: <0FF607C13B7F7A42B5B80DA9EB204C9109D44BDCBE@srv-ex01-dal.intervoice.int>
References: <AANLkTik4J3bbgSGjF-pkUtx6Jxcewxk+P_5sf2RowEUS@mail.gmail.com>
	<AANLkTimbCv6bwRmSqFq+88jt568-pOVr3HtqobMsf093@mail.gmail.com>
	<0FF607C13B7F7A42B5B80DA9EB204C9109D44BDCBE@srv-ex01-dal.intervoice.int>
Message-ID: <AANLkTimOz0tdhaWVAPwgXWKcJLzJ_m14BaKKg9qUya0a@mail.gmail.com>

d=250;w=250;1234

that would use on time of 250 and off time of 250 and play 1234


On Thu, Oct 14, 2010 at 2:39 PM, Esser, Holger
<Holger.Esser at convergys.com> wrote:
> Hi,
>
>
>
> Is there a way to configure the DTMF duration with gentones. Specifically is
> there a way to have the off time to be set as the on time per DTMF.
>
>
>
> Thanks,
>
> Holger
>
> ________________________________
> This e-mail transmission may contain information that is proprietary,
> privileged and/or confidential and is intended exclusively for the person(s)
> to whom it is addressed. Any use, copying, retention or disclosure by any
> person other than the intended recipient or the intended recipient's
> designees is strictly prohibited. If you are the intended recipient, you
> must treat the information in confidence and in accordance with all laws
> related to the privacy and confidentiality of such information. If you are
> not the intended recipient or their designee, please notify the sender
> immediately by return e-mail and delete all copies of this email, including
> all attachments.
>
> Intervoice, Inc. 17811 Waterview Parkway Dallas, TX 75252 USA
>
> Intervoice Limited, Registered in England and Wales with number 2601740, 50
> Park Road, Gatley, Cheshire, SK8 4HZ. VAT Number: 560421375
>
> Intervoice GmbH, Hagenauer Stra?e 55, 65203 Wiesbaden, Sitz der
> Gesellschaft: Wiesbaden, Handelsregister: HRB 8186 (Amtsgericht Wiesbaden),
> Gesch?ftsf?hrer: Wayne Barclay, Steffen Selbmann
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From Holger.Esser at Convergys.com  Thu Oct 14 13:45:09 2010
From: Holger.Esser at Convergys.com (Esser, Holger)
Date: Thu, 14 Oct 2010 15:45:09 -0500
Subject: [Freeswitch-users] DTMF off time
In-Reply-To: <AANLkTimOz0tdhaWVAPwgXWKcJLzJ_m14BaKKg9qUya0a@mail.gmail.com>
References: <AANLkTik4J3bbgSGjF-pkUtx6Jxcewxk+P_5sf2RowEUS@mail.gmail.com>
	<AANLkTimbCv6bwRmSqFq+88jt568-pOVr3HtqobMsf093@mail.gmail.com>
	<0FF607C13B7F7A42B5B80DA9EB204C9109D44BDCBE@srv-ex01-dal.intervoice.int>
	<AANLkTimOz0tdhaWVAPwgXWKcJLzJ_m14BaKKg9qUya0a@mail.gmail.com>
Message-ID: <0FF607C13B7F7A42B5B80DA9EB204C9109D44BDCD8@srv-ex01-dal.intervoice.int>

Cool, thank you very much.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Thursday, October 14, 2010 3:37 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] DTMF off time

d=250;w=250;1234

that would use on time of 250 and off time of 250 and play 1234


On Thu, Oct 14, 2010 at 2:39 PM, Esser, Holger
<Holger.Esser at convergys.com> wrote:
> Hi,
>
>
>
> Is there a way to configure the DTMF duration with gentones. Specifically is
> there a way to have the off time to be set as the on time per DTMF.
>
>
>
> Thanks,
>
> Holger
>
> ________________________________
> This e-mail transmission may contain information that is proprietary,
> privileged and/or confidential and is intended exclusively for the person(s)
> to whom it is addressed. Any use, copying, retention or disclosure by any
> person other than the intended recipient or the intended recipient's
> designees is strictly prohibited. If you are the intended recipient, you
> must treat the information in confidence and in accordance with all laws
> related to the privacy and confidentiality of such information. If you are
> not the intended recipient or their designee, please notify the sender
> immediately by return e-mail and delete all copies of this email, including
> all attachments.
>
> Intervoice, Inc. 17811 Waterview Parkway Dallas, TX 75252 USA
>
> Intervoice Limited, Registered in England and Wales with number 2601740, 50
> Park Road, Gatley, Cheshire, SK8 4HZ. VAT Number: 560421375
>
> Intervoice GmbH, Hagenauer Stra?e 55, 65203 Wiesbaden, Sitz der
> Gesellschaft: Wiesbaden, Handelsregister: HRB 8186 (Amtsgericht Wiesbaden),
> Gesch?ftsf?hrer: Wayne Barclay, Steffen Selbmann
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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

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

No virus found in this incoming message.
Checked by AVG - www.avg.com 
Version: 9.0.862 / Virus Database: 271.1.1/3192 - Release Date: 10/14/10 01:34:00

No virus found in this outgoing message.
Checked by AVG - www.avg.com 
Version: 9.0.862 / Virus Database: 271.1.1/3192 - Release Date: 10/14/10 01:34:00


From christian at yellox.de  Thu Oct 14 13:13:05 2010
From: christian at yellox.de (Christian Hiller)
Date: Thu, 14 Oct 2010 22:13:05 +0200 (CEST)
Subject: [Freeswitch-users] "ab" logging in cdr_csv
Message-ID: <13133318.24411287087185484.YELLOX.WebMail.wwwrun@mx.yellox.de>

Hello,

A-leg and B-leg have the same channel variables. When enabling "ab"
logging in cdr_csv, how can i distinguish the variables of both legs ?

Regards

Chris Powered by yellox.com



From Kyle.Haefner at colostate.edu  Thu Oct 14 13:19:01 2010
From: Kyle.Haefner at colostate.edu (Kyle Haefner)
Date: Thu, 14 Oct 2010 14:19:01 -0600
Subject: [Freeswitch-users] Configure anf libjpeg
Message-ID: <AANLkTimnUg+fY3nDqW8u0Y3yTjB++vL1F+Z4+5dxPpRr@mail.gmail.com>

Hi all,

I've run into this a couple of times now, bootstrap and or configure need to
test for libjpeg-devel, as mod_spandsp needs it.

Creating mod_spandsp.la
/usr/bin/ld: cannot find -ljpeg
collect2: ld returned 1 exit status


Kyle
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From djbinter at gmail.com  Thu Oct 14 15:08:59 2010
From: djbinter at gmail.com (DJB International)
Date: Thu, 14 Oct 2010 15:08:59 -0700
Subject: [Freeswitch-users] Configure anf libjpeg
In-Reply-To: <AANLkTimnUg+fY3nDqW8u0Y3yTjB++vL1F+Z4+5dxPpRr@mail.gmail.com>
References: <AANLkTimnUg+fY3nDqW8u0Y3yTjB++vL1F+Z4+5dxPpRr@mail.gmail.com>
Message-ID: <AANLkTikv19tSiZ0RUnjpc7nLjEA=SF81ffbt+bqL0dEr@mail.gmail.com>

yum install libtiff-devel libjpeg-devel


On Thu, Oct 14, 2010 at 1:19 PM, Kyle Haefner <Kyle.Haefner at colostate.edu>wrote:

> Hi all,
>
> I've run into this a couple of times now, bootstrap and or configure need
> to test for libjpeg-devel, as mod_spandsp needs it.
>
> Creating mod_spandsp.la
> /usr/bin/ld: cannot find -ljpeg
> collect2: ld returned 1 exit status
>
>
> Kyle
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Thu Oct 14 15:18:28 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 14 Oct 2010 17:18:28 -0500
Subject: [Freeswitch-users] Configure anf libjpeg
In-Reply-To: <AANLkTimnUg+fY3nDqW8u0Y3yTjB++vL1F+Z4+5dxPpRr@mail.gmail.com>
References: <AANLkTimnUg+fY3nDqW8u0Y3yTjB++vL1F+Z4+5dxPpRr@mail.gmail.com>
Message-ID: <C6005E92-4AAE-4E36-BFE0-BCDBBE41B418@freeswitch.org>

Patches welcome!

/b

On Oct 14, 2010, at 3:19 PM, Kyle Haefner wrote:

> I've run into this a couple of times now, bootstrap and or configure need to test for libjpeg-devel, as mod_spandsp needs it.



From david.ponzone at ipeva.fr  Thu Oct 14 15:57:41 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 15 Oct 2010 00:57:41 +0200
Subject: [Freeswitch-users] "ab" logging in cdr_csv
In-Reply-To: <13133318.24411287087185484.YELLOX.WebMail.wwwrun@mx.yellox.de>
References: <13133318.24411287087185484.YELLOX.WebMail.wwwrun@mx.yellox.de>
Message-ID: <72EA4B6C-C5C7-4D2E-9B71-BAD05E219EF4@ipeva.fr>

Christian,

I am not sure I get your question perfectly.

But when you enable "ab" loggin in CSV, you get one line per leg.
If you use the default CSV format, it's not that obvious to  
distinguish both lines.
The A-leg line will carry a value in ${uuid} and ${bleg_uuid} (this  
allows to know which B-leg was bridged to that A-leg).
The B-leg line will carry a value in ${uuid} only.

This mean if you want to locate the 2 lines corresponding to the same  
call, you have to find the lines with:
${bleg_uuid} of line 1 = ${uuid} of line 2

I hope this will clear things a bit for you.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 14/10/2010 ? 22:13, Christian Hiller a ?crit :

> Hello,
>
> A-leg and B-leg have the same channel variables. When enabling "ab"
> logging in cdr_csv, how can i distinguish the variables of both legs ?
>
> Regards
>
> Chris Powered by yellox.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.varnes at gmail.com  Thu Oct 14 16:17:32 2010
From: david.varnes at gmail.com (david varnes)
Date: Fri, 15 Oct 2010 10:17:32 +1100
Subject: [Freeswitch-users] Java_ESL_Client - unhandled exception caught
In-Reply-To: <AANLkTik4J3bbgSGjF-pkUtx6Jxcewxk+P_5sf2RowEUS@mail.gmail.com>
References: <AANLkTik4J3bbgSGjF-pkUtx6Jxcewxk+P_5sf2RowEUS@mail.gmail.com>
Message-ID: <AANLkTikgEBB5c4AtVZKkSFTjD62e8s_ws8tapmkebp+D@mail.gmail.com>

Vedran,
Thanks for reporting the issue.  As you can see from the stack trace
this is just a call to the regular java URL decoder.  Can you capture
the content of the event or message that is causing this to fail ?

Or at the least what the cause event is so that I can determine if
it is still a problem on trunk ?

thanks
davidv

On 14 October 2010 20:28, Vedran Zeljeznak <vedran.zeljeznak at gmail.com> wrote:
> hello,
>
> i've found a bug in Java_ESL_Client and the info on the website
> (http://wiki.freeswitch.org/wiki/Java_ESL_Client#Issues) said it
> should be reported here so i'm sending it.
>
> It looks like sometimes FS sends malformed URL encoded values through
> Event Socket, probably because value is not properly terminated
> (probably on my version of FS only - FreeSWITCH Version 1.0.6
> (hacked-20100420T130612Z) - official version).
>
> exception trace follows:
>
> #### TRACE / LOG info
> 14.10.2010. 11:16:29 org.jboss.netty.channel.SimpleChannelUpstreamHandler
> WARNING: EXCEPTION, please implement
> org.freeswitch.esl.client.inbound.InboundClientHandler.exceptionCaught()
> for proper handling.
> java.lang.IllegalArgumentException: URLDecoder: Illegal hex characters
> in escape (%) pattern - For input string: "\5"
> ? ? ? ?at java.net.URLDecoder.decode(URLDecoder.java:173)
> ? ? ? ?at org.freeswitch.esl.client.transport.event.EslEvent.parsePlainBody(EslEvent.java:187)
> ? ? ? ?at org.freeswitch.esl.client.transport.event.EslEvent.<init>(EslEvent.java:73)
> ? ? ? ?at org.freeswitch.esl.client.transport.event.EslEvent.<init>(EslEvent.java:62)
> ? ? ? ?at org.freeswitch.esl.client.internal.AbstractEslClientHandler.messageReceived(AbstractEslClientHandler.java:80)
> ? ? ? ?at org.freeswitch.esl.client.internal.debug.ChannelEventRunnable.run(ChannelEventRunnable.java:76)
> ? ? ? ?at org.jboss.netty.handler.execution.MemoryAwareThreadPoolExecutor$MemoryAwareRunnable.run(MemoryAwareThreadPoolExecutor.java:541)
> ? ? ? ?at java.util.concurrent.ThreadPoolExecutor$Worker.runTask(ThreadPoolExecutor.java:886)
> ? ? ? ?at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:908)
> ? ? ? ?at java.lang.Thread.run(Thread.java:619)
> #### TRACE / LOG info
>
>
> ---
> Vedran Zeljeznak
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
david varnes

e: david.varnes at gmail.com
p: +61 404 925 633


From riedinger at sns.eu  Thu Oct 14 16:30:10 2010
From: riedinger at sns.eu (Jan Riedinger)
Date: Fri, 15 Oct 2010 01:30:10 +0200
Subject: [Freeswitch-users] "ab" logging in cdr_csv
In-Reply-To: <13133318.24411287087185484.YELLOX.WebMail.wwwrun@mx.yellox.de>
References: <13133318.24411287087185484.YELLOX.WebMail.wwwrun@mx.yellox.de>
Message-ID: <4CB79282.8010908@sns.eu>

  If you use FS for standard routing you can set for the inbound channel:

<action application="set" data="call_leg=aleg"/>
<action application="export" data="nolocal:call_leg=bleg"/>

If you include "call_leg" in your cdr, you will know what leg it is.

BR
    Jan



Am 14.10.2010 22:13, schrieb Christian Hiller:
> Hello,
>
> A-leg and B-leg have the same channel variables. When enabling "ab"
> logging in cdr_csv, how can i distinguish the variables of both legs ?
>
> Regards
>
> Chris Powered by yellox.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                         E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973



From david.ponzone at ipeva.fr  Thu Oct 14 17:02:15 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 15 Oct 2010 02:02:15 +0200
Subject: [Freeswitch-users] "ab" logging in cdr_csv
In-Reply-To: <4CB79282.8010908@sns.eu>
References: <13133318.24411287087185484.YELLOX.WebMail.wwwrun@mx.yellox.de>
	<4CB79282.8010908@sns.eu>
Message-ID: <69A011AD-E2DC-4CED-A652-48B20598B4D8@ipeva.fr>

Jan,

Or you can use the wonderful ${direction} variable, which already does  
that, as inbound is a-leg and outbound is b-leg :)

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 15/10/2010 ? 01:30, Jan Riedinger a ?crit :

>  If you use FS for standard routing you can set for the inbound  
> channel:
>
> <action application="set" data="call_leg=aleg"/>
> <action application="export" data="nolocal:call_leg=bleg"/>
>
> If you include "call_leg" in your cdr, you will know what leg it is.
>
> BR
>    Jan
>
>
>
> Am 14.10.2010 22:13, schrieb Christian Hiller:
>> Hello,
>>
>> A-leg and B-leg have the same channel variables. When enabling "ab"
>> logging in cdr_csv, how can i distinguish the variables of both  
>> legs ?
>>
>> Regards
>>
>> Chris Powered by yellox.com
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> -- 
> Jan Riedinger                           Phone :  +49-30-39 73 19 66
> Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
>                                         E-Mail:  riedinger at sns.eu
> SNS Consult GmbH                        ICQ   :  163-237-041
> S?dwestkorso 49a                        MSN   :  jan at sns-consult.de
> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
>
> AG Charlottenburg - HRB 71973
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From b0ef at esben-stien.name  Thu Oct 14 18:05:05 2010
From: b0ef at esben-stien.name (Esben Stien)
Date: Fri, 15 Oct 2010 03:05:05 +0200
Subject: [Freeswitch-users] Freeswitch Video Capture and Playback
In-Reply-To: <87pr6yz5wa.fsf@quasar.esben-stien.name> (Esben Stien's message
	of "Tue\, 01 Dec 2009 23\:27\:49 +0100")
References: <87k4xlga1k.fsf@quasar.esben-stien.name>
	<87pr6yz5wa.fsf@quasar.esben-stien.name>
Message-ID: <87bp6we05a.fsf@quasar.esben-stien.name>

Esben Stien <b0ef at esben-stien.name> writes:

> So nobody is using video with freeswitch?. 

I've tried with Ekiga and Empathy.

Has anybody gotten Freeswitch to record some video with any free
client?. 

Is there any info on the Freeswitch Video Format, FSV?. Maybe how to
encode a video with gstreamer/mencoder?.

Does there exist a FSV test file?. If anyone got one of these FSV files,
please upload a sample.

Does something like video loopback exist, like a video echo
application for Freeswitch?.

-- 
Esben Stien is b0ef at e     s      a             
         http://www. s     t    n m
          irc://irc.  b  -  i  .   e/%23contact
           sip:b0ef@   e     e 
           jid:b0ef@    n     n


From msc at freeswitch.org  Thu Oct 14 17:31:12 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Oct 2010 17:31:12 -0700
Subject: [Freeswitch-users] Freeswitch Video Capture and Playback
In-Reply-To: <87bp6we05a.fsf@quasar.esben-stien.name>
References: <87k4xlga1k.fsf@quasar.esben-stien.name>
	<87pr6yz5wa.fsf@quasar.esben-stien.name>
	<87bp6we05a.fsf@quasar.esben-stien.name>
Message-ID: <AANLkTimo_+GehmBLV0JaamontZ-mDVomKtbUqXGHdk+q@mail.gmail.com>

Do you have a SIP trace? What was in the SDP's? Were the codecs all lined
up?
-MC

On Thu, Oct 14, 2010 at 6:05 PM, Esben Stien <b0ef at esben-stien.name> wrote:

> Esben Stien <b0ef at esben-stien.name> writes:
>
> > So nobody is using video with freeswitch?.
>
> I've tried with Ekiga and Empathy.
>
> Has anybody gotten Freeswitch to record some video with any free
> client?.
>
> Is there any info on the Freeswitch Video Format, FSV?. Maybe how to
> encode a video with gstreamer/mencoder?.
>
> Does there exist a FSV test file?. If anyone got one of these FSV files,
> please upload a sample.
>
> Does something like video loopback exist, like a video echo
> application for Freeswitch?.
>
> --
> Esben Stien is b0ef at e     s      a
>         http://www. s     t    n m
>          irc://irc.  b  -  i  .   e/%23contact
>           sip:b0ef@   e     e
>           jid:b0ef@    n     n
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tear152 at hotmail.com  Thu Oct 14 17:44:44 2010
From: tear152 at hotmail.com (=?gb2312?B?va0gQ2hpYW5nINbB1tAgQ2hpaC1DaHVuZyAgIFd5Ymll?=)
Date: Fri, 15 Oct 2010 08:44:44 +0800
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <AANLkTikTSTtaN1cd14U+G8yKA1u=A3dCgtKgSrEkYDb-@mail.gmail.com>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>,
	<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>,
	<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>,
	<SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>,
	<AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>,
	<SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>,
	<C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>,
	<SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>,
	<AANLkTimsBQtco2B-s6uXbkrQ7WoiQ0SzF8N5=wSskOML@mail.gmail.com>,
	<AANLkTikTSTtaN1cd14U+G8yKA1u=A3dCgtKgSrEkYDb-@mail.gmail.com>
Message-ID: <SNT130-w1462D55D67F6FED40D4D74FB570@phx.gbl>


Hello
 
but I can call to each other on the same server
 
for example 
 
I can use 1007 at 37  call 1013 at 37  ,  4001 at 37  , 2010 at 37 , 3009 at 37 
 
 
 Thank you for your attension
Best Regards
 Gary


 


Date: Thu, 14 Oct 2010 11:27:31 -0700
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite

FYI, we have several resources to help
http://wiki.freeswitch.org/wiki/Regular_Expression
http://bit.ly/aijtAC

:)

-MC


On Thu, Oct 14, 2010 at 11:18 AM, Jeffrey Leung <curriegrad2004 at gmail.com> wrote:

Dialplan's regex doesn't look right to me. You'll need to fix that
before you do anything else.

I'd suggest this regex instead of the one you provided:
^[0-4]0(10[1-9][0-9])$

instead of this one
^([0-4]0[01][0-9])$

If you can't get the grasp of regular expressions, I'd highly suggest
you to pick up a book and read up on it. Knowing regular expressions
is a great skill to have ;)

2010/10/14 ? Chiang ?? Chih-Chung   Wybie <tear152 at hotmail.com>:
> I tried use 1013 at 38 calls to 1019 at 37
>
> show log below
>
> 2010-10-14 17:50:41.249876 [NOTICE] switch_channel.c:675 New Channel
> sofia/internal/1013 at 60.248.175.38 [875fdeef-b94e-41fb-a621-ea005bbaedbd]
> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
> 1013->1019 in context public
> 2010-10-14 17:50:41.263809 [NOTICE] switch_ivr.c:1450 Transfer
> sofia/internal/1013 at 60.248.175.38 to XML[1019 at default]
> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
> 1013->1019 in context default
> 2010-10-14 17:50:41.283904 [INFO] switch_core_session.c:1764 Sending early
> media
> 2010-10-14 17:50:41.285023 [NOTICE] mod_sofia.c:1994 Pre-Answer
> sofia/internal/1013 at 60.248.175.38!
>
> seems not work
>
> Thank you for you attension
> Best Regards
>
> Gary
>
>
>
> ________________________________
> From: david.ponzone at ipeva.fr
> To: freeswitch-users at lists.freeswitch.org
> Date: Thu, 14 Oct 2010 10:16:28 +0200
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> I think we would need a complete log from the target FS, in order to provide
> you with any efficient help.
> Thank
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 14/10/2010 ? 09:55, ? Chiang ?? Chih-Chung Wybie a ?crit :
>
> I tried Connect Two FreeSWITCH
> Boxes  http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
> and my configuration :
>
> 60.248.175.37
>
> (autoload_configs/act.conf.xml)
> <list name="domains" default="deny">
>       <node type="allow" cidr="60.248.175.38/28"/>
>      <!-- <node type="allow" domain="$${domain}"/>-->
> </list>
>
> (dialplan/default.xml)
>
> <extension name="Dial to 38">
>       <condition field="destination_number"
> expression="^([0-4]0[01][0-9])$">
>         <action application="bridge"
> data="sofia/internal/$1 at 60.248.175.38"/>
>       </condition>
>  </extension>
>
> (dialplan/public.xml)
>
> <extension name="Calls from 38">
>       <condition field="destination_number"
> expression="^([0-4]0[01][0-9])$">
>         <action application="transfer" data="$1 XML default"/>
>       </condition>
>  </extension>
>
>
>
> 60.248.175.38
>
> (autoload_configs/act.conf.xml)
> <list name="domains" default="deny">
>       <node type="allow" cidr="60.248.175.37/28"/>
>      <!-- <node type="allow" domain="$${domain}"/>-->
> </list>
>
> (dialplan/default.xml)
>
> <extension name="Dial to 37">
>       <condition field="destination_number"
> expression="^([0-4]0[01][0-9])$">
>         <action application="bridge"
> data="sofia/internal/$1 at 60.248.175.37"/>
>       </condition>
>  </extension>
>
> (dialplan/public.xml)
>
> <extension name="Calls from 37">
>       <condition field="destination_number"
> expression="^([0-4]0[01][0-9])$">
>         <action application="transfer" data="$1 XML default"/>
>       </condition>
>  </extension>
>
> when I use 1007 at 38 calls to 1007 at 37
> it won't show errors but still not work
>
> how should I change something ?
>
> Thank you for your attension
>
> Best Regards
> Gary
>
>
>
>
>
>> Date: Wed, 13 Oct 2010 22:17:13 -0700
>> From: curriegrad2004 at gmail.com
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>>
>> You can create a SIP trunk between the two servers so calls can be
>> routed back and forth from the servers you've specified.
>>
>> 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
>> > I tried it
>> > but not working
>> > and show errors below
>> > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493 Cannot
>> > create
>> > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
>> > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493 Cannot
>> > create
>> > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
>> > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate Failed.
>> > Cause: USER_NOT_REGISTERED
>> >
>> > and for example
>> >  I use SIP server 60.248.175.38  user id=1007
>> > and I call to 1007 at 60.248.175.37    that will show these errors
>> >
>> > how should set more?
>> >
>> > thank you for your attension
>> >
>> > Best Regards
>> > Gary
>> >
>> > ________________________________
>> > Date: Thu, 14 Oct 2010 10:55:49 +0800
>> > From: xyangni at gmail.com
>> > To: freeswitch-users at lists.freeswitch.org
>> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>> >
>> > I think you can reg at server A and than call B as a normal SIP call to
>> > external domain.
>> >
>> > 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
>> >
>> >
>> >
>> > Hello
>> >
>> > I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X-Lite to
>> > call
>> >
>> > my question is how can I call between these servers?
>> >
>> > if I can , how shoould I set in *.xml? or X-Lite?
>> >
>> > thank you for your attension
>> >
>> >
>> > Best Regards
>> > Gary
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>> >
>> > _______________________________________________ FreeSWITCH-users mailing
>> > list FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

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


_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 		 	   		  
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From msc at freeswitch.org  Thu Oct 14 17:56:43 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Oct 2010 17:56:43 -0700
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <SNT130-w1462D55D67F6FED40D4D74FB570@phx.gbl>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>
	<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>
	<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>
	<SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>
	<AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>
	<SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>
	<C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>
	<SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>
	<AANLkTimsBQtco2B-s6uXbkrQ7WoiQ0SzF8N5=wSskOML@mail.gmail.com>
	<AANLkTikTSTtaN1cd14U+G8yKA1u=A3dCgtKgSrEkYDb-@mail.gmail.com>
	<SNT130-w1462D55D67F6FED40D4D74FB570@phx.gbl>
Message-ID: <AANLkTikbix9D=E+eAueVHWqzDM-5_SV=xb8z9_NastDF@mail.gmail.com>

Yes! You don't need a gateway for this. If the users are all on the same
server then you just need to create the necessary files in
conf/directory/default/
You already have 1000.xml, 1001.xml, ... 1019.xml
You just need to add 2010.xml, 4001.xml, 3009.xml, etc.
Check out this article:
http://bit.ly/EpVrv

It's a bit older, but the section on adding a new extension is very
accurate. In that example it shows how to add extension 1500 to your
directory and your dialplan. You can copy that example for all of your new
extension numbers.

-MC

2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>

>  Hello
>
> but I can call to each other on the same server
>
> for example
>
> I can use 1007 at 37  call 1013 at 37  ,  4001 at 37  , 2010 at 37 , 3009 at 37
>
>
>  Thank you for your attension
> Best Regards
>  Gary
>
>
>
> ------------------------------
> Date: Thu, 14 Oct 2010 11:27:31 -0700
> From: msc at freeswitch.org
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> FYI, we have several resources to help
> http://wiki.freeswitch.org/wiki/Regular_Expression
> http://bit.ly/aijtAC
>
> :)
>
> -MC
>
> On Thu, Oct 14, 2010 at 11:18 AM, Jeffrey Leung <curriegrad2004 at gmail.com>wrote:
>
> Dialplan's regex doesn't look right to me. You'll need to fix that
> before you do anything else.
>
> I'd suggest this regex instead of the one you provided:
> ^[0-4]0(10[1-9][0-9])$
>
> instead of this one
> ^([0-4]0[01][0-9])$
>
> If you can't get the grasp of regular expressions, I'd highly suggest
> you to pick up a book and read up on it. Knowing regular expressions
> is a great skill to have ;)
>
> 2010/10/14 ? Chiang ?? Chih-Chung   Wybie <tear152 at hotmail.com>:
> > I tried use 1013 at 38 calls to 1019 at 37
> >
> > show log below
> >
> > 2010-10-14 17:50:41.249876 [NOTICE] switch_channel.c:675 New Channel
> > sofia/internal/1013 at 60.248.175.38 [875fdeef-b94e-41fb-a621-ea005bbaedbd]
> > 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
> > 1013->1019 in context public
> > 2010-10-14 17:50:41.263809 [NOTICE] switch_ivr.c:1450 Transfer
> > sofia/internal/1013 at 60.248.175.38 to XML[1019 at default]
> > 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
> > 1013->1019 in context default
> > 2010-10-14 17:50:41.283904 [INFO] switch_core_session.c:1764 Sending
> early
> > media
> > 2010-10-14 17:50:41.285023 [NOTICE] mod_sofia.c:1994 Pre-Answer
> > sofia/internal/1013 at 60.248.175.38!
> >
> > seems not work
> >
> > Thank you for you attension
> > Best Regards
> >
> > Gary
> >
> >
> >
> > ________________________________
> > From: david.ponzone at ipeva.fr
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Thu, 14 Oct 2010 10:16:28 +0200
> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >
> > I think we would need a complete log from the target FS, in order to
> provide
> > you with any efficient help.
> > Thank
> > David Ponzone  Direction Technique
> > email: david.ponzone at ipeva.fr
> > tel:      01 74 03 18 97
> > gsm:   06 66 98 76 34
> > Service Client IPeva
> > tel:      0811 46 26 26
> > www.ipeva.fr  -   www.ipeva-studio.com
> > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> > l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion
> > non autoris?e est interdite. Tout message ?lectronique est susceptible
> > d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il
> > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> >
> >
> >
> > Le 14/10/2010 ? 09:55, ? Chiang ?? Chih-Chung Wybie a ?crit :
> >
> > I tried Connect Two FreeSWITCH
> > Boxes  http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
> > and my configuration :
> >
> > 60.248.175.37
> >
> > (autoload_configs/act.conf.xml)
> > <list name="domains" default="deny">
> >       <node type="allow" cidr="60.248.175.38/28"/>
> >      <!-- <node type="allow" domain="$${domain}"/>-->
> > </list>
> >
> > (dialplan/default.xml)
> >
> > <extension name="Dial to 38">
> >       <condition field="destination_number"
> > expression="^([0-4]0[01][0-9])$">
> >         <action application="bridge"
> > data="sofia/internal/$1 at 60.248.175.38"/>
> >       </condition>
> >  </extension>
> >
> > (dialplan/public.xml)
> >
> > <extension name="Calls from 38">
> >       <condition field="destination_number"
> > expression="^([0-4]0[01][0-9])$">
> >         <action application="transfer" data="$1 XML default"/>
> >       </condition>
> >  </extension>
> >
> >
> >
> > 60.248.175.38
> >
> > (autoload_configs/act.conf.xml)
> > <list name="domains" default="deny">
> >       <node type="allow" cidr="60.248.175.37/28"/>
> >      <!-- <node type="allow" domain="$${domain}"/>-->
> > </list>
> >
> > (dialplan/default.xml)
> >
> > <extension name="Dial to 37">
> >       <condition field="destination_number"
> > expression="^([0-4]0[01][0-9])$">
> >         <action application="bridge"
> > data="sofia/internal/$1 at 60.248.175.37"/>
> >       </condition>
> >  </extension>
> >
> > (dialplan/public.xml)
> >
> > <extension name="Calls from 37">
> >       <condition field="destination_number"
> > expression="^([0-4]0[01][0-9])$">
> >         <action application="transfer" data="$1 XML default"/>
> >       </condition>
> >  </extension>
> >
> > when I use 1007 at 38 calls to 1007 at 37
> > it won't show errors but still not work
> >
> > how should I change something ?
> >
> > Thank you for your attension
> >
> > Best Regards
> > Gary
> >
> >
> >
> >
> >
> >> Date: Wed, 13 Oct 2010 22:17:13 -0700
> >> From: curriegrad2004 at gmail.com
> >> To: freeswitch-users at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >>
> >> You can create a SIP trunk between the two servers so calls can be
> >> routed back and forth from the servers you've specified.
> >>
> >> 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
> >> > I tried it
> >> > but not working
> >> > and show errors below
> >> > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493 Cannot
> >> > create
> >> > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> >> > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493 Cannot
> >> > create
> >> > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> >> > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate Failed.
> >> > Cause: USER_NOT_REGISTERED
> >> >
> >> > and for example
> >> >  I use SIP server 60.248.175.38  user id=1007
> >> > and I call to 1007 at 60.248.175.37    that will show these errors
> >> >
> >> > how should set more?
> >> >
> >> > thank you for your attension
> >> >
> >> > Best Regards
> >> > Gary
> >> >
> >> > ________________________________
> >> > Date: Thu, 14 Oct 2010 10:55:49 +0800
> >> > From: xyangni at gmail.com
> >> > To: freeswitch-users at lists.freeswitch.org
> >> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >> >
> >> > I think you can reg at server A and than call B as a normal SIP call
> to
> >> > external domain.
> >> >
> >> > 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
> >> >
> >> >
> >> >
> >> > Hello
> >> >
> >> > I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X-Lite to
> >> > call
> >> >
> >> > my question is how can I call between these servers?
> >> >
> >> > if I can , how shoould I set in *.xml? or X-Lite?
> >> >
> >> > thank you for your attension
> >> >
> >> >
> >> > Best Regards
> >> > Gary
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >> >
> >> > _______________________________________________ FreeSWITCH-users
> mailing
> >> > list FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________ FreeSWITCH-users mailing
> > list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Oct 14 17:58:47 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Oct 2010 17:58:47 -0700
Subject: [Freeswitch-users] Freeswitch Video Capture and Playback
In-Reply-To: <87bp6we05a.fsf@quasar.esben-stien.name>
References: <87k4xlga1k.fsf@quasar.esben-stien.name>
	<87pr6yz5wa.fsf@quasar.esben-stien.name>
	<87bp6we05a.fsf@quasar.esben-stien.name>
Message-ID: <AANLkTimtQsE7+=TQ4eedswLNy7my223z=rEtrjXiFqV+@mail.gmail.com>

Also, Brian informs me that there may be a bug with this. I will confer with
Tony tomorrow and find out.
-MC

On Thu, Oct 14, 2010 at 6:05 PM, Esben Stien <b0ef at esben-stien.name> wrote:

> Esben Stien <b0ef at esben-stien.name> writes:
>
> > So nobody is using video with freeswitch?.
>
> I've tried with Ekiga and Empathy.
>
> Has anybody gotten Freeswitch to record some video with any free
> client?.
>
> Is there any info on the Freeswitch Video Format, FSV?. Maybe how to
> encode a video with gstreamer/mencoder?.
>
> Does there exist a FSV test file?. If anyone got one of these FSV files,
> please upload a sample.
>
> Does something like video loopback exist, like a video echo
> application for Freeswitch?.
>
> --
> Esben Stien is b0ef at e     s      a
>         http://www. s     t    n m
>          irc://irc.  b  -  i  .   e/%23contact
>           sip:b0ef@   e     e
>           jid:b0ef@    n     n
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tear152 at hotmail.com  Thu Oct 14 18:12:31 2010
From: tear152 at hotmail.com (=?gb2312?B?va0gQ2hpYW5nINbB1tAgQ2hpaC1DaHVuZyAgIFd5Ymll?=)
Date: Fri, 15 Oct 2010 09:12:31 +0800
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <AANLkTikbix9D=E+eAueVHWqzDM-5_SV=xb8z9_NastDF@mail.gmail.com>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>,
	<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>,
	<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>,
	<SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>,
	<AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>,
	<SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>,
	<C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>,
	<SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>,
	<AANLkTimsBQtco2B-s6uXbkrQ7WoiQ0SzF8N5=wSskOML@mail.gmail.com>,
	<AANLkTikTSTtaN1cd14U+G8yKA1u=A3dCgtKgSrEkYDb-@mail.gmail.com>,
	<SNT130-w1462D55D67F6FED40D4D74FB570@phx.gbl>,
	<AANLkTikbix9D=E+eAueVHWqzDM-5_SV=xb8z9_NastDF@mail.gmail.com>
Message-ID: <SNT130-w3B5414501256848B3E382FB570@phx.gbl>


But my question is 
 
I set two fs
 
and I want to connect them
 
I refer to http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
 
it 's not working
 
so I give my configuration already
 
should I change some setting?
 
Thank you for your attension
 
Best Regards
 
Gary
 


Date: Thu, 14 Oct 2010 17:56:43 -0700
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite

Yes! You don't need a gateway for this. If the users are all on the same server then you just need to create the necessary files in conf/directory/default/
You already have 1000.xml, 1001.xml, ... 1019.xml
You just need to add 2010.xml, 4001.xml, 3009.xml, etc.
Check out this article:
http://bit.ly/EpVrv

It's a bit older, but the section on adding a new extension is very accurate. In that example it shows how to add extension 1500 to your directory and your dialplan. You can copy that example for all of your new extension numbers.

-MC


2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>


Hello
 
but I can call to each other on the same server
 
for example 
 
I can use 1007 at 37  call 1013 at 37  ,  4001 at 37  , 2010 at 37 , 3009 at 37 

 
 
 Thank you for your attension
Best Regards
 Gary


 


Date: Thu, 14 Oct 2010 11:27:31 -0700
From: msc at freeswitch.org



To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite

FYI, we have several resources to help
http://wiki.freeswitch.org/wiki/Regular_Expression
http://bit.ly/aijtAC

:)

-MC


On Thu, Oct 14, 2010 at 11:18 AM, Jeffrey Leung <curriegrad2004 at gmail.com> wrote:

Dialplan's regex doesn't look right to me. You'll need to fix that
before you do anything else.

I'd suggest this regex instead of the one you provided:
^[0-4]0(10[1-9][0-9])$

instead of this one
^([0-4]0[01][0-9])$

If you can't get the grasp of regular expressions, I'd highly suggest
you to pick up a book and read up on it. Knowing regular expressions
is a great skill to have ;)

2010/10/14 ? Chiang ?? Chih-Chung   Wybie <tear152 at hotmail.com>:
> I tried use 1013 at 38 calls to 1019 at 37
>
> show log below
>
> 2010-10-14 17:50:41.249876 [NOTICE] switch_channel.c:675 New Channel
> sofia/internal/1013 at 60.248.175.38 [875fdeef-b94e-41fb-a621-ea005bbaedbd]
> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
> 1013->1019 in context public
> 2010-10-14 17:50:41.263809 [NOTICE] switch_ivr.c:1450 Transfer
> sofia/internal/1013 at 60.248.175.38 to XML[1019 at default]
> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
> 1013->1019 in context default
> 2010-10-14 17:50:41.283904 [INFO] switch_core_session.c:1764 Sending early
> media
> 2010-10-14 17:50:41.285023 [NOTICE] mod_sofia.c:1994 Pre-Answer
> sofia/internal/1013 at 60.248.175.38!
>
> seems not work
>
> Thank you for you attension
> Best Regards
>
> Gary
>
>
>
> ________________________________
> From: david.ponzone at ipeva.fr
> To: freeswitch-users at lists.freeswitch.org
> Date: Thu, 14 Oct 2010 10:16:28 +0200
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> I think we would need a complete log from the target FS, in order to provide
> you with any efficient help.
> Thank
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 14/10/2010 ? 09:55, ? Chiang ?? Chih-Chung Wybie a ?crit :
>
> I tried Connect Two FreeSWITCH
> Boxes  http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
> and my configuration :
>
> 60.248.175.37
>
> (autoload_configs/act.conf.xml)
> <list name="domains" default="deny">
>       <node type="allow" cidr="60.248.175.38/28"/>
>      <!-- <node type="allow" domain="$${domain}"/>-->
> </list>
>
> (dialplan/default.xml)
>
> <extension name="Dial to 38">
>       <condition field="destination_number"
> expression="^([0-4]0[01][0-9])$">
>         <action application="bridge"
> data="sofia/internal/$1 at 60.248.175.38"/>
>       </condition>
>  </extension>
>
> (dialplan/public.xml)
>
> <extension name="Calls from 38">
>       <condition field="destination_number"
> expression="^([0-4]0[01][0-9])$">
>         <action application="transfer" data="$1 XML default"/>
>       </condition>
>  </extension>
>
>
>
> 60.248.175.38
>
> (autoload_configs/act.conf.xml)
> <list name="domains" default="deny">
>       <node type="allow" cidr="60.248.175.37/28"/>
>      <!-- <node type="allow" domain="$${domain}"/>-->
> </list>
>
> (dialplan/default.xml)
>
> <extension name="Dial to 37">
>       <condition field="destination_number"
> expression="^([0-4]0[01][0-9])$">
>         <action application="bridge"
> data="sofia/internal/$1 at 60.248.175.37"/>
>       </condition>
>  </extension>
>
> (dialplan/public.xml)
>
> <extension name="Calls from 37">
>       <condition field="destination_number"
> expression="^([0-4]0[01][0-9])$">
>         <action application="transfer" data="$1 XML default"/>
>       </condition>
>  </extension>
>
> when I use 1007 at 38 calls to 1007 at 37
> it won't show errors but still not work
>
> how should I change something ?
>
> Thank you for your attension
>
> Best Regards
> Gary
>
>
>
>
>
>> Date: Wed, 13 Oct 2010 22:17:13 -0700
>> From: curriegrad2004 at gmail.com
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>>
>> You can create a SIP trunk between the two servers so calls can be
>> routed back and forth from the servers you've specified.
>>
>> 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
>> > I tried it
>> > but not working
>> > and show errors below
>> > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493 Cannot
>> > create
>> > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
>> > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493 Cannot
>> > create
>> > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
>> > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate Failed.
>> > Cause: USER_NOT_REGISTERED
>> >
>> > and for example
>> >  I use SIP server 60.248.175.38  user id=1007
>> > and I call to 1007 at 60.248.175.37    that will show these errors
>> >
>> > how should set more?
>> >
>> > thank you for your attension
>> >
>> > Best Regards
>> > Gary
>> >
>> > ________________________________
>> > Date: Thu, 14 Oct 2010 10:55:49 +0800
>> > From: xyangni at gmail.com
>> > To: freeswitch-users at lists.freeswitch.org
>> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>> >
>> > I think you can reg at server A and than call B as a normal SIP call to
>> > external domain.
>> >
>> > 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
>> >
>> >
>> >
>> > Hello
>> >
>> > I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X-Lite to
>> > call
>> >
>> > my question is how can I call between these servers?
>> >
>> > if I can , how shoould I set in *.xml? or X-Lite?
>> >
>> > thank you for your attension
>> >
>> >
>> > Best Regards
>> > Gary
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>> >
>> > _______________________________________________ FreeSWITCH-users mailing
>> > list FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

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


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



_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 		 	   		  
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From mario_fs at mgtech.com  Thu Oct 14 19:43:37 2010
From: mario_fs at mgtech.com (Mario)
Date: Thu, 14 Oct 2010 19:43:37 -0700
Subject: [Freeswitch-users] early media problem?
Message-ID: <4CB7BFD9.3070000@mgtech.com>

I had an extension working fine that played a file, ringed, etc. to an
inbound caller before we picked up. Trying to fix a problem the ITSP
moved the account and now the caller only hears ringing, none of my
stuff is heard even though it's shows as executed in trace. I looked on
the wiki and tried adding <action application="ring_ready"/> at the top
but that did not help. Anyone have any idea how to get all my work
working again? Is there something I can change or the ITSP? Thanks.
Mario


From msc at freeswitch.org  Thu Oct 14 20:02:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Oct 2010 20:02:16 -0700
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <SNT130-w3B5414501256848B3E382FB570@phx.gbl>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>
	<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>
	<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>
	<SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>
	<AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>
	<SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>
	<C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>
	<SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>
	<AANLkTimsBQtco2B-s6uXbkrQ7WoiQ0SzF8N5=wSskOML@mail.gmail.com>
	<AANLkTikTSTtaN1cd14U+G8yKA1u=A3dCgtKgSrEkYDb-@mail.gmail.com>
	<SNT130-w1462D55D67F6FED40D4D74FB570@phx.gbl>
	<AANLkTikbix9D=E+eAueVHWqzDM-5_SV=xb8z9_NastDF@mail.gmail.com>
	<SNT130-w3B5414501256848B3E382FB570@phx.gbl>
Message-ID: <AANLkTik=eus1MrmnpCFRvfDCGpE7WYmrf1QgQpnJ=Vg3@mail.gmail.com>

Did you try using ACLs or with putting a user/gateway on each side?
Put your configs on pastebin.freeswitch.org.

This is actually a simple exercise once you know what to do. Start by
deciding if you want to do authentication by digest or IP address.

-MC

2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>

>  But my question is
>
> I set two fs
>
> and I want to connect them
>
> I refer to http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
>
> it 's not working
>
> so I give my configuration already
>
> should I change some setting?
>
>
> Thank you for your attension
>
> Best Regards
>
> Gary
>
> ------------------------------
> Date: Thu, 14 Oct 2010 17:56:43 -0700
>
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> Yes! You don't need a gateway for this. If the users are all on the same
> server then you just need to create the necessary files in
> conf/directory/default/
> You already have 1000.xml, 1001.xml, ... 1019.xml
> You just need to add 2010.xml, 4001.xml, 3009.xml, etc.
> Check out this article:
> http://bit.ly/EpVrv
>
> It's a bit older, but the section on adding a new extension is very
> accurate. In that example it shows how to add extension 1500 to your
> directory and your dialplan. You can copy that example for all of your new
> extension numbers.
>
> -MC
>
> 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
>
> Hello
>
> but I can call to each other on the same server
>
> for example
>
> I can use 1007 at 37  call 1013 at 37  ,  4001 at 37  , 2010 at 37 , 3009 at 37
>
>
>  Thank you for your attension
> Best Regards
>  Gary
>
>
>
> ------------------------------
> Date: Thu, 14 Oct 2010 11:27:31 -0700
> From: msc at freeswitch.org
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> FYI, we have several resources to help
> http://wiki.freeswitch.org/wiki/Regular_Expression
> http://bit.ly/aijtAC
>
> :)
>
> -MC
>
> On Thu, Oct 14, 2010 at 11:18 AM, Jeffrey Leung <curriegrad2004 at gmail.com>wrote:
>
> Dialplan's regex doesn't look right to me. You'll need to fix that
> before you do anything else.
>
> I'd suggest this regex instead of the one you provided:
> ^[0-4]0(10[1-9][0-9])$
>
> instead of this one
> ^([0-4]0[01][0-9])$
>
> If you can't get the grasp of regular expressions, I'd highly suggest
> you to pick up a book and read up on it. Knowing regular expressions
> is a great skill to have ;)
>
> 2010/10/14 ? Chiang ?? Chih-Chung   Wybie <tear152 at hotmail.com>:
> > I tried use 1013 at 38 calls to 1019 at 37
> >
> > show log below
> >
> > 2010-10-14 17:50:41.249876 [NOTICE] switch_channel.c:675 New Channel
> > sofia/internal/1013 at 60.248.175.38 [875fdeef-b94e-41fb-a621-ea005bbaedbd]
> > 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
> > 1013->1019 in context public
> > 2010-10-14 17:50:41.263809 [NOTICE] switch_ivr.c:1450 Transfer
> > sofia/internal/1013 at 60.248.175.38 to XML[1019 at default]
> > 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
> > 1013->1019 in context default
> > 2010-10-14 17:50:41.283904 [INFO] switch_core_session.c:1764 Sending
> early
> > media
> > 2010-10-14 17:50:41.285023 [NOTICE] mod_sofia.c:1994 Pre-Answer
> > sofia/internal/1013 at 60.248.175.38!
> >
> > seems not work
> >
> > Thank you for you attension
> > Best Regards
> >
> > Gary
> >
> >
> >
> > ________________________________
> > From: david.ponzone at ipeva.fr
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Thu, 14 Oct 2010 10:16:28 +0200
> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >
> > I think we would need a complete log from the target FS, in order to
> provide
> > you with any efficient help.
> > Thank
> > David Ponzone  Direction Technique
> > email: david.ponzone at ipeva.fr
> > tel:      01 74 03 18 97
> > gsm:   06 66 98 76 34
> > Service Client IPeva
> > tel:      0811 46 26 26
> > www.ipeva.fr  -   www.ipeva-studio.com
> > Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> > l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion
> > non autoris?e est interdite. Tout message ?lectronique est susceptible
> > d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il
> > a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> > message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> >
> >
> >
> > Le 14/10/2010 ? 09:55, ? Chiang ?? Chih-Chung Wybie a ?crit :
> >
> > I tried Connect Two FreeSWITCH
> > Boxes  http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
> > and my configuration :
> >
> > 60.248.175.37
> >
> > (autoload_configs/act.conf.xml)
> > <list name="domains" default="deny">
> >       <node type="allow" cidr="60.248.175.38/28"/>
> >      <!-- <node type="allow" domain="$${domain}"/>-->
> > </list>
> >
> > (dialplan/default.xml)
> >
> > <extension name="Dial to 38">
> >       <condition field="destination_number"
> > expression="^([0-4]0[01][0-9])$">
> >         <action application="bridge"
> > data="sofia/internal/$1 at 60.248.175.38"/>
> >       </condition>
> >  </extension>
> >
> > (dialplan/public.xml)
> >
> > <extension name="Calls from 38">
> >       <condition field="destination_number"
> > expression="^([0-4]0[01][0-9])$">
> >         <action application="transfer" data="$1 XML default"/>
> >       </condition>
> >  </extension>
> >
> >
> >
> > 60.248.175.38
> >
> > (autoload_configs/act.conf.xml)
> > <list name="domains" default="deny">
> >       <node type="allow" cidr="60.248.175.37/28"/>
> >      <!-- <node type="allow" domain="$${domain}"/>-->
> > </list>
> >
> > (dialplan/default.xml)
> >
> > <extension name="Dial to 37">
> >       <condition field="destination_number"
> > expression="^([0-4]0[01][0-9])$">
> >         <action application="bridge"
> > data="sofia/internal/$1 at 60.248.175.37"/>
> >       </condition>
> >  </extension>
> >
> > (dialplan/public.xml)
> >
> > <extension name="Calls from 37">
> >       <condition field="destination_number"
> > expression="^([0-4]0[01][0-9])$">
> >         <action application="transfer" data="$1 XML default"/>
> >       </condition>
> >  </extension>
> >
> > when I use 1007 at 38 calls to 1007 at 37
> > it won't show errors but still not work
> >
> > how should I change something ?
> >
> > Thank you for your attension
> >
> > Best Regards
> > Gary
> >
> >
> >
> >
> >
> >> Date: Wed, 13 Oct 2010 22:17:13 -0700
> >> From: curriegrad2004 at gmail.com
> >> To: freeswitch-users at lists.freeswitch.org
> >> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >>
> >> You can create a SIP trunk between the two servers so calls can be
> >> routed back and forth from the servers you've specified.
> >>
> >> 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
> >> > I tried it
> >> > but not working
> >> > and show errors below
> >> > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493 Cannot
> >> > create
> >> > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> >> > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493 Cannot
> >> > create
> >> > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> >> > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate Failed.
> >> > Cause: USER_NOT_REGISTERED
> >> >
> >> > and for example
> >> >  I use SIP server 60.248.175.38  user id=1007
> >> > and I call to 1007 at 60.248.175.37    that will show these errors
> >> >
> >> > how should set more?
> >> >
> >> > thank you for your attension
> >> >
> >> > Best Regards
> >> > Gary
> >> >
> >> > ________________________________
> >> > Date: Thu, 14 Oct 2010 10:55:49 +0800
> >> > From: xyangni at gmail.com
> >> > To: freeswitch-users at lists.freeswitch.org
> >> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >> >
> >> > I think you can reg at server A and than call B as a normal SIP call
> to
> >> > external domain.
> >> >
> >> > 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
> >> >
> >> >
> >> >
> >> > Hello
> >> >
> >> > I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X-Lite to
> >> > call
> >> >
> >> > my question is how can I call between these servers?
> >> >
> >> > if I can , how shoould I set in *.xml? or X-Lite?
> >> >
> >> > thank you for your attension
> >> >
> >> >
> >> > Best Regards
> >> > Gary
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >> >
> >> > _______________________________________________ FreeSWITCH-users
> mailing
> >> > list FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________ FreeSWITCH-users mailing
> > list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Oct 14 20:03:10 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Oct 2010 20:03:10 -0700
Subject: [Freeswitch-users] early media problem?
In-Reply-To: <4CB7BFD9.3070000@mgtech.com>
References: <4CB7BFD9.3070000@mgtech.com>
Message-ID: <AANLkTi=t0Fyaeu8=4N9MNN-Y-=X29EMGdpV-pUH69X1d@mail.gmail.com>

pastebin the debug log of the call coming in so we can see exactly what is
happening.
-MC

On Thu, Oct 14, 2010 at 7:43 PM, Mario <mario_fs at mgtech.com> wrote:

> I had an extension working fine that played a file, ringed, etc. to an
> inbound caller before we picked up. Trying to fix a problem the ITSP
> moved the account and now the caller only hears ringing, none of my
> stuff is heard even though it's shows as executed in trace. I looked on
> the wiki and tried adding <action application="ring_ready"/> at the top
> but that did not help. Anyone have any idea how to get all my work
> working again? Is there something I can change or the ITSP? Thanks.
> Mario
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
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From mario_fs at mgtech.com  Thu Oct 14 20:27:26 2010
From: mario_fs at mgtech.com (Mario)
Date: Thu, 14 Oct 2010 20:27:26 -0700
Subject: [Freeswitch-users] early media problem?
In-Reply-To: <AANLkTi=t0Fyaeu8=4N9MNN-Y-=X29EMGdpV-pUH69X1d@mail.gmail.com>
References: <4CB7BFD9.3070000@mgtech.com>
	<AANLkTi=t0Fyaeu8=4N9MNN-Y-=X29EMGdpV-pUH69X1d@mail.gmail.com>
Message-ID: <4CB7CA1E.2040507@mgtech.com>

http://pastebin.freeswitch.org/14241

My first pastebin! Thanks..

On 10/14/10 20:03, Michael Collins wrote:
> pastebin the debug log of the call coming in so we can see exactly what
> is happening.
> -MC
> 
> On Thu, Oct 14, 2010 at 7:43 PM, Mario <mario_fs at mgtech.com
> <mailto:mario_fs at mgtech.com>> wrote:
> 
>     I had an extension working fine that played a file, ringed, etc. to an
>     inbound caller before we picked up. Trying to fix a problem the ITSP
>     moved the account and now the caller only hears ringing, none of my
>     stuff is heard even though it's shows as executed in trace. I looked on
>     the wiki and tried adding <action application="ring_ready"/> at the top
>     but that did not help. Anyone have any idea how to get all my work
>     working again? Is there something I can change or the ITSP? Thanks.
>     Mario
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From tear152 at hotmail.com  Thu Oct 14 20:26:53 2010
From: tear152 at hotmail.com (=?gb2312?B?va0gQ2hpYW5nINbB1tAgQ2hpaC1DaHVuZyAgIFd5Ymll?=)
Date: Fri, 15 Oct 2010 11:26:53 +0800
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <AANLkTik=eus1MrmnpCFRvfDCGpE7WYmrf1QgQpnJ=Vg3@mail.gmail.com>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>,
	<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>,
	<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>,
	<SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>,
	<AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>,
	<SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>,
	<C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>,
	<SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>,
	<AANLkTimsBQtco2B-s6uXbkrQ7WoiQ0SzF8N5=wSskOML@mail.gmail.com>,
	<AANLkTikTSTtaN1cd14U+G8yKA1u=A3dCgtKgSrEkYDb-@mail.gmail.com>,
	<SNT130-w1462D55D67F6FED40D4D74FB570@phx.gbl>,
	<AANLkTikbix9D=E+eAueVHWqzDM-5_SV=xb8z9_NastDF@mail.gmail.com>,
	<SNT130-w3B5414501256848B3E382FB570@phx.gbl>,
	<AANLkTik=eus1MrmnpCFRvfDCGpE7WYmrf1QgQpnJ=Vg3@mail.gmail.com>
Message-ID: <SNT130-w486F7895E102C75B06D2F3FB570@phx.gbl>


Well I can't login pastebin.freeswitch.org
and that's my configs below
 
IP address: 60.248.175.37
Mask       :255.255.255.240
GateWay : 60.248.175.33
DNS        :168.95.1.1

 (autoload_configs/act.conf.xml)
 <list name="domains" default="deny">
      <node type="allow" cidr="60.248.175.32/28"/>
      <!-- <node type="allow" domain="$${domain}"/>-->
 </list>

 (dialplan/default.xml)
 <extension name="Dial to 38">
       <condition field="destination_number"
 expression="^([0-4]0[01][0-9])$">
       <action application="bridge"
 data="sofia/internal/$1 at 60.248.175.38"/>
       </condition>
 </extension>

 (dialplan/public.xml)

 <extension name="Calls from 38">
      <condition field="destination_number"
 expression="^([0-4]0[01][0-9])$">
        <action application="transfer" data="$1 XML default"/>
       </condition>
  </extension>




IP address: 60.248.175.38
Mask       :255.255.255.240
GateWay : 60.248.175.33
DNS        :168.95.1.1
 

 (autoload_configs/act.conf.xml)
 <list name="domains" default="deny">
       <node type="allow" cidr="60.248.175.32/28"/>
      <!-- <node type="allow" domain="$${domain}"/>-->
 </list>

 (dialplan/default.xml)

 <extension name="Dial to 37">
       <condition field="destination_number"
 expression="^([0-4]0[01][0-9])$">
         <action application="bridge"
 data="sofia/internal/$1 at 60.248.175.37"/>
       </condition>
  </extension>

 (dialplan/public.xml)

 <extension name="Calls from 37">
       <condition field="destination_number"
 expression="^([0-4]0[01][0-9])$">
         <action application="transfer" data="$1 XML default"/>
       </condition>
  </extension>

Thank you for your attension
 
Best Regards
 
Gary
 

 


Date: Thu, 14 Oct 2010 20:02:16 -0700
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite

Did you try using ACLs or with putting a user/gateway on each side?
Put your configs on pastebin.freeswitch.org.

This is actually a simple exercise once you know what to do. Start by deciding if you want to do authentication by digest or IP address.

-MC


2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>


But my question is 
 
I set two fs
 
and I want to connect them
 
I refer to http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
 
it 's not working
 
so I give my configuration already
 
should I change some setting?

 
Thank you for your attension
 
Best Regards
 
Gary
 


Date: Thu, 14 Oct 2010 17:56:43 -0700



From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite

Yes! You don't need a gateway for this. If the users are all on the same server then you just need to create the necessary files in conf/directory/default/
You already have 1000.xml, 1001.xml, ... 1019.xml
You just need to add 2010.xml, 4001.xml, 3009.xml, etc.
Check out this article:
http://bit.ly/EpVrv

It's a bit older, but the section on adding a new extension is very accurate. In that example it shows how to add extension 1500 to your directory and your dialplan. You can copy that example for all of your new extension numbers.

-MC


2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>


Hello
 
but I can call to each other on the same server
 
for example 
 
I can use 1007 at 37  call 1013 at 37  ,  4001 at 37  , 2010 at 37 , 3009 at 37 

 
 
 Thank you for your attension
Best Regards
 Gary


 


Date: Thu, 14 Oct 2010 11:27:31 -0700
From: msc at freeswitch.org 



To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite

FYI, we have several resources to help
http://wiki.freeswitch.org/wiki/Regular_Expression
http://bit.ly/aijtAC

:)

-MC


On Thu, Oct 14, 2010 at 11:18 AM, Jeffrey Leung <curriegrad2004 at gmail.com> wrote:

Dialplan's regex doesn't look right to me. You'll need to fix that
before you do anything else.

I'd suggest this regex instead of the one you provided:
^[0-4]0(10[1-9][0-9])$

instead of this one
^([0-4]0[01][0-9])$

If you can't get the grasp of regular expressions, I'd highly suggest
you to pick up a book and read up on it. Knowing regular expressions
is a great skill to have ;)

2010/10/14 ? Chiang ?? Chih-Chung   Wybie <tear152 at hotmail.com>:
> I tried use 1013 at 38 calls to 1019 at 37
>
> show log below
>
> 2010-10-14 17:50:41.249876 [NOTICE] switch_channel.c:675 New Channel
> sofia/internal/1013 at 60.248.175.38 [875fdeef-b94e-41fb-a621-ea005bbaedbd]
> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
> 1013->1019 in context public
> 2010-10-14 17:50:41.263809 [NOTICE] switch_ivr.c:1450 Transfer
> sofia/internal/1013 at 60.248.175.38 to XML[1019 at default]
> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
> 1013->1019 in context default
> 2010-10-14 17:50:41.283904 [INFO] switch_core_session.c:1764 Sending early
> media
> 2010-10-14 17:50:41.285023 [NOTICE] mod_sofia.c:1994 Pre-Answer
> sofia/internal/1013 at 60.248.175.38!
>
> seems not work
>
> Thank you for you attension
> Best Regards
>
> Gary
>
>
>
> ________________________________
> From: david.ponzone at ipeva.fr
> To: freeswitch-users at lists.freeswitch.org
> Date: Thu, 14 Oct 2010 10:16:28 +0200
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> I think we would need a complete log from the target FS, in order to provide
> you with any efficient help.
> Thank
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 14/10/2010 ? 09:55, ? Chiang ?? Chih-Chung Wybie a ?crit :
>
> I tried Connect Two FreeSWITCH
> Boxes  http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
> and my configuration :
>
> 60.248.175.37
>
> (autoload_configs/act.conf.xml)
> <list name="domains" default="deny">
>       <node type="allow" cidr="60.248.175.38/28"/>
>      <!-- <node type="allow" domain="$${domain}"/>-->
> </list>
>
> (dialplan/default.xml)
>
> <extension name="Dial to 38">
>       <condition field="destination_number"
> expression="^([0-4]0[01][0-9])$">
>         <action application="bridge"
> data="sofia/internal/$1 at 60.248.175.38"/>
>       </condition>
>  </extension>
>
> (dialplan/public.xml)
>
> <extension name="Calls from 38">
>       <condition field="destination_number"
> expression="^([0-4]0[01][0-9])$">
>         <action application="transfer" data="$1 XML default"/>
>       </condition>
>  </extension>
>
>
>
> 60.248.175.38
>
> (autoload_configs/act.conf.xml)
> <list name="domains" default="deny">
>       <node type="allow" cidr="60.248.175.37/28"/>
>      <!-- <node type="allow" domain="$${domain}"/>-->
> </list>
>
> (dialplan/default.xml)
>
> <extension name="Dial to 37">
>       <condition field="destination_number"
> expression="^([0-4]0[01][0-9])$">
>         <action application="bridge"
> data="sofia/internal/$1 at 60.248.175.37"/>
>       </condition>
>  </extension>
>
> (dialplan/public.xml)
>
> <extension name="Calls from 37">
>       <condition field="destination_number"
> expression="^([0-4]0[01][0-9])$">
>         <action application="transfer" data="$1 XML default"/>
>       </condition>
>  </extension>
>
> when I use 1007 at 38 calls to 1007 at 37
> it won't show errors but still not work
>
> how should I change something ?
>
> Thank you for your attension
>
> Best Regards
> Gary
>
>
>
>
>
>> Date: Wed, 13 Oct 2010 22:17:13 -0700
>> From: curriegrad2004 at gmail.com
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>>
>> You can create a SIP trunk between the two servers so calls can be
>> routed back and forth from the servers you've specified.
>>
>> 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
>> > I tried it
>> > but not working
>> > and show errors below
>> > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493 Cannot
>> > create
>> > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
>> > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493 Cannot
>> > create
>> > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
>> > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate Failed.
>> > Cause: USER_NOT_REGISTERED
>> >
>> > and for example
>> >  I use SIP server 60.248.175.38  user id=1007
>> > and I call to 1007 at 60.248.175.37    that will show these errors
>> >
>> > how should set more?
>> >
>> > thank you for your attension
>> >
>> > Best Regards
>> > Gary
>> >
>> > ________________________________
>> > Date: Thu, 14 Oct 2010 10:55:49 +0800
>> > From: xyangni at gmail.com
>> > To: freeswitch-users at lists.freeswitch.org
>> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>> >
>> > I think you can reg at server A and than call B as a normal SIP call to
>> > external domain.
>> >
>> > 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
>> >
>> >
>> >
>> > Hello
>> >
>> > I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X-Lite to
>> > call
>> >
>> > my question is how can I call between these servers?
>> >
>> > if I can , how shoould I set in *.xml? or X-Lite?
>> >
>> > thank you for your attension
>> >
>> >
>> > Best Regards
>> > Gary
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>> >
>> > _______________________________________________ FreeSWITCH-users mailing
>> > list FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From curriegrad2004 at gmail.com  Thu Oct 14 21:22:19 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Thu, 14 Oct 2010 21:22:19 -0700
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <SNT130-w486F7895E102C75B06D2F3FB570@phx.gbl>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>
	<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>
	<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>
	<SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>
	<AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>
	<SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>
	<C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>
	<SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>
	<AANLkTimsBQtco2B-s6uXbkrQ7WoiQ0SzF8N5=wSskOML@mail.gmail.com>
	<AANLkTikTSTtaN1cd14U+G8yKA1u=A3dCgtKgSrEkYDb-@mail.gmail.com>
	<SNT130-w1462D55D67F6FED40D4D74FB570@phx.gbl>
	<AANLkTikbix9D=E+eAueVHWqzDM-5_SV=xb8z9_NastDF@mail.gmail.com>
	<SNT130-w3B5414501256848B3E382FB570@phx.gbl>
	<AANLkTik=eus1MrmnpCFRvfDCGpE7WYmrf1QgQpnJ=Vg3@mail.gmail.com>
	<SNT130-w486F7895E102C75B06D2F3FB570@phx.gbl>
Message-ID: <AANLkTi=6GCyB1FX4ma1i8-khYKv0jo+qgy2Jeiq2K_un@mail.gmail.com>

I would just say forget about the IP ACL stuff and go with the
username combination instead. Makes things easier for me, as I would
comment on.

2010/10/14 ? Chiang ?? Chih-Chung   Wybie <tear152 at hotmail.com>:
> Well I can't login pastebin.freeswitch.org
> and that's my configs below
>
> IP address:?60.248.175.37
> Mask?????? :255.255.255.240
> GateWay : 60.248.175.33
> DNS??????? :168.95.1.1
>
> ?(autoload_configs/act.conf.xml)
> ?<list name="domains" default="deny">
> ????? <node type="allow" cidr="60.248.175.32/28"/>
> ????? <!-- <node type="allow" domain="$${domain}"/>-->
> ?</list>
>
> ?(dialplan/default.xml)
> ?<extension name="Dial to 38">
> ?????? <condition field="destination_number"
> ?expression="^([0-4]0[01][0-9])$">
> ?????? <action application="bridge"
> ?data="sofia/internal/$1 at 60.248.175.38"/>
> ?????? </condition>
> ?</extension>
>
> ?(dialplan/public.xml)
>
> ?<extension name="Calls from 38">
> ????? <condition field="destination_number"
> ?expression="^([0-4]0[01][0-9])$">
> ??????? <action application="transfer" data="$1 XML default"/>
> ?????? </condition>
> ??</extension>
>
>
>
>
> IP address:?60.248.175.38
> Mask?????? :255.255.255.240
> GateWay : 60.248.175.33
> DNS??????? :168.95.1.1
>
>
> ?(autoload_configs/act.conf.xml)
> ?<list name="domains" default="deny">
> ?????? <node type="allow" cidr="60.248.175.32/28"/>
> ????? <!-- <node type="allow" domain="$${domain}"/>-->
> ?</list>
>
> ?(dialplan/default.xml)
>
> ?<extension name="Dial to 37">
> ?????? <condition field="destination_number"
> ?expression="^([0-4]0[01][0-9])$">
> ???????? <action application="bridge"
> ?data="sofia/internal/$1 at 60.248.175.37"/>
> ?????? </condition>
> ??</extension>
>
> ?(dialplan/public.xml)
>
> ?<extension name="Calls from 37">
> ?????? <condition field="destination_number"
> ?expression="^([0-4]0[01][0-9])$">
> ???????? <action application="transfer" data="$1 XML default"/>
> ?????? </condition>
> ??</extension>
>
> Thank you for your attension
>
> Best Regards
>
> Gary
>
>
>
> ________________________________
> Date: Thu, 14 Oct 2010 20:02:16 -0700
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> Did you try using ACLs or with putting a user/gateway on each side?
> Put your configs on pastebin.freeswitch.org.
>
> This is actually a simple exercise once you know what to do. Start by
> deciding if you want to do authentication by digest or IP address.
>
> -MC
>
> 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
>
> But my question is
>
> I set two fs
>
> and I want to connect them
>
> I refer to http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
>
> it?'s not working
>
> so I give my configuration already
>
> should I change?some setting?
>
> Thank you for your attension
>
> Best Regards
>
> Gary
>
> ________________________________
> Date: Thu, 14 Oct 2010 17:56:43 -0700
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> Yes! You don't need a gateway for this. If the users are all on the same
> server then you just need to create the necessary files in
> conf/directory/default/
> You already have 1000.xml, 1001.xml, ... 1019.xml
> You just need to add 2010.xml, 4001.xml, 3009.xml, etc.
> Check out this article:
> http://bit.ly/EpVrv
>
> It's a bit older, but the section on adding a new extension is very
> accurate. In that example it shows how to add extension 1500 to your
> directory and your dialplan. You can copy that example for all of your new
> extension numbers.
>
> -MC
>
> 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
>
> Hello
>
> but I can call to each other on the same server
>
> for example
>
> I can use 1007 at 37? call 1013 at 37??, ?4001 at 37??, 2010 at 37 , 3009 at 37
>
>
> ?Thank you for your attension
> Best Regards
> ?Gary
>
>
>
> ________________________________
> Date: Thu, 14 Oct 2010 11:27:31 -0700
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> FYI, we have several resources to help
> http://wiki.freeswitch.org/wiki/Regular_Expression
> http://bit.ly/aijtAC
>
> :)
>
> -MC
>
> On Thu, Oct 14, 2010 at 11:18 AM, Jeffrey Leung <curriegrad2004 at gmail.com>
> wrote:
>
> Dialplan's regex doesn't look right to me. You'll need to fix that
> before you do anything else.
>
> I'd suggest this regex instead of the one you provided:
> ^[0-4]0(10[1-9][0-9])$
>
> instead of this one
> ^([0-4]0[01][0-9])$
>
> If you can't get the grasp of regular expressions, I'd highly suggest
> you to pick up a book and read up on it. Knowing regular expressions
> is a great skill to have ;)
>
> 2010/10/14 ? Chiang ?? Chih-Chung ? Wybie <tear152 at hotmail.com>:
>> I tried use 1013 at 38 calls to 1019 at 37
>>
>> show log below
>>
>> 2010-10-14 17:50:41.249876 [NOTICE] switch_channel.c:675 New Channel
>> sofia/internal/1013 at 60.248.175.38 [875fdeef-b94e-41fb-a621-ea005bbaedbd]
>> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
>> 1013->1019 in context public
>> 2010-10-14 17:50:41.263809 [NOTICE] switch_ivr.c:1450 Transfer
>> sofia/internal/1013 at 60.248.175.38 to XML[1019 at default]
>> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
>> 1013->1019 in context default
>> 2010-10-14 17:50:41.283904 [INFO] switch_core_session.c:1764 Sending early
>> media
>> 2010-10-14 17:50:41.285023 [NOTICE] mod_sofia.c:1994 Pre-Answer
>> sofia/internal/1013 at 60.248.175.38!
>>
>> seems not work
>>
>> Thank you for you attension
>> Best Regards
>>
>> Gary
>>
>>
>>
>> ________________________________
>> From: david.ponzone at ipeva.fr
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Thu, 14 Oct 2010 10:16:28 +0200
>> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>>
>> I think we would need a complete log from the target FS, in order to
>> provide
>> you with any efficient help.
>> Thank
>> David Ponzone ?Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel: ? ? ?01 74 03 18 97
>> gsm: ? 06 66 98 76 34
>> Service Client?IPeva
>> tel: ? ? ?0811 46 26 26
>> www.ipeva.fr? -? ?www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message
>> s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>> Le 14/10/2010 ? 09:55, ? Chiang ?? Chih-Chung Wybie a ?crit :
>>
>> I tried Connect Two FreeSWITCH
>> Boxes??http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
>> and my configuration :
>>
>> 60.248.175.37
>>
>> (autoload_configs/act.conf.xml)
>> <list name="domains" default="deny">
>> ????? <node type="allow" cidr="60.248.175.38/28"/>
>> ???? <!-- <node type="allow" domain="$${domain}"/>-->
>> </list>
>>
>> (dialplan/default.xml)
>>
>> <extension name="Dial to 38">
>> ????? <condition field="destination_number"
>> expression="^([0-4]0[01][0-9])$">
>> ??????? <action application="bridge"
>> data="sofia/internal/$1 at 60.248.175.38"/>
>> ????? </condition>
>> ?</extension>
>>
>> (dialplan/public.xml)
>>
>> <extension name="Calls from 38">
>> ????? <condition field="destination_number"
>> expression="^([0-4]0[01][0-9])$">
>> ??????? <action application="transfer" data="$1 XML default"/>
>> ????? </condition>
>> ?</extension>
>>
>>
>>
>> 60.248.175.38
>>
>> (autoload_configs/act.conf.xml)
>> <list name="domains" default="deny">
>> ????? <node type="allow" cidr="60.248.175.37/28"/>
>> ???? <!-- <node type="allow" domain="$${domain}"/>-->
>> </list>
>>
>> (dialplan/default.xml)
>>
>> <extension name="Dial to 37">
>> ????? <condition field="destination_number"
>> expression="^([0-4]0[01][0-9])$">
>> ??????? <action application="bridge"
>> data="sofia/internal/$1 at 60.248.175.37"/>
>> ????? </condition>
>> ?</extension>
>>
>> (dialplan/public.xml)
>>
>> <extension name="Calls from 37">
>> ????? <condition field="destination_number"
>> expression="^([0-4]0[01][0-9])$">
>> ??????? <action application="transfer" data="$1 XML default"/>
>> ????? </condition>
>> ?</extension>
>>
>> when I use?1007 at 38?calls to?1007 at 37
>> it won't show errors but still not work
>>
>> how should I change something ?
>>
>> Thank you for your attension
>>
>> Best Regards
>> Gary
>>
>>
>>
>>
>>
>>> Date: Wed, 13 Oct 2010 22:17:13 -0700
>>> From:?curriegrad2004 at gmail.com
>>> To:?freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>>>
>>> You can create a SIP trunk between the two servers so calls can be
>>> routed back and forth from the servers you've specified.
>>>
>>> 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
>>> > I tried it
>>> > but not working
>>> > and show errors below
>>> > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493 Cannot
>>> > create
>>> > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
>>> > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493 Cannot
>>> > create
>>> > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
>>> > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate Failed.
>>> > Cause: USER_NOT_REGISTERED
>>> >
>>> > and for example
>>> > ?I use SIP server 60.248.175.38? user id=1007
>>> > and I call to?1007 at 60.248.175.37??? that will show these errors
>>> >
>>> > how should set more?
>>> >
>>> > thank you for your attension
>>> >
>>> > Best Regards
>>> > Gary
>>> >
>>> > ________________________________
>>> > Date: Thu, 14 Oct 2010 10:55:49 +0800
>>> > From:?xyangni at gmail.com
>>> > To:?freeswitch-users at lists.freeswitch.org
>>> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>>> >
>>> > I think you can reg at server A and than call B as a normal SIP call to
>>> > external domain.
>>> >
>>> > 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
>>> >
>>> >
>>> >
>>> > Hello
>>> >
>>> > I set two SIP servers ?60.248.175.37 & 60.248.175.38 and use X-Lite to
>>> > call
>>> >
>>> > my question is how can I call between these servers?
>>> >
>>> > if I can , how shoould I set in *.xml? or X-Lite?
>>> >
>>> > thank you for your attension
>>> >
>>> >
>>> > Best Regards
>>> > Gary
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> >?FreeSWITCH-users at lists.freeswitch.org
>>> >?http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >?http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> > _______________________________________________ FreeSWITCH-users
>>> > mailing
>>> > list?FreeSWITCH-users at lists.freeswitch.org
>>> >?http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >?http://www.freeswitch.org
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> >?FreeSWITCH-users at lists.freeswitch.org
>>> >?http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >?http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>?FreeSWITCH-users at lists.freeswitch.org
>>>?http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>?http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________ FreeSWITCH-users mailing
>> list FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From daniel.neubert at solomo.de  Thu Oct 14 23:12:25 2010
From: daniel.neubert at solomo.de (Daniel Neubert)
Date: Fri, 15 Oct 2010 08:12:25 +0200
Subject: [Freeswitch-users] mod_cepstral 5.1 known issues?
In-Reply-To: <AANLkTimj=SwX7d1sot+NhOxMgD4F-PN=bM3hxrh3bF3O@mail.gmail.com>
References: <AANLkTik8KZdxAnUUfv0nKF6EkTHe559PtC2bF74eng11@mail.gmail.com>	<AANLkTi=MvWh14MQ9wPEboXZmMgAtURAY5zBoobn32Pkq@mail.gmail.com>
	<AANLkTimj=SwX7d1sot+NhOxMgD4F-PN=bM3hxrh3bF3O@mail.gmail.com>
Message-ID: <4CB7F0C9.2040708@solomo.de>

I've exactly this setup which is in productive use for a few months now.

Using cepstral directly on our FreeSWITCH nodes did not work (since they 
are running on FreeBSD and Cepstral is only available for GNU Linux).

Current setup is using Voice Katrin (German) Version 5.1.0 on Gentoo 
Linux 64Bit via UniMRCP Server 1.0.0.


Best regards / Mit freundlichen Gr??en,
Daniel Neubert


On 14.10.2010 17:58, Marc Kellem wrote:
> Is anyone using mod_unimrcp to access Cepstral 5.1 voices in a UniMRCP 
> server?
>
> On Thu, Oct 14, 2010 at 11:26 AM, Anthony Minessale 
> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
>
>     random deadlocks in the cepstral engine blocking all the speech
>     generation calls.
>
>
>     On Thu, Oct 14, 2010 at 9:56 AM, Marc Kellem <mkellem at vontoo.com
>     <mailto:mkellem at vontoo.com>> wrote:
>     > The mod_cepstral wiki page says "Warning: the suggested version
>     to use
>     > is 4.x since there are known issues with 5.1 (which is closed
>     source)".
>     >  What exactly are the known issues?
>     > http://wiki.freeswitch.org/wiki/Mod_cepstral
>     > Thanks,
>     > Marc Kellem
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > 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
>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
>     FreeSWITCH Developer Conference
>     sip:888 at conference.freeswitch.org
>     <mailto:sip%3A888 at conference.freeswitch.org>
>     googletalk:conf+888 at conference.freeswitch.org
>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>     pstn:+19193869900
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
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From tear152 at hotmail.com  Fri Oct 15 00:28:26 2010
From: tear152 at hotmail.com (=?gb2312?B?va0gQ2hpYW5nINbB1tAgQ2hpaC1DaHVuZyAgIFd5Ymll?=)
Date: Fri, 15 Oct 2010 15:28:26 +0800
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <AANLkTi=6GCyB1FX4ma1i8-khYKv0jo+qgy2Jeiq2K_un@mail.gmail.com>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>,
	<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>,
	<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>,
	<SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>,
	<AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>,
	<SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>,
	<C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>,
	<SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>,
	<AANLkTimsBQtco2B-s6uXbkrQ7WoiQ0SzF8N5=wSskOML@mail.gmail.com>,
	<AANLkTikTSTtaN1cd14U+G8yKA1u=A3dCgtKgSrEkYDb-@mail.gmail.com>,
	<SNT130-w1462D55D67F6FED40D4D74FB570@phx.gbl>,
	<AANLkTikbix9D=E+eAueVHWqzDM-5_SV=xb8z9_NastDF@mail.gmail.com>,
	<SNT130-w3B5414501256848B3E382FB570@phx.gbl>,
	<AANLkTik=eus1MrmnpCFRvfDCGpE7WYmrf1QgQpnJ=Vg3@mail.gmail.com>,
	<SNT130-w486F7895E102C75B06D2F3FB570@phx.gbl>,
	<AANLkTi=6GCyB1FX4ma1i8-khYKv0jo+qgy2Jeiq2K_un@mail.gmail.com>
Message-ID: <SNT130-w63D50A8E3566B682F71111FB570@phx.gbl>


 
But when I changed configs to your suggestions
 
I can't call anyone on the same server , before changed it  I can call
 
 

 
> Date: Thu, 14 Oct 2010 21:22:19 -0700
> From: curriegrad2004 at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> 
> I would just say forget about the IP ACL stuff and go with the
> username combination instead. Makes things easier for me, as I would
> comment on.
> 
> 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
> > Well I can't login pastebin.freeswitch.org
> > and that's my configs below
> >
> > IP address: 60.248.175.37
> > Mask       :255.255.255.240
> > GateWay : 60.248.175.33
> > DNS        :168.95.1.1
> >
> >  (autoload_configs/act.conf.xml)
> >  <list name="domains" default="deny">
> >       <node type="allow" cidr="60.248.175.32/28"/>
> >       <!-- <node type="allow" domain="$${domain}"/>-->
> >  </list>
> >
> >  (dialplan/default.xml)
> >  <extension name="Dial to 38">
> >        <condition field="destination_number"
> >  expression="^([0-4]0[01][0-9])$">
> >        <action application="bridge"
> >  data="sofia/internal/$1 at 60.248.175.38"/>
> >        </condition>
> >  </extension>
> >
> >  (dialplan/public.xml)
> >
> >  <extension name="Calls from 38">
> >       <condition field="destination_number"
> >  expression="^([0-4]0[01][0-9])$">
> >         <action application="transfer" data="$1 XML default"/>
> >        </condition>
> >   </extension>
> >
> >
> >
> >
> > IP address: 60.248.175.38
> > Mask       :255.255.255.240
> > GateWay : 60.248.175.33
> > DNS        :168.95.1.1
> >
> >
> >  (autoload_configs/act.conf.xml)
> >  <list name="domains" default="deny">
> >        <node type="allow" cidr="60.248.175.32/28"/>
> >       <!-- <node type="allow" domain="$${domain}"/>-->
> >  </list>
> >
> >  (dialplan/default.xml)
> >
> >  <extension name="Dial to 37">
> >        <condition field="destination_number"
> >  expression="^([0-4]0[01][0-9])$">
> >          <action application="bridge"
> >  data="sofia/internal/$1 at 60.248.175.37"/>
> >        </condition>
> >   </extension>
> >
> >  (dialplan/public.xml)
> >
> >  <extension name="Calls from 37">
> >        <condition field="destination_number"
> >  expression="^([0-4]0[01][0-9])$">
> >          <action application="transfer" data="$1 XML default"/>
> >        </condition>
> >   </extension>
> >
> > Thank you for your attension
> >
> > Best Regards
> >
> > Gary
> >
> >
> >
> > ________________________________
> > Date: Thu, 14 Oct 2010 20:02:16 -0700
> > From: msc at freeswitch.org
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >
> > Did you try using ACLs or with putting a user/gateway on each side?
> > Put your configs on pastebin.freeswitch.org.
> >
> > This is actually a simple exercise once you know what to do. Start by
> > deciding if you want to do authentication by digest or IP address.
> >
> > -MC
> >
> > 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
> >
> > But my question is
> >
> > I set two fs
> >
> > and I want to connect them
> >
> > I refer to http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
> >
> > it 's not working
> >
> > so I give my configuration already
> >
> > should I change some setting?
> >
> > Thank you for your attension
> >
> > Best Regards
> >
> > Gary
> >
> > ________________________________
> > Date: Thu, 14 Oct 2010 17:56:43 -0700
> > From: msc at freeswitch.org
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >
> > Yes! You don't need a gateway for this. If the users are all on the same
> > server then you just need to create the necessary files in
> > conf/directory/default/
> > You already have 1000.xml, 1001.xml, ... 1019.xml
> > You just need to add 2010.xml, 4001.xml, 3009.xml, etc.
> > Check out this article:
> > http://bit.ly/EpVrv
> >
> > It's a bit older, but the section on adding a new extension is very
> > accurate. In that example it shows how to add extension 1500 to your
> > directory and your dialplan. You can copy that example for all of your new
> > extension numbers.
> >
> > -MC
> >
> > 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
> >
> > Hello
> >
> > but I can call to each other on the same server
> >
> > for example
> >
> > I can use 1007 at 37  call 1013 at 37  ,  4001 at 37  , 2010 at 37 , 3009 at 37
> >
> >
> >  Thank you for your attension
> > Best Regards
> >  Gary
> >
> >
> >
> > ________________________________
> > Date: Thu, 14 Oct 2010 11:27:31 -0700
> > From: msc at freeswitch.org
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >
> > FYI, we have several resources to help
> > http://wiki.freeswitch.org/wiki/Regular_Expression
> > http://bit.ly/aijtAC
> >
> > :)
> >
> > -MC
> >
> > On Thu, Oct 14, 2010 at 11:18 AM, Jeffrey Leung <curriegrad2004 at gmail.com>
> > wrote:
> >
> > Dialplan's regex doesn't look right to me. You'll need to fix that
> > before you do anything else.
> >
> > I'd suggest this regex instead of the one you provided:
> > ^[0-4]0(10[1-9][0-9])$
> >
> > instead of this one
> > ^([0-4]0[01][0-9])$
> >
> > If you can't get the grasp of regular expressions, I'd highly suggest
> > you to pick up a book and read up on it. Knowing regular expressions
> > is a great skill to have ;)
> >
> > 2010/10/14 ? Chiang ?? Chih-Chung   Wybie <tear152 at hotmail.com>:
> >> I tried use 1013 at 38 calls to 1019 at 37
> >>
> >> show log below
> >>
> >> 2010-10-14 17:50:41.249876 [NOTICE] switch_channel.c:675 New Channel
> >> sofia/internal/1013 at 60.248.175.38 [875fdeef-b94e-41fb-a621-ea005bbaedbd]
> >> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
> >> 1013->1019 in context public
> >> 2010-10-14 17:50:41.263809 [NOTICE] switch_ivr.c:1450 Transfer
> >> sofia/internal/1013 at 60.248.175.38 to XML[1019 at default]
> >> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
> >> 1013->1019 in context default
> >> 2010-10-14 17:50:41.283904 [INFO] switch_core_session.c:1764 Sending early
> >> media
> >> 2010-10-14 17:50:41.285023 [NOTICE] mod_sofia.c:1994 Pre-Answer
> >> sofia/internal/1013 at 60.248.175.38!
> >>
> >> seems not work
> >>
> >> Thank you for you attension
> >> Best Regards
> >>
> >> Gary
> >>
> >>
> >>
> >> ________________________________
> >> From: david.ponzone at ipeva.fr
> >> To: freeswitch-users at lists.freeswitch.org
> >> Date: Thu, 14 Oct 2010 10:16:28 +0200
> >> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >>
> >> I think we would need a complete log from the target FS, in order to
> >> provide
> >> you with any efficient help.
> >> Thank
> >> David Ponzone  Direction Technique
> >> email: david.ponzone at ipeva.fr
> >> tel:      01 74 03 18 97
> >> gsm:   06 66 98 76 34
> >> Service Client IPeva
> >> tel:      0811 46 26 26
> >> www.ipeva.fr  -   www.ipeva-studio.com
> >> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> >> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> >> non autoris?e est interdite. Tout message ?lectronique est susceptible
> >> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> >> s'il
> >> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> >> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> >>
> >>
> >>
> >> Le 14/10/2010 ? 09:55, ? Chiang ?? Chih-Chung Wybie a ?crit :
> >>
> >> I tried Connect Two FreeSWITCH
> >> Boxes  http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
> >> and my configuration :
> >>
> >> 60.248.175.37
> >>
> >> (autoload_configs/act.conf.xml)
> >> <list name="domains" default="deny">
> >>       <node type="allow" cidr="60.248.175.38/28"/>
> >>      <!-- <node type="allow" domain="$${domain}"/>-->
> >> </list>
> >>
> >> (dialplan/default.xml)
> >>
> >> <extension name="Dial to 38">
> >>       <condition field="destination_number"
> >> expression="^([0-4]0[01][0-9])$">
> >>         <action application="bridge"
> >> data="sofia/internal/$1 at 60.248.175.38"/>
> >>       </condition>
> >>  </extension>
> >>
> >> (dialplan/public.xml)
> >>
> >> <extension name="Calls from 38">
> >>       <condition field="destination_number"
> >> expression="^([0-4]0[01][0-9])$">
> >>         <action application="transfer" data="$1 XML default"/>
> >>       </condition>
> >>  </extension>
> >>
> >>
> >>
> >> 60.248.175.38
> >>
> >> (autoload_configs/act.conf.xml)
> >> <list name="domains" default="deny">
> >>       <node type="allow" cidr="60.248.175.37/28"/>
> >>      <!-- <node type="allow" domain="$${domain}"/>-->
> >> </list>
> >>
> >> (dialplan/default.xml)
> >>
> >> <extension name="Dial to 37">
> >>       <condition field="destination_number"
> >> expression="^([0-4]0[01][0-9])$">
> >>         <action application="bridge"
> >> data="sofia/internal/$1 at 60.248.175.37"/>
> >>       </condition>
> >>  </extension>
> >>
> >> (dialplan/public.xml)
> >>
> >> <extension name="Calls from 37">
> >>       <condition field="destination_number"
> >> expression="^([0-4]0[01][0-9])$">
> >>         <action application="transfer" data="$1 XML default"/>
> >>       </condition>
> >>  </extension>
> >>
> >> when I use 1007 at 38 calls to 1007 at 37
> >> it won't show errors but still not work
> >>
> >> how should I change something ?
> >>
> >> Thank you for your attension
> >>
> >> Best Regards
> >> Gary
> >>
> >>
> >>
> >>
> >>
> >>> Date: Wed, 13 Oct 2010 22:17:13 -0700
> >>> From: curriegrad2004 at gmail.com
> >>> To: freeswitch-users at lists.freeswitch.org
> >>> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >>>
> >>> You can create a SIP trunk between the two servers so calls can be
> >>> routed back and forth from the servers you've specified.
> >>>
> >>> 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
> >>> > I tried it
> >>> > but not working
> >>> > and show errors below
> >>> > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493 Cannot
> >>> > create
> >>> > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> >>> > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493 Cannot
> >>> > create
> >>> > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> >>> > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate Failed.
> >>> > Cause: USER_NOT_REGISTERED
> >>> >
> >>> > and for example
> >>> >  I use SIP server 60.248.175.38  user id=1007
> >>> > and I call to 1007 at 60.248.175.37    that will show these errors
> >>> >
> >>> > how should set more?
> >>> >
> >>> > thank you for your attension
> >>> >
> >>> > Best Regards
> >>> > Gary
> >>> >
> >>> > ________________________________
> >>> > Date: Thu, 14 Oct 2010 10:55:49 +0800
> >>> > From: xyangni at gmail.com
> >>> > To: freeswitch-users at lists.freeswitch.org
> >>> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >>> >
> >>> > I think you can reg at server A and than call B as a normal SIP call to
> >>> > external domain.
> >>> >
> >>> > 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
> >>> >
> >>> >
> >>> >
> >>> > Hello
> >>> >
> >>> > I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X-Lite to
> >>> > call
> >>> >
> >>> > my question is how can I call between these servers?
> >>> >
> >>> > if I can , how shoould I set in *.xml? or X-Lite?
> >>> >
> >>> > thank you for your attension
> >>> >
> >>> >
> >>> > Best Regards
> >>> > Gary
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>> >
> >>> > _______________________________________________ FreeSWITCH-users
> >>> > mailing
> >>> > list FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________ FreeSWITCH-users mailing
> >> list FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________ FreeSWITCH-users mailing
> > list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________ FreeSWITCH-users mailing
> > list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________ FreeSWITCH-users mailing
> > list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
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From tear152 at hotmail.com  Fri Oct 15 01:52:07 2010
From: tear152 at hotmail.com (=?gb2312?B?va0gQ2hpYW5nINbB1tAgQ2hpaC1DaHVuZyAgIFd5Ymll?=)
Date: Fri, 15 Oct 2010 16:52:07 +0800
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <6A8D7D0E-4DF5-4F35-8A00-F52FECC4F826@ipeva.fr>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>,
	,,<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>,
	,,<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>,
	,,<SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>,
	, , <AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>, ,
	<SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>, ,
	<C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>,
	<SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>,
	<6A8D7D0E-4DF5-4F35-8A00-F52FECC4F826@ipeva.fr>
Message-ID: <SNT130-w6005F900786BDFBE48FF43FB570@phx.gbl>


Hello this is my debug log
 
I use 1013 at 60.248.175.38 call to 4001 at 60.248.175.37
 
EXECUTE sofia/internal/1013 at 60.248.175.38 transfer(4001 XML default)
2010-10-15 16:49:30.666644 [DEBUG] switch_ivr.c:1444 (sofia/internal/1013 at 60.248.175.38) State Change CS_EXECUTE -> CS_ROUTING
2010-10-15 16:49:30.666644 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:30.666644 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:30.666644 [NOTICE] switch_ivr.c:1450 Transfer sofia/internal/1013 at 60.248.175.38 to XML[4001 at default]
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38) State EXECUTE going to sleep
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_ROUTING
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38) State ROUTING
2010-10-15 16:49:30.666644 [DEBUG] mod_sofia.c:140 sofia/internal/1013 at 60.248.175.38 SOFIA ROUTING
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1013 at 60.248.175.38 Standard ROUTING
2010-10-15 16:49:30.666644 [INFO] mod_dialplan_xml.c:331 Processing 1013->4001 in context default
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->unloop] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->tod_example] continue=true
Dialplan: sofia/internal/1013 at 60.248.175.38 Date/Time Match (PASS) [tod_example] break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(open=true)
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->holiday_example] continue=true
Dialplan: sofia/internal/1013 at 60.248.175.38 Date/Time Match (FAIL) [holiday_example] break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->global-intercept] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [global-intercept] destination_number(4001) =~ /^886$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->group-intercept] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [group-intercept] destination_number(4001) =~ /^\*8$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->intercept-ext] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [intercept-ext] destination_number(4001) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->redial] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [redial] destination_number(4001) =~ /^(redial|870)$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->global] continue=true
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: sofia/internal/1013 at 60.248.175.38 Absolute Condition [global]
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-last_dial/global/${uuid})
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->snom-demo-2] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [snom-demo-2] destination_number(4001) =~ /^9001$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->snom-demo-1] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [snom-demo-1] destination_number(4001) =~ /^9000$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->eavesdrop] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [eavesdrop] destination_number(4001) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->eavesdrop] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [eavesdrop] destination_number(4001) =~ /^779$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->call_return] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call_return] destination_number(4001) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->del-group] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [del-group] destination_number(4001) =~ /^80(\d{2})$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->add-group] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [add-group] destination_number(4001) =~ /^81(\d{2})$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->call-group-simo] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call-group-simo] destination_number(4001) =~ /^82(\d{2})$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->call-group-order] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call-group-order] destination_number(4001) =~ /^83(\d{2})$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->extension-intercom] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [extension-intercom] destination_number(4001) =~ /^8(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->Local_Extension] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (PASS) [Local_Extension] destination_number(4001) =~ /^([0-4]0[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_COPYRIGHT=(c) 2010)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_SOFTWARE=FreeSwitch)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_ARTIST=FreeSwitch)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_COMMENT=FreeSwitch)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_DATE=${strftime(%Y-%m-%d %H:%M)})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_STEREO=true)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(dialed_extension=4001)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action export(dialed_extension=4001)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set_user(${dialed_extension}@${domain_name})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action limit(60.248.175.38 ${dialed_extension} ${max_calls} ${fail_over})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action bind_meta_app(1 b s execute_extension::dx XML features)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action bind_meta_app(3 b s execute_extension::cf XML features)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(ringback=${us-ring})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(transfer_ringback=local_stream://moh)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(call_timeout=30)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(hangup_after_bridge=true)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(continue_on_fail=true)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action bridge(user/${dialed_extension}@${domain_name})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action answer()
Dialplan: sofia/internal/1013 at 60.248.175.38 Action sleep(1000)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension})
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1013 at 60.248.175.38) State Change CS_ROUTING -> CS_EXECUTE
2010-10-15 16:49:30.666644 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38) State ROUTING going to sleep
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_EXECUTE
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38) State EXECUTE
2010-10-15 16:49:30.666644 [DEBUG] mod_sofia.c:233 sofia/internal/1013 at 60.248.175.38 SOFIA EXECUTE
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1013 at 60.248.175.38 Standard EXECUTE
EXECUTE sofia/internal/1013 at 60.248.175.38 set(open=true)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [open]=[true]
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-spymap/1013/fe66f80a-992b-43f7-b130-47f0c87f4c15)
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-last_dial/1013/4001)
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-last_dial/global/fe66f80a-992b-43f7-b130-47f0c87f4c15)
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_COPYRIGHT=(c) 2010)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_COPYRIGHT]=[(c) 2010]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_SOFTWARE=FreeSwitch)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_SOFTWARE]=[FreeSwitch]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_ARTIST=FreeSwitch)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_ARTIST]=[FreeSwitch]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_COMMENT=FreeSwitch)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_COMMENT]=[FreeSwitch]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_DATE=2010-10-15 16:49)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_DATE]=[2010-10-15 16:49]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_STEREO=true)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_STEREO]=[true]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(dialed_extension=4001)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [dialed_extension]=[4001]
EXECUTE sofia/internal/1013 at 60.248.175.38 export(dialed_extension=4001)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:901 EXPORT [dialed_extension]=[4001]
EXECUTE sofia/internal/1013 at 60.248.175.38 set_user(4001 at 60.248.175.38)
EXECUTE sofia/internal/1013 at 60.248.175.38 limit(60.248.175.38 4001  )
2010-10-15 16:49:30.666644 [WARNING] mod_limit.c:779 USAGE: limit <realm> <id> [<max> [number  [dialplan [context]]]]
EXECUTE sofia/internal/1013 at 60.248.175.38 bind_meta_app(1 b s execute_extension::dx XML features)
2010-10-15 16:49:30.666644 [INFO] switch_ivr_async.c:2372 Bound B-Leg: 1 execute_extension::dx XML features
EXECUTE sofia/internal/1013 at 60.248.175.38 bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/1013.2010-10-15-16-49-30.wav)
2010-10-15 16:49:30.666644 [INFO] switch_ivr_async.c:2372 Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/1013.2010-10-15-16-49-30.wav
EXECUTE sofia/internal/1013 at 60.248.175.38 bind_meta_app(3 b s execute_extension::cf XML features)
2010-10-15 16:49:30.666644 [INFO] switch_ivr_async.c:2372 Bound B-Leg: 3 execute_extension::cf XML features
EXECUTE sofia/internal/1013 at 60.248.175.38 set(ringback=%(2000,4000,440.0,480.0))
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [ringback]=[%(2000,4000,440.0,480.0)]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(transfer_ringback=local_stream://moh)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [transfer_ringback]=[local_stream://moh]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(call_timeout=30)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [call_timeout]=[30]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(hangup_after_bridge=true)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(continue_on_fail=true)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [continue_on_fail]=[true]
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-call_return/4001/1013)
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-last_dial_ext/4001/fe66f80a-992b-43f7-b130-47f0c87f4c15)
EXECUTE sofia/internal/1013 at 60.248.175.38 set(called_party_callgroup=techsupport)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [called_party_callgroup]=[techsupport]
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-last_dial/techsupport/fe66f80a-992b-43f7-b130-47f0c87f4c15)
EXECUTE sofia/internal/1013 at 60.248.175.38 bridge(user/4001 at 60.248.175.38)
2010-10-15 16:49:30.666644 [DEBUG] switch_ivr_originate.c:1946 variable string 0 = [presence_id=4001 at 60.248.175.38]
2010-10-15 16:49:30.666644 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
2010-10-15 16:49:30.666644 [DEBUG] switch_ivr_originate.c:3308 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2010-10-15 16:49:30.666644 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
2010-10-15 16:49:30.666644 [DEBUG] switch_ivr_originate.c:3308 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2010-10-15 16:49:30.666644 [INFO] mod_dptools.c:2356 Originate Failed.  Cause: USER_NOT_REGISTERED
EXECUTE sofia/internal/1013 at 60.248.175.38 answer()
2010-10-15 16:49:30.694130 [DEBUG] sofia_glue.c:2616 AUDIO RTP [sofia/internal/1013 at 60.248.175.38] 60.248.175.38 port 25606 -> 60.248.175.43 port 35070 codec: 0 ms: 20
2010-10-15 16:49:30.694130 [DEBUG] switch_rtp.c:1343 Starting timer [soft] 160 bytes per 20ms
2010-10-15 16:49:30.695467 [DEBUG] sofia_glue.c:2822 Set 2833 dtmf send payload to 101
2010-10-15 16:49:30.695467 [DEBUG] sofia_glue.c:2827 Set 2833 dtmf receive payload to 101
2010-10-15 16:49:30.695467 [DEBUG] mod_sofia.c:663 Local SDP sofia/internal/1013 at 60.248.175.38:
v=0
o=FreeSWITCH 1287106964 1287106965 IN IP4 60.248.175.38
s=FreeSWITCH
c=IN IP4 60.248.175.38
t=0 0
m=audio 25606 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
2010-10-15 16:49:30.695467 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:30.695467 [NOTICE] mod_dptools.c:720 Channel [sofia/internal/1013 at 60.248.175.38] has been answered
EXECUTE sofia/internal/1013 at 60.248.175.38 sleep(1000)
2010-10-15 16:49:30.698172 [DEBUG] sofia.c:4200 Channel sofia/internal/1013 at 60.248.175.38 entering state [completed][200]
2010-10-15 16:49:30.745378 [DEBUG] switch_rtp.c:2443 Correct ip/port confirmed.
2010-10-15 16:49:30.807388 [DEBUG] sofia.c:4200 Channel sofia/internal/1013 at 60.248.175.38 entering state [ready][200]
EXECUTE sofia/internal/1013 at 60.248.175.38 bridge(loopback/app=voicemail:default 60.248.175.38 4001)
2010-10-15 16:49:31.704837 [NOTICE] switch_channel.c:675 New Channel loopback/app=voicemail:default 60.248.175.38 4001-a [23f3d5de-f985-44bd-8b1a-118b98bb5eae]
2010-10-15 16:49:31.704837 [DEBUG] mod_loopback.c:127 loopback/app=voicemail:default 60.248.175.38 4001-a setup codec PCMU/8000/20
2010-10-15 16:49:31.704837 [NOTICE] switch_channel.c:673 Rename Channel loopback/app=voicemail:default 60.248.175.38 4001-a->loopback/voicemail-a [23f3d5de-f985-44bd-8b1a-118b98bb5eae]
2010-10-15 16:49:31.704837 [DEBUG] mod_loopback.c:911 (loopback/voicemail-a) State Change CS_NEW -> CS_INIT
2010-10-15 16:49:31.704837 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.704837 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.706218 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-a) Running State Change CS_INIT
2010-10-15 16:49:31.706218 [DEBUG] switch_core_state_machine.c:338 (loopback/voicemail-a) State INIT
2010-10-15 16:49:31.706218 [NOTICE] switch_channel.c:675 New Channel loopback/voicemail-b [1abc53a2-df93-40db-96ab-547f9ddd1b1d]
2010-10-15 16:49:31.706218 [DEBUG] mod_loopback.c:127 loopback/voicemail-b setup codec PCMU/8000/20
2010-10-15 16:49:31.706218 [DEBUG] mod_loopback.c:240 (loopback/voicemail-b) State Change CS_NEW -> CS_INIT
2010-10-15 16:49:31.706218 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-b [BREAK]
2010-10-15 16:49:31.706218 [DEBUG] mod_loopback.c:452 loopback/voicemail-b CHANNEL KILL
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-b) Running State Change CS_INIT
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:338 (loopback/voicemail-b) State INIT
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:281 (loopback/voicemail-b) State Change CS_INIT -> CS_ROUTING
2010-10-15 16:49:31.707132 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-b [BREAK]
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:452 loopback/voicemail-b CHANNEL KILL
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:338 (loopback/voicemail-b) State INIT going to sleep
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-b) Running State Change CS_ROUTING
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:341 (loopback/voicemail-b) State ROUTING
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:315 loopback/voicemail-b CHANNEL ROUTING
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:326 (loopback/voicemail-b) State Change CS_ROUTING -> CS_EXECUTE
2010-10-15 16:49:31.707132 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-b [BREAK]
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:452 loopback/voicemail-b CHANNEL KILL
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:341 (loopback/voicemail-b) State ROUTING going to sleep
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-b) Running State Change CS_EXECUTE
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:348 (loopback/voicemail-b) State EXECUTE
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:346 loopback/voicemail-b CHANNEL EXECUTE
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:157 loopback/voicemail-b Standard EXECUTE
EXECUTE loopback/voicemail-b pre_answer()
2010-10-15 16:49:31.707132 [NOTICE] mod_loopback.c:716 Pre-Answer loopback/voicemail-a!
2010-10-15 16:49:31.707132 [DEBUG] switch_channel.c:2254 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:31.707132 [DEBUG] switch_core_session.c:642 Send signal loopback/voicemail-b [BREAK]
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:452 loopback/voicemail-b CHANNEL KILL
2010-10-15 16:49:31.707132 [NOTICE] mod_dptools.c:746 Pre-Answer loopback/voicemail-b!
2010-10-15 16:49:31.707132 [DEBUG] switch_channel.c:2254 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
EXECUTE loopback/voicemail-b voicemail(default 60.248.175.38 4001)
2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:281 (loopback/voicemail-a) State Change CS_INIT -> CS_ROUTING
2010-10-15 16:49:31.708787 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:338 (loopback/voicemail-a) State INIT going to sleep
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-a) Running State Change CS_ROUTING
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:341 (loopback/voicemail-a) State ROUTING
2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:315 loopback/voicemail-a CHANNEL ROUTING
2010-10-15 16:49:31.708787 [DEBUG] switch_ivr_originate.c:66 (loopback/voicemail-a) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2010-10-15 16:49:31.708787 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:341 (loopback/voicemail-a) State ROUTING going to sleep
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-a) Running State Change CS_CONSUME_MEDIA
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:360 (loopback/voicemail-a) State CONSUME_MEDIA
2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:512 CHANNEL CONSUME_MEDIA
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:360 (loopback/voicemail-a) State CONSUME_MEDIA going to sleep
2010-10-15 16:49:31.710698 [DEBUG] switch_ivr_originate.c:3228 Originate Resulted in Success: [loopback/voicemail-a]
2010-10-15 16:49:31.710698 [DEBUG] switch_core_session.c:642 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.710698 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.710698 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:31.710698 [DEBUG] switch_ivr_bridge.c:1182 (loopback/voicemail-a) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2010-10-15 16:49:31.710698 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.710698 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.710698 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-a) Running State Change CS_EXCHANGE_MEDIA
2010-10-15 16:49:31.710698 [DEBUG] switch_core_state_machine.c:351 (loopback/voicemail-a) State EXCHANGE_MEDIA
2010-10-15 16:49:31.710698 [DEBUG] mod_loopback.c:474 CHANNEL LOOPBACK
2010-10-15 16:49:31.712661 [DEBUG] switch_core_session.c:703 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.712661 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.712661 [DEBUG] switch_core_session.c:703 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:31.824953 [DEBUG] switch_ivr_play_say.c:63 No language specified - Using [en]
2010-10-15 16:49:31.824953 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-person.wav] (en:en)
2010-10-15 16:49:31.824953 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16 at 8000hz 1 channels 20ms
2010-10-15 16:49:33.185370 [DEBUG] switch_ivr_play_say.c:1444 done playing file
2010-10-15 16:49:33.305429 [DEBUG] switch_ivr_play_say.c:244 Handle say:[4001] (en:en)
2010-10-15 16:49:33.305429 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16 at 8000hz 1 channels 20ms
2010-10-15 16:49:33.744961 [DEBUG] switch_ivr_play_say.c:1444 done playing file
2010-10-15 16:49:33.744961 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16 at 8000hz 1 channels 20ms

 


From: david.ponzone at ipeva.fr
To: freeswitch-users at lists.freeswitch.org
Date: Thu, 14 Oct 2010 12:20:56 +0200
Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite

Don't be shy on the logs, send the full ones!


in fs_cli:
fsctl loglevel debug
/log 7


and then make a call





David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34


Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr  -   www.ipeva-studio.com



Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 14/10/2010 ? 11:53, ? Chiang ?? Chih-Chung Wybie a ?crit :

I tried use 1013 at 38 calls to 1019 at 37
 
show log below
 
2010-10-14 17:50:41.249876 [NOTICE] switch_channel.c:675 New Channel sofia/internal/1013 at 60.248.175.38[875fdeef-b94e-41fb-a621-ea005bbaedbd]
2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing 1013->1019 in context public
2010-10-14 17:50:41.263809 [NOTICE] switch_ivr.c:1450 Transfer sofia/internal/1013 at 60.248.175.38 to XML[1019 at default]
2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing 1013->1019 in context default
2010-10-14 17:50:41.283904 [INFO] switch_core_session.c:1764 Sending early media
2010-10-14 17:50:41.285023 [NOTICE] mod_sofia.c:1994 Pre-Answer sofia/internal/1013 at 60.248.175.38!

seems not work
 
Thank you for you attension
Best Regards
 
Gary
 

 


From: david.ponzone at ipeva.fr
To: freeswitch-users at lists.freeswitch.org
Date: Thu, 14 Oct 2010 10:16:28 +0200
Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite

I think we would need a complete log from the target FS, in order to provide you with any efficient help.


Thank





David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34


Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr  -   www.ipeva-studio.com



Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 14/10/2010 ? 09:55, ? Chiang ?? Chih-Chung Wybie a ?crit :

I tried Connect Two FreeSWITCH Boxes  http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
and my configuration :
 
60.248.175.37
 
(autoload_configs/act.conf.xml)
<list name="domains" default="deny">
      <node type="allow" cidr="60.248.175.38/28"/>
     <!-- <node type="allow" domain="$${domain}"/>-->
</list>

(dialplan/default.xml)
 
<extension name="Dial to 38">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="bridge" data="sofia/internal/$1 at 60.248.175.38"/>
      </condition>
 </extension>
 
(dialplan/public.xml)

<extension name="Calls from 38">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="transfer" data="$1 XML default"/>
      </condition>
 </extension>
 
 
 
60.248.175.38
 
(autoload_configs/act.conf.xml)
<list name="domains" default="deny">
      <node type="allow" cidr="60.248.175.37/28"/>
     <!-- <node type="allow" domain="$${domain}"/>-->
</list>

(dialplan/default.xml)
 
<extension name="Dial to 37">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="bridge" data="sofia/internal/$1 at 60.248.175.37"/>
      </condition>
 </extension>
 
(dialplan/public.xml)

<extension name="Calls from 37">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="transfer" data="$1 XML default"/>
      </condition>
 </extension>
 
when I use 1007 at 38 calls to 1007 at 37
it won't show errors but still not work
 
how should I change something ?
 
Thank you for your attension
 
Best Regards
Gary
 
 
 
 
 
> Date: Wed, 13 Oct 2010 22:17:13 -0700
> From: curriegrad2004 at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> 
> You can create a SIP trunk between the two servers so calls can be
> routed back and forth from the servers you've specified.
> 
> 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
> > I tried it
> > but not working
> > and show errors below
> > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493 Cannot create
> > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493 Cannot create
> > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate Failed.
> > Cause: USER_NOT_REGISTERED
> >
> > and for example
> >  I use SIP server 60.248.175.38  user id=1007
> > and I call to 1007 at 60.248.175.37    that will show these errors
> >
> > how should set more?
> >
> > thank you for your attension
> >
> > Best Regards
> > Gary
> >
> > ________________________________
> > Date: Thu, 14 Oct 2010 10:55:49 +0800
> > From: xyangni at gmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >
> > I think you can reg at server A and than call B as a normal SIP call to
> > external domain.
> >
> > 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
> >
> >
> >
> > Hello
> >
> > I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X-Lite to call
> >
> > my question is how can I call between these servers?
> >
> > if I can , how shoould I set in *.xml? or X-Lite?
> >
> > thank you for your attension
> >
> >
> > Best Regards
> > Gary
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________ FreeSWITCH-users mailing
> > list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
_______________________________________________
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_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org_______________________________________________
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_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 		 	   		  
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From david.ponzone at ipeva.fr  Fri Oct 15 02:16:41 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 15 Oct 2010 11:16:41 +0200
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <SNT130-w6005F900786BDFBE48FF43FB570@phx.gbl>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>, , ,
	<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>, , ,
	<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>, ,
	, <SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>, , ,
	<AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>, ,
	<SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>, ,
	<C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>,
	<SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>,
	<6A8D7D0E-4DF5-4F35-8A00-F52FECC4F826@ipeva.fr>
	<SNT130-w6005F900786BDFBE48FF43FB570@phx.gbl>
Message-ID: <AD407430-9024-4E0D-A860-9026FD2837A3@ipeva.fr>

You need to understand that if you dial a number that you want to be  
routed to your 60.248.175.37, you need NOT TO HAVE this number  
intercepted by the default context on 60.248.175.38.
In the log you sent, the call is staying local to 60.248.175.38.
It never reaches 60.248.175.37.

You need to have different numbering plan (you can't have a 4001 on  
both machines and hope to reach both with the same number).
Or you keep the same numbering plan, but you use a prefix in the  
dialplan to reach the other host.
For instance you would dial 994001, but your dialplan will match it  
with a regexp like ^99(4\d{3})$ and bridge $1 (so only 4001) to the  
other FS.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 15/10/2010 ? 10:52, ? Chiang ?? Chih-Chung Wybie a ?crit :

> Hello this is my debug log
>
> I use 1013 at 60.248.175.38 call to 4001 at 60.248.175.37
>
> EXECUTE sofia/internal/1013 at 60.248.175.38 transfer(4001 XML default)
> 2010-10-15 16:49:30.666644 [DEBUG] switch_ivr.c:1444 (sofia/internal/1013 at 60.248.175.38 
> ) State Change CS_EXECUTE -> CS_ROUTING
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_session.c:642 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-15 16:49:30.666644 [NOTICE] switch_ivr.c:1450 Transfer sofia/internal/1013 at 60.248.175.38 
>  to XML[4001 at default]
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38 
> ) State EXECUTE going to sleep
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_ROUTING
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38 
> ) State ROUTING
> 2010-10-15 16:49:30.666644 [DEBUG] mod_sofia.c:140 sofia/internal/1013 at 60.248.175.38 
>  SOFIA ROUTING
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1013 at 60.248.175.38Standard 
>  ROUTING
> 2010-10-15 16:49:30.666644 [INFO] mod_dialplan_xml.c:331 Processing  
> 1013->4001 in context default
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >unloop] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (PASS) [unloop] $ 
> {unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [unloop] $ 
> {sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >tod_example] continue=true
> Dialplan: sofia/internal/1013 at 60.248.175.38 Date/Time Match (PASS)  
> [tod_example] break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(open=true)
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >holiday_example] continue=true
> Dialplan: sofia/internal/1013 at 60.248.175.38 Date/Time Match (FAIL)  
> [holiday_example] break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->global- 
> intercept] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [global- 
> intercept] destination_number(4001) =~ /^886$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->group- 
> intercept] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [group- 
> intercept] destination_number(4001) =~ /^\*8$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >intercept-ext] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [intercept- 
> ext] destination_number(4001) =~ /^\*\*(\d+)$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >redial] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [redial]  
> destination_number(4001) =~ /^(redial|870)$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >global] continue=true
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [global] $ 
> {call_debug}(false) =~ /^true$/ break=never
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [global] $ 
> {sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32| 
> AES_CM_128_HMAC_SHA1_80)$/ break=never
> Dialplan: sofia/internal/1013 at 60.248.175.38 Absolute Condition  
> [global]
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/$ 
> {domain_name}-spymap/${caller_id_number}/${uuid})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/$ 
> {domain_name}-last_dial/${caller_id_number}/${destination_number})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/$ 
> {domain_name}-last_dial/global/${uuid})
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->snom- 
> demo-2] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [snom- 
> demo-2] destination_number(4001) =~ /^9001$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->snom- 
> demo-1] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [snom- 
> demo-1] destination_number(4001) =~ /^9000$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >eavesdrop] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [eavesdrop]  
> destination_number(4001) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >eavesdrop] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [eavesdrop]  
> destination_number(4001) =~ /^779$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >call_return] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL)  
> [call_return] destination_number(4001) =~ /^\*69$|^869$|^lcr$/  
> break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->del- 
> group] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [del-group]  
> destination_number(4001) =~ /^80(\d{2})$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->add- 
> group] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [add-group]  
> destination_number(4001) =~ /^81(\d{2})$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->call- 
> group-simo] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call-group- 
> simo] destination_number(4001) =~ /^82(\d{2})$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->call- 
> group-order] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call-group- 
> order] destination_number(4001) =~ /^83(\d{2})$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >extension-intercom] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [extension- 
> intercom] destination_number(4001) =~ /^8(10[01][0-9])$/ break=on- 
> false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >Local_Extension] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (PASS)  
> [Local_Extension] destination_number(4001) =~ /^([0-4]0[01][0-9])$/  
> break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(RECORD_COPYRIGHT=(c) 2010)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(RECORD_SOFTWARE=FreeSwitch)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(RECORD_ARTIST=FreeSwitch)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(RECORD_COMMENT=FreeSwitch)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_DATE=$ 
> {strftime(%Y-%m-%d %H:%M)})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(RECORD_STEREO=true)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(dialed_extension=4001)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> export(dialed_extension=4001)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action set_user($ 
> {dialed_extension}@${domain_name})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> limit(60.248.175.38 ${dialed_extension} ${max_calls} ${fail_over})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action bind_meta_app(1 b  
> s execute_extension::dx XML features)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action bind_meta_app(2 b  
> s record_session::/usr/local/freeswitch/recordings/$ 
> {caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action bind_meta_app(3 b  
> s execute_extension::cf XML features)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(ringback=${us- 
> ring})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(transfer_ringback=local_stream://moh)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(call_timeout=30)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(hangup_after_bridge=true)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(continue_on_fail=true)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/$ 
> {domain_name}-call_return/${dialed_extension}/${caller_id_number})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/$ 
> {domain_name}-last_dial_ext/${dialed_extension}/${uuid})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(called_party_callgroup=${user_data(${dialed_extension}@$ 
> {domain_name} var callgroup)})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/$ 
> {domain_name}-last_dial/${called_party_callgroup}/${uuid})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action bridge(user/$ 
> {dialed_extension}@${domain_name})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action answer()
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action sleep(1000)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action bridge(loopback/ 
> app=voicemail:default ${domain_name} ${dialed_extension})
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1013 at 60.248.175.38 
> ) State Change CS_ROUTING -> CS_EXECUTE
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38 
> ) State ROUTING going to sleep
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_EXECUTE
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38 
> ) State EXECUTE
> 2010-10-15 16:49:30.666644 [DEBUG] mod_sofia.c:233 sofia/internal/1013 at 60.248.175.38 
>  SOFIA EXECUTE
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1013 at 60.248.175.38Standard 
>  EXECUTE
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(open=true)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [open]=[true]
> EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38- 
> spymap/1013/fe66f80a-992b-43f7-b130-47f0c87f4c15)
> EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38- 
> last_dial/1013/4001)
> EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38- 
> last_dial/global/fe66f80a-992b-43f7-b130-47f0c87f4c15)
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_COPYRIGHT=(c)  
> 2010)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [RECORD_COPYRIGHT]=[(c) 2010]
> EXECUTE sofia/internal/1013 at 60.248.175.38  
> set(RECORD_SOFTWARE=FreeSwitch)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [RECORD_SOFTWARE]=[FreeSwitch]
> EXECUTE sofia/internal/1013 at 60.248.175.38  
> set(RECORD_ARTIST=FreeSwitch)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [RECORD_ARTIST]=[FreeSwitch]
> EXECUTE sofia/internal/1013 at 60.248.175.38  
> set(RECORD_COMMENT=FreeSwitch)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [RECORD_COMMENT]=[FreeSwitch]
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_DATE=2010-10-15  
> 16:49)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [RECORD_DATE]=[2010-10-15 16:49]
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_STEREO=true)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [RECORD_STEREO]=[true]
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(dialed_extension=4001)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [dialed_extension]=[4001]
> EXECUTE sofia/internal/1013 at 60.248.175.38  
> export(dialed_extension=4001)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:901 EXPORT  
> [dialed_extension]=[4001]
> EXECUTE sofia/internal/1013 at 60.248.175.38 set_user(4001 at 60.248.175.38)
> EXECUTE sofia/internal/1013 at 60.248.175.38 limit(60.248.175.38 4001  )
> 2010-10-15 16:49:30.666644 [WARNING] mod_limit.c:779 USAGE: limit  
> <realm> <id> [<max> [number  [dialplan [context]]]]
> EXECUTE sofia/internal/1013 at 60.248.175.38 bind_meta_app(1 b s  
> execute_extension::dx XML features)
> 2010-10-15 16:49:30.666644 [INFO] switch_ivr_async.c:2372 Bound B- 
> Leg: 1 execute_extension::dx XML features
> EXECUTE sofia/internal/1013 at 60.248.175.38 bind_meta_app(2 b s  
> record_session::/usr/local/freeswitch/recordings/ 
> 1013.2010-10-15-16-49-30.wav)
> 2010-10-15 16:49:30.666644 [INFO] switch_ivr_async.c:2372 Bound B- 
> Leg: 2 record_session::/usr/local/freeswitch/recordings/ 
> 1013.2010-10-15-16-49-30.wav
> EXECUTE sofia/internal/1013 at 60.248.175.38 bind_meta_app(3 b s  
> execute_extension::cf XML features)
> 2010-10-15 16:49:30.666644 [INFO] switch_ivr_async.c:2372 Bound B- 
> Leg: 3 execute_extension::cf XML features
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(ringback= 
> %(2000,4000,440.0,480.0))
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [ringback]=[%(2000,4000,440.0,480.0)]
> EXECUTE sofia/internal/1013 at 60.248.175.38  
> set(transfer_ringback=local_stream://moh)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [transfer_ringback]=[local_stream://moh]
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(call_timeout=30)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [call_timeout]=[30]
> EXECUTE sofia/internal/1013 at 60.248.175.38  
> set(hangup_after_bridge=true)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [hangup_after_bridge]=[true]
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(continue_on_fail=true)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [continue_on_fail]=[true]
> EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38- 
> call_return/4001/1013)
> EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38- 
> last_dial_ext/4001/fe66f80a-992b-43f7-b130-47f0c87f4c15)
> EXECUTE sofia/internal/1013 at 60.248.175.38  
> set(called_party_callgroup=techsupport)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [called_party_callgroup]=[techsupport]
> EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38- 
> last_dial/techsupport/fe66f80a-992b-43f7-b130-47f0c87f4c15)
> EXECUTE sofia/internal/1013 at 60.248.175.38 bridge(user/4001 at 60.248.175.38 
> )
> 2010-10-15 16:49:30.666644 [DEBUG] switch_ivr_originate.c:1946  
> variable string 0 = [presence_id=4001 at 60.248.175.38]
> 2010-10-15 16:49:30.666644 [ERR] switch_ivr_originate.c:2493 Cannot  
> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> 2010-10-15 16:49:30.666644 [DEBUG] switch_ivr_originate.c:3308  
> Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
> 2010-10-15 16:49:30.666644 [ERR] switch_ivr_originate.c:2493 Cannot  
> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> 2010-10-15 16:49:30.666644 [DEBUG] switch_ivr_originate.c:3308  
> Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
> 2010-10-15 16:49:30.666644 [INFO] mod_dptools.c:2356 Originate  
> Failed.  Cause: USER_NOT_REGISTERED
> EXECUTE sofia/internal/1013 at 60.248.175.38 answer()
> 2010-10-15 16:49:30.694130 [DEBUG] sofia_glue.c:2616 AUDIO RTP [sofia/internal/1013 at 60.248.175.38 
> ] 60.248.175.38 port 25606 -> 60.248.175.43 port 35070 codec: 0 ms: 20
> 2010-10-15 16:49:30.694130 [DEBUG] switch_rtp.c:1343 Starting timer  
> [soft] 160 bytes per 20ms
> 2010-10-15 16:49:30.695467 [DEBUG] sofia_glue.c:2822 Set 2833 dtmf  
> send payload to 101
> 2010-10-15 16:49:30.695467 [DEBUG] sofia_glue.c:2827 Set 2833 dtmf  
> receive payload to 101
> 2010-10-15 16:49:30.695467 [DEBUG] mod_sofia.c:663 Local SDP sofia/internal/1013 at 60.248.175.38 
> :
> v=0
> o=FreeSWITCH 1287106964 1287106965 IN IP4 60.248.175.38
> s=FreeSWITCH
> c=IN IP4 60.248.175.38
> t=0 0
> m=audio 25606 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> 2010-10-15 16:49:30.695467 [DEBUG] switch_core_session.c:642 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-15 16:49:30.695467 [NOTICE] mod_dptools.c:720 Channel [sofia/internal/1013 at 60.248.175.38 
> ] has been answered
> EXECUTE sofia/internal/1013 at 60.248.175.38 sleep(1000)
> 2010-10-15 16:49:30.698172 [DEBUG] sofia.c:4200 Channel sofia/internal/1013 at 60.248.175.38 
>  entering state [completed][200]
> 2010-10-15 16:49:30.745378 [DEBUG] switch_rtp.c:2443 Correct ip/port  
> confirmed.
> 2010-10-15 16:49:30.807388 [DEBUG] sofia.c:4200 Channel sofia/internal/1013 at 60.248.175.38 
>  entering state [ready][200]
> EXECUTE sofia/internal/1013 at 60.248.175.38 bridge(loopback/ 
> app=voicemail:default 60.248.175.38 4001)
> 2010-10-15 16:49:31.704837 [NOTICE] switch_channel.c:675 New Channel  
> loopback/app=voicemail:default 60.248.175.38 4001-a [23f3d5de- 
> f985-44bd-8b1a-118b98bb5eae]
> 2010-10-15 16:49:31.704837 [DEBUG] mod_loopback.c:127 loopback/ 
> app=voicemail:default 60.248.175.38 4001-a setup codec PCMU/8000/20
> 2010-10-15 16:49:31.704837 [NOTICE] switch_channel.c:673 Rename  
> Channel loopback/app=voicemail:default 60.248.175.38 4001-a- 
> >loopback/voicemail-a [23f3d5de-f985-44bd-8b1a-118b98bb5eae]
> 2010-10-15 16:49:31.704837 [DEBUG] mod_loopback.c:911 (loopback/ 
> voicemail-a) State Change CS_NEW -> CS_INIT
> 2010-10-15 16:49:31.704837 [DEBUG] switch_core_session.c:1022 Send  
> signal loopback/voicemail-a [BREAK]
> 2010-10-15 16:49:31.704837 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-a CHANNEL KILL
> 2010-10-15 16:49:31.706218 [DEBUG] switch_core_state_machine.c:314  
> (loopback/voicemail-a) Running State Change CS_INIT
> 2010-10-15 16:49:31.706218 [DEBUG] switch_core_state_machine.c:338  
> (loopback/voicemail-a) State INIT
> 2010-10-15 16:49:31.706218 [NOTICE] switch_channel.c:675 New Channel  
> loopback/voicemail-b [1abc53a2-df93-40db-96ab-547f9ddd1b1d]
> 2010-10-15 16:49:31.706218 [DEBUG] mod_loopback.c:127 loopback/ 
> voicemail-b setup codec PCMU/8000/20
> 2010-10-15 16:49:31.706218 [DEBUG] mod_loopback.c:240 (loopback/ 
> voicemail-b) State Change CS_NEW -> CS_INIT
> 2010-10-15 16:49:31.706218 [DEBUG] switch_core_session.c:1022 Send  
> signal loopback/voicemail-b [BREAK]
> 2010-10-15 16:49:31.706218 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-b CHANNEL KILL
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:314  
> (loopback/voicemail-b) Running State Change CS_INIT
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:338  
> (loopback/voicemail-b) State INIT
> 2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:281 (loopback/ 
> voicemail-b) State Change CS_INIT -> CS_ROUTING
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_session.c:1022 Send  
> signal loopback/voicemail-b [BREAK]
> 2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-b CHANNEL KILL
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:338  
> (loopback/voicemail-b) State INIT going to sleep
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:314  
> (loopback/voicemail-b) Running State Change CS_ROUTING
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:341  
> (loopback/voicemail-b) State ROUTING
> 2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:315 loopback/ 
> voicemail-b CHANNEL ROUTING
> 2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:326 (loopback/ 
> voicemail-b) State Change CS_ROUTING -> CS_EXECUTE
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_session.c:1022 Send  
> signal loopback/voicemail-b [BREAK]
> 2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-b CHANNEL KILL
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:341  
> (loopback/voicemail-b) State ROUTING going to sleep
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:314  
> (loopback/voicemail-b) Running State Change CS_EXECUTE
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:348  
> (loopback/voicemail-b) State EXECUTE
> 2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:346 loopback/ 
> voicemail-b CHANNEL EXECUTE
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:157  
> loopback/voicemail-b Standard EXECUTE
> EXECUTE loopback/voicemail-b pre_answer()
> 2010-10-15 16:49:31.707132 [NOTICE] mod_loopback.c:716 Pre-Answer  
> loopback/voicemail-a!
> 2010-10-15 16:49:31.707132 [DEBUG] switch_channel.c:2254 Send signal sofia/internal/1013 at 60.248.175.38 
> [BREAK]
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_session.c:642 Send  
> signal loopback/voicemail-b [BREAK]
> 2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-b CHANNEL KILL
> 2010-10-15 16:49:31.707132 [NOTICE] mod_dptools.c:746 Pre-Answer  
> loopback/voicemail-b!
> 2010-10-15 16:49:31.707132 [DEBUG] switch_channel.c:2254 Send signal sofia/internal/1013 at 60.248.175.38 
> [BREAK]
> EXECUTE loopback/voicemail-b voicemail(default 60.248.175.38 4001)
> 2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:281 (loopback/ 
> voicemail-a) State Change CS_INIT -> CS_ROUTING
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_session.c:1022 Send  
> signal loopback/voicemail-a [BREAK]
> 2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-a CHANNEL KILL
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:338  
> (loopback/voicemail-a) State INIT going to sleep
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:314  
> (loopback/voicemail-a) Running State Change CS_ROUTING
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:341  
> (loopback/voicemail-a) State ROUTING
> 2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:315 loopback/ 
> voicemail-a CHANNEL ROUTING
> 2010-10-15 16:49:31.708787 [DEBUG] switch_ivr_originate.c:66  
> (loopback/voicemail-a) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_session.c:1022 Send  
> signal loopback/voicemail-a [BREAK]
> 2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-a CHANNEL KILL
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:341  
> (loopback/voicemail-a) State ROUTING going to sleep
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:314  
> (loopback/voicemail-a) Running State Change CS_CONSUME_MEDIA
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:360  
> (loopback/voicemail-a) State CONSUME_MEDIA
> 2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:512 CHANNEL  
> CONSUME_MEDIA
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:360  
> (loopback/voicemail-a) State CONSUME_MEDIA going to sleep
> 2010-10-15 16:49:31.710698 [DEBUG] switch_ivr_originate.c:3228  
> Originate Resulted in Success: [loopback/voicemail-a]
> 2010-10-15 16:49:31.710698 [DEBUG] switch_core_session.c:642 Send  
> signal loopback/voicemail-a [BREAK]
> 2010-10-15 16:49:31.710698 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-a CHANNEL KILL
> 2010-10-15 16:49:31.710698 [DEBUG] switch_core_session.c:642 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-15 16:49:31.710698 [DEBUG] switch_ivr_bridge.c:1182  
> (loopback/voicemail-a) State Change CS_CONSUME_MEDIA ->  
> CS_EXCHANGE_MEDIA
> 2010-10-15 16:49:31.710698 [DEBUG] switch_core_session.c:1022 Send  
> signal loopback/voicemail-a [BREAK]
> 2010-10-15 16:49:31.710698 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-a CHANNEL KILL
> 2010-10-15 16:49:31.710698 [DEBUG] switch_core_state_machine.c:314  
> (loopback/voicemail-a) Running State Change CS_EXCHANGE_MEDIA
> 2010-10-15 16:49:31.710698 [DEBUG] switch_core_state_machine.c:351  
> (loopback/voicemail-a) State EXCHANGE_MEDIA
> 2010-10-15 16:49:31.710698 [DEBUG] mod_loopback.c:474 CHANNEL LOOPBACK
> 2010-10-15 16:49:31.712661 [DEBUG] switch_core_session.c:703 Send  
> signal loopback/voicemail-a [BREAK]
> 2010-10-15 16:49:31.712661 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-a CHANNEL KILL
> 2010-10-15 16:49:31.712661 [DEBUG] switch_core_session.c:703 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-15 16:49:31.824953 [DEBUG] switch_ivr_play_say.c:63 No  
> language specified - Using [en]
> 2010-10-15 16:49:31.824953 [DEBUG] switch_ivr_play_say.c:244 Handle  
> play-file:[voicemail/vm-person.wav] (en:en)
> 2010-10-15 16:49:31.824953 [DEBUG] switch_ivr_play_say.c:1152 Codec  
> Activated L16 at 8000hz 1 channels 20ms
> 2010-10-15 16:49:33.185370 [DEBUG] switch_ivr_play_say.c:1444 done  
> playing file
> 2010-10-15 16:49:33.305429 [DEBUG] switch_ivr_play_say.c:244 Handle  
> say:[4001] (en:en)
> 2010-10-15 16:49:33.305429 [DEBUG] switch_ivr_play_say.c:1152 Codec  
> Activated L16 at 8000hz 1 channels 20ms
> 2010-10-15 16:49:33.744961 [DEBUG] switch_ivr_play_say.c:1444 done  
> playing file
> 2010-10-15 16:49:33.744961 [DEBUG] switch_ivr_play_say.c:1152 Codec  
> Activated L16 at 8000hz 1 channels 20ms
>
>
> From: david.ponzone at ipeva.fr
> To: freeswitch-users at lists.freeswitch.org
> Date: Thu, 14 Oct 2010 12:20:56 +0200
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> Don't be shy on the logs, send the full ones!
>
> in fs_cli:
> fsctl loglevel debug
> /log 7
>
> and then make a call
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
> message, merci de le d?truire imm?diatement et d'avertir  
> l'exp?diteur.
>
>
>
>
> Le 14/10/2010 ? 11:53, ? Chiang ?? Chih-Chung Wybie a ?crit :
>
> I tried use 1013 at 38 calls to 1019 at 37
>
> show log below
>
> 2010-10-14 17:50:41.249876 [NOTICE] switch_channel.c:675 New Channel sofia/internal/1013 at 60.248.175.38 
> [875fdeef-b94e-41fb-a621-ea005bbaedbd]
> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing  
> 1013->1019 in context public
> 2010-10-14 17:50:41.263809 [NOTICE] switch_ivr.c:1450 Transfer sofia/internal/1013 at 60.248.175.38 
>  to XML[1019 at default]
> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing  
> 1013->1019 in context default
> 2010-10-14 17:50:41.283904 [INFO] switch_core_session.c:1764 Sending  
> early media
> 2010-10-14 17:50:41.285023 [NOTICE] mod_sofia.c:1994 Pre-Answer sofia/internal/1013 at 60.248.175.38 
> !
>
> seems not work
>
> Thank you for you attension
> Best Regards
>
> Gary
>
>
>
> From: david.ponzone at ipeva.fr
> To: freeswitch-users at lists.freeswitch.org
> Date: Thu, 14 Oct 2010 10:16:28 +0200
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> I think we would need a complete log from the target FS, in order to  
> provide you with any efficient help.
>
> Thank
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
> message, merci de le d?truire imm?diatement et d'avertir  
> l'exp?diteur.
>
>
>
>
> Le 14/10/2010 ? 09:55, ? Chiang ?? Chih-Chung Wybie a ?crit :
>
> I tried Connect Two FreeSWITCH Boxes  http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
> and my configuration :
>
> 60.248.175.37
>
> (autoload_configs/act.conf.xml)
> <list name="domains" default="deny">
>       <node type="allow" cidr="60.248.175.38/28"/>
>      <!-- <node type="allow" domain="$${domain}"/>-->
> </list>
>
> (dialplan/default.xml)
>
> <extension name="Dial to 38">
>       <condition field="destination_number" expression="^([0-4]0[01] 
> [0-9])$">
>         <action application="bridge" data="sofia/internal/$1 at 60.248.175.38 
> "/>
>       </condition>
>  </extension>
>
> (dialplan/public.xml)
>
> <extension name="Calls from 38">
>       <condition field="destination_number" expression="^([0-4]0[01] 
> [0-9])$">
>         <action application="transfer" data="$1 XML default"/>
>       </condition>
>  </extension>
>
>
>
> 60.248.175.38
>
> (autoload_configs/act.conf.xml)
> <list name="domains" default="deny">
>       <node type="allow" cidr="60.248.175.37/28"/>
>      <!-- <node type="allow" domain="$${domain}"/>-->
> </list>
>
> (dialplan/default.xml)
>
> <extension name="Dial to 37">
>       <condition field="destination_number" expression="^([0-4]0[01] 
> [0-9])$">
>         <action application="bridge" data="sofia/internal/$1 at 60.248.175.37 
> "/>
>       </condition>
>  </extension>
>
> (dialplan/public.xml)
>
> <extension name="Calls from 37">
>       <condition field="destination_number" expression="^([0-4]0[01] 
> [0-9])$">
>         <action application="transfer" data="$1 XML default"/>
>       </condition>
>  </extension>
>
> when I use 1007 at 38 calls to 1007 at 37
> it won't show errors but still not work
>
> how should I change something ?
>
> Thank you for your attension
>
> Best Regards
> Gary
>
>
>
>
>
> > Date: Wed, 13 Oct 2010 22:17:13 -0700
> > From: curriegrad2004 at gmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X- 
> Lite
> >
> > You can create a SIP trunk between the two servers so calls can be
> > routed back and forth from the servers you've specified.
> >
> > 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
> > > I tried it
> > > but not working
> > > and show errors below
> > > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493  
> Cannot create
> > > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> > > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493  
> Cannot create
> > > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> > > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate  
> Failed.
> > > Cause: USER_NOT_REGISTERED
> > >
> > > and for example
> > >  I use SIP server 60.248.175.38  user id=1007
> > > and I call to 1007 at 60.248.175.37    that will show these errors
> > >
> > > how should set more?
> > >
> > > thank you for your attension
> > >
> > > Best Regards
> > > Gary
> > >
> > > ________________________________
> > > Date: Thu, 14 Oct 2010 10:55:49 +0800
> > > From: xyangni at gmail.com
> > > To: freeswitch-users at lists.freeswitch.org
> > > Subject: Re: [Freeswitch-users] some questions of freeswitch & X- 
> Lite
> > >
> > > I think you can reg at server A and than call B as a normal SIP  
> call to
> > > external domain.
> > >
> > > 2010/10/14 ? Chiang ?? Chih-Chung Wybie  
> <tear152 at hotmail.com>
> > >
> > >
> > >
> > > Hello
> > >
> > > I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X- 
> Lite to call
> > >
> > > my question is how can I call between these servers?
> > >
> > > if I can , how shoould I set in *.xml? or X-Lite?
> > >
> > > thank you for your attension
> > >
> > >
> > > Best Regards
> > > Gary
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> > >
> > > _______________________________________________ FreeSWITCH-users  
> mailing
> > > list FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________ FreeSWITCH-users  
> mailing listFreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
>  http:// 
> www.freeswitch.org_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________ FreeSWITCH-users  
> mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
>  http:// 
> www.freeswitch.org_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From steveayre at gmail.com  Fri Oct 15 02:47:14 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 15 Oct 2010 10:47:14 +0100
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <SNT130-w486F7895E102C75B06D2F3FB570@phx.gbl>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>
	<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>
	<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>
	<SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>
	<AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>
	<SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>
	<C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>
	<SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>
	<AANLkTimsBQtco2B-s6uXbkrQ7WoiQ0SzF8N5=wSskOML@mail.gmail.com>
	<AANLkTikTSTtaN1cd14U+G8yKA1u=A3dCgtKgSrEkYDb-@mail.gmail.com>
	<SNT130-w1462D55D67F6FED40D4D74FB570@phx.gbl>
	<AANLkTikbix9D=E+eAueVHWqzDM-5_SV=xb8z9_NastDF@mail.gmail.com>
	<SNT130-w3B5414501256848B3E382FB570@phx.gbl>
	<AANLkTik=eus1MrmnpCFRvfDCGpE7WYmrf1QgQpnJ=Vg3@mail.gmail.com>
	<SNT130-w486F7895E102C75B06D2F3FB570@phx.gbl>
Message-ID: <AANLkTikzO-gd5Xkr1THncvUrWsPqqd1M2wNsG0aUDzqR@mail.gmail.com>

"Well I can't login pastebin.freeswitch.org"

Read the login message carefully...

-Steve


2010/10/15 ? Chiang ?? Chih-Chung   Wybie <tear152 at hotmail.com>:
> Well I can't login pastebin.freeswitch.org
> and that's my configs below
>
> IP address: 60.248.175.37
> Mask       :255.255.255.240
> GateWay : 60.248.175.33
> DNS        :168.95.1.1
>
>  (autoload_configs/act.conf.xml)
>  <list name="domains" default="deny">
>       <node type="allow" cidr="60.248.175.32/28"/>
>       <!-- <node type="allow" domain="$${domain}"/>-->
>  </list>
>
>  (dialplan/default.xml)
>  <extension name="Dial to 38">
>        <condition field="destination_number"
>  expression="^([0-4]0[01][0-9])$">
>        <action application="bridge"
>  data="sofia/internal/$1 at 60.248.175.38"/>
>        </condition>
>  </extension>
>
>  (dialplan/public.xml)
>
>  <extension name="Calls from 38">
>       <condition field="destination_number"
>  expression="^([0-4]0[01][0-9])$">
>         <action application="transfer" data="$1 XML default"/>
>        </condition>
>   </extension>
>
>
>
>
> IP address: 60.248.175.38
> Mask       :255.255.255.240
> GateWay : 60.248.175.33
> DNS        :168.95.1.1
>
>
>  (autoload_configs/act.conf.xml)
>  <list name="domains" default="deny">
>        <node type="allow" cidr="60.248.175.32/28"/>
>       <!-- <node type="allow" domain="$${domain}"/>-->
>  </list>
>
>  (dialplan/default.xml)
>
>  <extension name="Dial to 37">
>        <condition field="destination_number"
>  expression="^([0-4]0[01][0-9])$">
>          <action application="bridge"
>  data="sofia/internal/$1 at 60.248.175.37"/>
>        </condition>
>   </extension>
>
>  (dialplan/public.xml)
>
>  <extension name="Calls from 37">
>        <condition field="destination_number"
>  expression="^([0-4]0[01][0-9])$">
>          <action application="transfer" data="$1 XML default"/>
>        </condition>
>   </extension>
>
> Thank you for your attension
>
> Best Regards
>
> Gary
>
>
>
> ________________________________
> Date: Thu, 14 Oct 2010 20:02:16 -0700
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> Did you try using ACLs or with putting a user/gateway on each side?
> Put your configs on pastebin.freeswitch.org.
>
> This is actually a simple exercise once you know what to do. Start by
> deciding if you want to do authentication by digest or IP address.
>
> -MC
>
> 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
>
> But my question is
>
> I set two fs
>
> and I want to connect them
>
> I refer to http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
>
> it 's not working
>
> so I give my configuration already
>
> should I change some setting?
>
> Thank you for your attension
>
> Best Regards
>
> Gary
>
> ________________________________
> Date: Thu, 14 Oct 2010 17:56:43 -0700
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> Yes! You don't need a gateway for this. If the users are all on the same
> server then you just need to create the necessary files in
> conf/directory/default/
> You already have 1000.xml, 1001.xml, ... 1019.xml
> You just need to add 2010.xml, 4001.xml, 3009.xml, etc.
> Check out this article:
> http://bit.ly/EpVrv
>
> It's a bit older, but the section on adding a new extension is very
> accurate. In that example it shows how to add extension 1500 to your
> directory and your dialplan. You can copy that example for all of your new
> extension numbers.
>
> -MC
>
> 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
>
> Hello
>
> but I can call to each other on the same server
>
> for example
>
> I can use 1007 at 37  call 1013 at 37  ,  4001 at 37  , 2010 at 37 , 3009 at 37
>
>
>  Thank you for your attension
> Best Regards
>  Gary
>
>
>
> ________________________________
> Date: Thu, 14 Oct 2010 11:27:31 -0700
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> FYI, we have several resources to help
> http://wiki.freeswitch.org/wiki/Regular_Expression
> http://bit.ly/aijtAC
>
> :)
>
> -MC
>
> On Thu, Oct 14, 2010 at 11:18 AM, Jeffrey Leung <curriegrad2004 at gmail.com>
> wrote:
>
> Dialplan's regex doesn't look right to me. You'll need to fix that
> before you do anything else.
>
> I'd suggest this regex instead of the one you provided:
> ^[0-4]0(10[1-9][0-9])$
>
> instead of this one
> ^([0-4]0[01][0-9])$
>
> If you can't get the grasp of regular expressions, I'd highly suggest
> you to pick up a book and read up on it. Knowing regular expressions
> is a great skill to have ;)
>
> 2010/10/14 ? Chiang ?? Chih-Chung   Wybie <tear152 at hotmail.com>:
>> I tried use 1013 at 38 calls to 1019 at 37
>>
>> show log below
>>
>> 2010-10-14 17:50:41.249876 [NOTICE] switch_channel.c:675 New Channel
>> sofia/internal/1013 at 60.248.175.38 [875fdeef-b94e-41fb-a621-ea005bbaedbd]
>> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
>> 1013->1019 in context public
>> 2010-10-14 17:50:41.263809 [NOTICE] switch_ivr.c:1450 Transfer
>> sofia/internal/1013 at 60.248.175.38 to XML[1019 at default]
>> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
>> 1013->1019 in context default
>> 2010-10-14 17:50:41.283904 [INFO] switch_core_session.c:1764 Sending early
>> media
>> 2010-10-14 17:50:41.285023 [NOTICE] mod_sofia.c:1994 Pre-Answer
>> sofia/internal/1013 at 60.248.175.38!
>>
>> seems not work
>>
>> Thank you for you attension
>> Best Regards
>>
>> Gary
>>
>>
>>
>> ________________________________
>> From: david.ponzone at ipeva.fr
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Thu, 14 Oct 2010 10:16:28 +0200
>> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>>
>> I think we would need a complete log from the target FS, in order to
>> provide
>> you with any efficient help.
>> Thank
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
>> s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>> Le 14/10/2010 ? 09:55, ? Chiang ?? Chih-Chung Wybie a ?crit :
>>
>> I tried Connect Two FreeSWITCH
>> Boxes  http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
>> and my configuration :
>>
>> 60.248.175.37
>>
>> (autoload_configs/act.conf.xml)
>> <list name="domains" default="deny">
>>       <node type="allow" cidr="60.248.175.38/28"/>
>>      <!-- <node type="allow" domain="$${domain}"/>-->
>> </list>
>>
>> (dialplan/default.xml)
>>
>> <extension name="Dial to 38">
>>       <condition field="destination_number"
>> expression="^([0-4]0[01][0-9])$">
>>         <action application="bridge"
>> data="sofia/internal/$1 at 60.248.175.38"/>
>>       </condition>
>>  </extension>
>>
>> (dialplan/public.xml)
>>
>> <extension name="Calls from 38">
>>       <condition field="destination_number"
>> expression="^([0-4]0[01][0-9])$">
>>         <action application="transfer" data="$1 XML default"/>
>>       </condition>
>>  </extension>
>>
>>
>>
>> 60.248.175.38
>>
>> (autoload_configs/act.conf.xml)
>> <list name="domains" default="deny">
>>       <node type="allow" cidr="60.248.175.37/28"/>
>>      <!-- <node type="allow" domain="$${domain}"/>-->
>> </list>
>>
>> (dialplan/default.xml)
>>
>> <extension name="Dial to 37">
>>       <condition field="destination_number"
>> expression="^([0-4]0[01][0-9])$">
>>         <action application="bridge"
>> data="sofia/internal/$1 at 60.248.175.37"/>
>>       </condition>
>>  </extension>
>>
>> (dialplan/public.xml)
>>
>> <extension name="Calls from 37">
>>       <condition field="destination_number"
>> expression="^([0-4]0[01][0-9])$">
>>         <action application="transfer" data="$1 XML default"/>
>>       </condition>
>>  </extension>
>>
>> when I use 1007 at 38 calls to 1007 at 37
>> it won't show errors but still not work
>>
>> how should I change something ?
>>
>> Thank you for your attension
>>
>> Best Regards
>> Gary
>>
>>
>>
>>
>>
>>> Date: Wed, 13 Oct 2010 22:17:13 -0700
>>> From: curriegrad2004 at gmail.com
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>>>
>>> You can create a SIP trunk between the two servers so calls can be
>>> routed back and forth from the servers you've specified.
>>>
>>> 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
>>> > I tried it
>>> > but not working
>>> > and show errors below
>>> > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493 Cannot
>>> > create
>>> > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
>>> > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493 Cannot
>>> > create
>>> > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
>>> > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate Failed.
>>> > Cause: USER_NOT_REGISTERED
>>> >
>>> > and for example
>>> >  I use SIP server 60.248.175.38  user id=1007
>>> > and I call to 1007 at 60.248.175.37    that will show these errors
>>> >
>>> > how should set more?
>>> >
>>> > thank you for your attension
>>> >
>>> > Best Regards
>>> > Gary
>>> >
>>> > ________________________________
>>> > Date: Thu, 14 Oct 2010 10:55:49 +0800
>>> > From: xyangni at gmail.com
>>> > To: freeswitch-users at lists.freeswitch.org
>>> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>>> >
>>> > I think you can reg at server A and than call B as a normal SIP call to
>>> > external domain.
>>> >
>>> > 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
>>> >
>>> >
>>> >
>>> > Hello
>>> >
>>> > I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X-Lite to
>>> > call
>>> >
>>> > my question is how can I call between these servers?
>>> >
>>> > if I can , how shoould I set in *.xml? or X-Lite?
>>> >
>>> > thank you for your attension
>>> >
>>> >
>>> > Best Regards
>>> > Gary
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> > _______________________________________________ FreeSWITCH-users
>>> > mailing
>>> > list FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________ FreeSWITCH-users mailing
>> list FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From Holger.Esser at Convergys.com  Fri Oct 15 04:32:07 2010
From: Holger.Esser at Convergys.com (Esser, Holger)
Date: Fri, 15 Oct 2010 06:32:07 -0500
Subject: [Freeswitch-users] mod_cepstral 5.1 known issues?
In-Reply-To: <4CB7F0C9.2040708@solomo.de>
References: <AANLkTik8KZdxAnUUfv0nKF6EkTHe559PtC2bF74eng11@mail.gmail.com>
	<AANLkTi=MvWh14MQ9wPEboXZmMgAtURAY5zBoobn32Pkq@mail.gmail.com>
	<AANLkTimj=SwX7d1sot+NhOxMgD4F-PN=bM3hxrh3bF3O@mail.gmail.com>
	<4CB7F0C9.2040708@solomo.de>
Message-ID: <0FF607C13B7F7A42B5B80DA9EB204C9109D44BDD8E@srv-ex01-dal.intervoice.int>

Hi Daniel,

May I ask which guide you used to set it up?

Thanks,
Holger

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Daniel Neubert
Sent: Friday, October 15, 2010 1:12 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_cepstral 5.1 known issues?

I've exactly this setup which is in productive use for a few months now.

Using cepstral directly on our FreeSWITCH nodes did not work (since they are running on FreeBSD and Cepstral is only available for GNU Linux).

Current setup is using Voice Katrin (German) Version 5.1.0 on Gentoo Linux 64Bit via UniMRCP Server 1.0.0.

Best regards / Mit freundlichen Gr??en,
Daniel Neubert

On 14.10.2010 17:58, Marc Kellem wrote:
Is anyone using mod_unimrcp to access Cepstral 5.1 voices in a UniMRCP server?

On Thu, Oct 14, 2010 at 11:26 AM, Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>> wrote:
random deadlocks in the cepstral engine blocking all the speech
generation calls.


On Thu, Oct 14, 2010 at 9:56 AM, Marc Kellem <mkellem at vontoo.com<mailto:mkellem at vontoo.com>> wrote:
> The mod_cepstral wiki page says "Warning: the suggested version to use
> is 4.x since there are known issues with 5.1 (which is closed source)".
>  What exactly are the known issues?
> http://wiki.freeswitch.org/wiki/Mod_cepstral
> Thanks,
> Marc Kellem
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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<mailto:MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


No virus found in this incoming message.
Checked by AVG - www.avg.com
Version: 9.0.862 / Virus Database: 271.1.1/3192 - Release Date: 10/14/10 13:34:00

________________________________
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From woodydickson at gmail.com  Fri Oct 15 06:06:23 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Fri, 15 Oct 2010 21:06:23 +0800
Subject: [Freeswitch-users] reload caused freeswitch to hang
Message-ID: <AANLkTikTytjhWO0Gfu9zriNBys3J=Hbu_cyKhKAi0+T1@mail.gmail.com>

Hi,

I was able to use the reload API to reload any of my application, but after
I upgraded to git version, I found that the RELOAD APP would cause the whole
freeswitch to hang.

Even "sofia profile internal restart" causes the entire freeswitch console
to hang.

Does anyone encounter the same issue I am having with the latest git?

Thank,
Woody
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From abid_freeswitch at live.com  Fri Oct 15 06:39:17 2010
From: abid_freeswitch at live.com (Abid Saleem)
Date: Fri, 15 Oct 2010 19:39:17 +0600
Subject: [Freeswitch-users] Radius AAA
In-Reply-To: <4CADC2E4.90608@gmail.com>
References: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>
	<AANLkTi=GiQaO-48BP=T8c-A+09sJMKzQH7BX=ewBdB8R@mail.gmail.com>,
	<AANLkTikw8LGDqQki16D30b8JL0eJid2UOTs-bEaZSc2k@mail.gmail.com>,
	<4CADC2E4.90608@gmail.com>
Message-ID: <COL109-W528C7DF8FD0E4D203FD866F4570@phx.gbl>


Hi,
I have tried making and installing mod_rad_auth but it gives me the following error when loading the module.
2010-10-15 17:58:02.088712 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_rad_auth.so**libfreeradius-client.so.2: cannot open shared object file: No such file or directory**
Could you please help what is the issue and any resolution available. I installed freeradius-client-1.1.6 already.
Regards------------Abid SaleemSr. Product ManagerTerminus Technologies
> Date: Thu, 7 Oct 2010 17:53:56 +0500
> From: nazim.aghabayov at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Radius AAA
> 
>   Hello Tihomir,
> 
> I've just updated the Mod_rad_auth wiki with installation instructions.
> Thanks a lot for the mod. I use it in production and it works just great!
> 
> Regards,
> Nazim
> 
> On 10/07/2010 01:48 PM, Tihomir Culjaga wrote:
> > On Thu, Oct 7, 2010 at 1:15 AM, Michael Collins<msc at freeswitch.org>  wrote:
> >
> >> I'm afraid no such comprehensive documentation exists. Submissions to our
> >> wiki are most welcomed. This is all we have at the moment:
> >>
> >> http://wiki.freeswitch.org/wiki/Mod_rad_auth
> >>
> >> Be sure to install freeradius2 (server) and freeradius-client before trying
> >> to install mod_auth_rad.
> >>
> >> -MC
> >>
> >>
> >>
> > correct,
> >
> > Im the author of the module and im going to provide the documentation and
> > how-to.
> >
> >
> > in brief, this module does radius auth (not accounting). It is based on
> > freeradius-client library and as such this is the only dependency.
> >
> > you can specify your own list of VSAs to be included in the packet along
> > with the standard ones that are being used.
> >
> >
> >           name:       just a description
> >           value:      direct input or variable
> >           pec:        vendor ID (0 for default, 9 for cisco...)
> >           expr:       1 for channel variable, 2 for direct input (string)
> >           direction:  in for radius-request, out for radius-response
> >
> >
> > Im not going to describe it here... its better i do it on the wiki itself...
> >
> >
> >
> >
> > T.
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
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From tculjaga at gmail.com  Fri Oct 15 06:55:18 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 15 Oct 2010 15:55:18 +0200
Subject: [Freeswitch-users] Radius AAA
In-Reply-To: <COL109-W528C7DF8FD0E4D203FD866F4570@phx.gbl>
References: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>
	<AANLkTi=GiQaO-48BP=T8c-A+09sJMKzQH7BX=ewBdB8R@mail.gmail.com>
	<AANLkTikw8LGDqQki16D30b8JL0eJid2UOTs-bEaZSc2k@mail.gmail.com>
	<4CADC2E4.90608@gmail.com>
	<COL109-W528C7DF8FD0E4D203FD866F4570@phx.gbl>
Message-ID: <AANLkTi=5XSbZPkCr0_J3qzvjZ8h5gkwXi-fdgPntAs35@mail.gmail.com>

On Fri, Oct 15, 2010 at 3:39 PM, Abid Saleem <abid_freeswitch at live.com>wrote:

>  Hi,
>
> I have tried making and installing mod_rad_auth but it gives me the
> following error when loading the module.
>
> 2010-10-15 17:58:02.088712 [CRIT] switch_loadable_module.c:882 Error
> Loading module /usr/local/freeswitch/mod/mod_rad_auth.so
> **libfreeradius-client.so.2: cannot open shared object file: No such file
> or directory**
>
> Could you please help what is the issue and any resolution available. I
> installed freeradius-client-1.1.6 already.
>
>

just link the library into freeswitch/lib directory.

T.
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From anthony.minessale at gmail.com  Fri Oct 15 08:24:15 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Oct 2010 10:24:15 -0500
Subject: [Freeswitch-users] reload caused freeswitch to hang
In-Reply-To: <AANLkTikTytjhWO0Gfu9zriNBys3J=Hbu_cyKhKAi0+T1@mail.gmail.com>
References: <AANLkTikTytjhWO0Gfu9zriNBys3J=Hbu_cyKhKAi0+T1@mail.gmail.com>
Message-ID: <AANLkTimJurwTDWP3Ly6e=Sz8VTgWb1Z_nwqY+FVyqk_3@mail.gmail.com>

do you update with "make current" or at least "make update-clean &&
git pull && make install"


On Fri, Oct 15, 2010 at 8:06 AM, Woody Dickson <woodydickson at gmail.com> wrote:
> Hi,
> I was able to use the reload API to reload any of my application, but after
> I upgraded to git version, I found that the RELOAD APP would cause the whole
> freeswitch to hang.
> Even "sofia profile internal restart" causes the entire freeswitch console
> to hang.
> Does anyone encounter the same issue I am having with the latest git?
> Thank,
> Woody
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From djbinter at gmail.com  Fri Oct 15 08:55:19 2010
From: djbinter at gmail.com (DJB International)
Date: Fri, 15 Oct 2010 08:55:19 -0700
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <AANLkTikzO-gd5Xkr1THncvUrWsPqqd1M2wNsG0aUDzqR@mail.gmail.com>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>
	<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>
	<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>
	<SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>
	<AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>
	<SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>
	<C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>
	<SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>
	<AANLkTimsBQtco2B-s6uXbkrQ7WoiQ0SzF8N5=wSskOML@mail.gmail.com>
	<AANLkTikTSTtaN1cd14U+G8yKA1u=A3dCgtKgSrEkYDb-@mail.gmail.com>
	<SNT130-w1462D55D67F6FED40D4D74FB570@phx.gbl>
	<AANLkTikbix9D=E+eAueVHWqzDM-5_SV=xb8z9_NastDF@mail.gmail.com>
	<SNT130-w3B5414501256848B3E382FB570@phx.gbl>
	<AANLkTik=eus1MrmnpCFRvfDCGpE7WYmrf1QgQpnJ=Vg3@mail.gmail.com>
	<SNT130-w486F7895E102C75B06D2F3FB570@phx.gbl>
	<AANLkTikzO-gd5Xkr1THncvUrWsPqqd1M2wNsG0aUDzqR@mail.gmail.com>
Message-ID: <AANLkTikzKT6X5iXspoQEvgpUmagWZTxEoaq+a0=zUzdT@mail.gmail.com>

 (autoload_configs/act.conf.xml)
 <list name="domains" default="deny">
      <node type="allow" cidr="60.248.175.32/28 <http://60.248.175.38/28>"/>

      <!-- <node type="allow" domain="$${domain}"/>-->
 </list>

I also noticed and wondered why you have your acl in *act.conf.xml* instead
of acl.conf.xml



2010/10/15 Steven Ayre <steveayre at gmail.com>

> "Well I can't login pastebin.freeswitch.org"
>
> Read the login message carefully...
>
> -Steve
>
>
> 2010/10/15 ? Chiang ?? Chih-Chung   Wybie <tear152 at hotmail.com>:
> > Well I can't login pastebin.freeswitch.org
> > and that's my configs below
> >
> > IP address: 60.248.175.37
> > Mask       :255.255.255.240
> > GateWay : 60.248.175.33
> > DNS        :168.95.1.1
> >
> >  (autoload_configs/act.conf.xml)
> >  <list name="domains" default="deny">
> >       <node type="allow" cidr="60.248.175.32/28"/>
> >       <!-- <node type="allow" domain="$${domain}"/>-->
> >  </list>
> >
> >  (dialplan/default.xml)
> >  <extension name="Dial to 38">
> >        <condition field="destination_number"
> >  expression="^([0-4]0[01][0-9])$">
> >        <action application="bridge"
> >  data="sofia/internal/$1 at 60.248.175.38"/>
> >        </condition>
> >  </extension>
> >
> >  (dialplan/public.xml)
> >
> >  <extension name="Calls from 38">
> >       <condition field="destination_number"
> >  expression="^([0-4]0[01][0-9])$">
> >         <action application="transfer" data="$1 XML default"/>
> >        </condition>
> >   </extension>
> >
> >
> >
> >
> > IP address: 60.248.175.38
> > Mask       :255.255.255.240
> > GateWay : 60.248.175.33
> > DNS        :168.95.1.1
> >
> >
> >  (autoload_configs/act.conf.xml)
> >  <list name="domains" default="deny">
> >        <node type="allow" cidr="60.248.175.32/28"/>
> >       <!-- <node type="allow" domain="$${domain}"/>-->
> >  </list>
> >
> >  (dialplan/default.xml)
> >
> >  <extension name="Dial to 37">
> >        <condition field="destination_number"
> >  expression="^([0-4]0[01][0-9])$">
> >          <action application="bridge"
> >  data="sofia/internal/$1 at 60.248.175.37"/>
> >        </condition>
> >   </extension>
> >
> >  (dialplan/public.xml)
> >
> >  <extension name="Calls from 37">
> >        <condition field="destination_number"
> >  expression="^([0-4]0[01][0-9])$">
> >          <action application="transfer" data="$1 XML default"/>
> >        </condition>
> >   </extension>
> >
> > Thank you for your attension
> >
> > Best Regards
> >
> > Gary
> >
> >
> >
> > ________________________________
> > Date: Thu, 14 Oct 2010 20:02:16 -0700
> > From: msc at freeswitch.org
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >
> > Did you try using ACLs or with putting a user/gateway on each side?
> > Put your configs on pastebin.freeswitch.org.
> >
> > This is actually a simple exercise once you know what to do. Start by
> > deciding if you want to do authentication by digest or IP address.
> >
> > -MC
> >
> > 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
> >
> > But my question is
> >
> > I set two fs
> >
> > and I want to connect them
> >
> > I refer to http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
> >
> > it 's not working
> >
> > so I give my configuration already
> >
> > should I change some setting?
> >
> > Thank you for your attension
> >
> > Best Regards
> >
> > Gary
> >
> > ________________________________
> > Date: Thu, 14 Oct 2010 17:56:43 -0700
> > From: msc at freeswitch.org
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >
> > Yes! You don't need a gateway for this. If the users are all on the same
> > server then you just need to create the necessary files in
> > conf/directory/default/
> > You already have 1000.xml, 1001.xml, ... 1019.xml
> > You just need to add 2010.xml, 4001.xml, 3009.xml, etc.
> > Check out this article:
> > http://bit.ly/EpVrv
> >
> > It's a bit older, but the section on adding a new extension is very
> > accurate. In that example it shows how to add extension 1500 to your
> > directory and your dialplan. You can copy that example for all of your
> new
> > extension numbers.
> >
> > -MC
> >
> > 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
> >
> > Hello
> >
> > but I can call to each other on the same server
> >
> > for example
> >
> > I can use 1007 at 37  call 1013 at 37  ,  4001 at 37  , 2010 at 37 , 3009 at 37
> >
> >
> >  Thank you for your attension
> > Best Regards
> >  Gary
> >
> >
> >
> > ________________________________
> > Date: Thu, 14 Oct 2010 11:27:31 -0700
> > From: msc at freeswitch.org
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >
> > FYI, we have several resources to help
> > http://wiki.freeswitch.org/wiki/Regular_Expression
> > http://bit.ly/aijtAC
> >
> > :)
> >
> > -MC
> >
> > On Thu, Oct 14, 2010 at 11:18 AM, Jeffrey Leung <
> curriegrad2004 at gmail.com>
> > wrote:
> >
> > Dialplan's regex doesn't look right to me. You'll need to fix that
> > before you do anything else.
> >
> > I'd suggest this regex instead of the one you provided:
> > ^[0-4]0(10[1-9][0-9])$
> >
> > instead of this one
> > ^([0-4]0[01][0-9])$
> >
> > If you can't get the grasp of regular expressions, I'd highly suggest
> > you to pick up a book and read up on it. Knowing regular expressions
> > is a great skill to have ;)
> >
> > 2010/10/14 ? Chiang ?? Chih-Chung   Wybie <tear152 at hotmail.com>:
> >> I tried use 1013 at 38 calls to 1019 at 37
> >>
> >> show log below
> >>
> >> 2010-10-14 17:50:41.249876 [NOTICE] switch_channel.c:675 New Channel
> >> sofia/internal/1013 at 60.248.175.38[875fdeef-b94e-41fb-a621-ea005bbaedbd]
> >> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
> >> 1013->1019 in context public
> >> 2010-10-14 17:50:41.263809 [NOTICE] switch_ivr.c:1450 Transfer
> >> sofia/internal/1013 at 60.248.175.38 to XML[1019 at default]
> >> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing
> >> 1013->1019 in context default
> >> 2010-10-14 17:50:41.283904 [INFO] switch_core_session.c:1764 Sending
> early
> >> media
> >> 2010-10-14 17:50:41.285023 [NOTICE] mod_sofia.c:1994 Pre-Answer
> >> sofia/internal/1013 at 60.248.175.38!
> >>
> >> seems not work
> >>
> >> Thank you for you attension
> >> Best Regards
> >>
> >> Gary
> >>
> >>
> >>
> >> ________________________________
> >> From: david.ponzone at ipeva.fr
> >> To: freeswitch-users at lists.freeswitch.org
> >> Date: Thu, 14 Oct 2010 10:16:28 +0200
> >> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >>
> >> I think we would need a complete log from the target FS, in order to
> >> provide
> >> you with any efficient help.
> >> Thank
> >> David Ponzone  Direction Technique
> >> email: david.ponzone at ipeva.fr
> >> tel:      01 74 03 18 97
> >> gsm:   06 66 98 76 34
> >> Service Client IPeva
> >> tel:      0811 46 26 26
> >> www.ipeva.fr  -   www.ipeva-studio.com
> >> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> >> l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion
> >> non autoris?e est interdite. Tout message ?lectronique est susceptible
> >> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> >> s'il
> >> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> >> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> >>
> >>
> >>
> >> Le 14/10/2010 ? 09:55, ? Chiang ?? Chih-Chung Wybie a ?crit :
> >>
> >> I tried Connect Two FreeSWITCH
> >> Boxes  http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
> >> and my configuration :
> >>
> >> 60.248.175.37
> >>
> >> (autoload_configs/act.conf.xml)
> >> <list name="domains" default="deny">
> >>       <node type="allow" cidr="60.248.175.38/28"/>
> >>      <!-- <node type="allow" domain="$${domain}"/>-->
> >> </list>
> >>
> >> (dialplan/default.xml)
> >>
> >> <extension name="Dial to 38">
> >>       <condition field="destination_number"
> >> expression="^([0-4]0[01][0-9])$">
> >>         <action application="bridge"
> >> data="sofia/internal/$1 at 60.248.175.38"/>
> >>       </condition>
> >>  </extension>
> >>
> >> (dialplan/public.xml)
> >>
> >> <extension name="Calls from 38">
> >>       <condition field="destination_number"
> >> expression="^([0-4]0[01][0-9])$">
> >>         <action application="transfer" data="$1 XML default"/>
> >>       </condition>
> >>  </extension>
> >>
> >>
> >>
> >> 60.248.175.38
> >>
> >> (autoload_configs/act.conf.xml)
> >> <list name="domains" default="deny">
> >>       <node type="allow" cidr="60.248.175.37/28"/>
> >>      <!-- <node type="allow" domain="$${domain}"/>-->
> >> </list>
> >>
> >> (dialplan/default.xml)
> >>
> >> <extension name="Dial to 37">
> >>       <condition field="destination_number"
> >> expression="^([0-4]0[01][0-9])$">
> >>         <action application="bridge"
> >> data="sofia/internal/$1 at 60.248.175.37"/>
> >>       </condition>
> >>  </extension>
> >>
> >> (dialplan/public.xml)
> >>
> >> <extension name="Calls from 37">
> >>       <condition field="destination_number"
> >> expression="^([0-4]0[01][0-9])$">
> >>         <action application="transfer" data="$1 XML default"/>
> >>       </condition>
> >>  </extension>
> >>
> >> when I use 1007 at 38 calls to 1007 at 37
> >> it won't show errors but still not work
> >>
> >> how should I change something ?
> >>
> >> Thank you for your attension
> >>
> >> Best Regards
> >> Gary
> >>
> >>
> >>
> >>
> >>
> >>> Date: Wed, 13 Oct 2010 22:17:13 -0700
> >>> From: curriegrad2004 at gmail.com
> >>> To: freeswitch-users at lists.freeswitch.org
> >>> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >>>
> >>> You can create a SIP trunk between the two servers so calls can be
> >>> routed back and forth from the servers you've specified.
> >>>
> >>> 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
> >>> > I tried it
> >>> > but not working
> >>> > and show errors below
> >>> > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493 Cannot
> >>> > create
> >>> > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> >>> > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493 Cannot
> >>> > create
> >>> > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> >>> > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate
> Failed.
> >>> > Cause: USER_NOT_REGISTERED
> >>> >
> >>> > and for example
> >>> >  I use SIP server 60.248.175.38  user id=1007
> >>> > and I call to 1007 at 60.248.175.37    that will show these errors
> >>> >
> >>> > how should set more?
> >>> >
> >>> > thank you for your attension
> >>> >
> >>> > Best Regards
> >>> > Gary
> >>> >
> >>> > ________________________________
> >>> > Date: Thu, 14 Oct 2010 10:55:49 +0800
> >>> > From: xyangni at gmail.com
> >>> > To: freeswitch-users at lists.freeswitch.org
> >>> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >>> >
> >>> > I think you can reg at server A and than call B as a normal SIP call
> to
> >>> > external domain.
> >>> >
> >>> > 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
> >>> >
> >>> >
> >>> >
> >>> > Hello
> >>> >
> >>> > I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X-Lite
> to
> >>> > call
> >>> >
> >>> > my question is how can I call between these servers?
> >>> >
> >>> > if I can , how shoould I set in *.xml? or X-Lite?
> >>> >
> >>> > thank you for your attension
> >>> >
> >>> >
> >>> > Best Regards
> >>> > Gary
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>> >
> >>> > _______________________________________________ FreeSWITCH-users
> >>> > mailing
> >>> > list FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________ FreeSWITCH-users mailing
> >> list FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________ FreeSWITCH-users mailing
> > list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________ FreeSWITCH-users mailing
> > list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________ FreeSWITCH-users mailing
> > list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From mario_fs at mgtech.com  Fri Oct 15 09:54:47 2010
From: mario_fs at mgtech.com (Mario)
Date: Fri, 15 Oct 2010 09:54:47 -0700
Subject: [Freeswitch-users] early media problem?
In-Reply-To: <4CB7CA1E.2040507@mgtech.com>
References: <4CB7BFD9.3070000@mgtech.com>
	<AANLkTi=t0Fyaeu8=4N9MNN-Y-=X29EMGdpV-pUH69X1d@mail.gmail.com>
	<4CB7CA1E.2040507@mgtech.com>
Message-ID: <4CB88757.9070700@mgtech.com>

Update: I had the vendor go back to the old system. Both were asterisk
and the new one would not honor/play my early media. caller only heard
ringing. If anyone knows what the ITSP might need to change on their end
(asterisk) I would really appreciate it. They plan to move to the new
system. Thanks.
Mario

On 10/14/10 20:27, Mario wrote:
> http://pastebin.freeswitch.org/14241
> 
> My first pastebin! Thanks..
> 
> On 10/14/10 20:03, Michael Collins wrote:
>> pastebin the debug log of the call coming in so we can see exactly what
>> is happening.
>> -MC
>>
>> On Thu, Oct 14, 2010 at 7:43 PM, Mario <mario_fs at mgtech.com
>> <mailto:mario_fs at mgtech.com>> wrote:
>>
>>     I had an extension working fine that played a file, ringed, etc. to an
>>     inbound caller before we picked up. Trying to fix a problem the ITSP
>>     moved the account and now the caller only hears ringing, none of my
>>     stuff is heard even though it's shows as executed in trace. I looked on
>>     the wiki and tried adding <action application="ring_ready"/> at the top
>>     but that did not help. Anyone have any idea how to get all my work
>>     working again? Is there something I can change or the ITSP? Thanks.
>>     Mario


From jstricker at lightnex.com  Fri Oct 15 10:21:36 2010
From: jstricker at lightnex.com (Jeremy Stricker)
Date: Fri, 15 Oct 2010 11:21:36 -0600
Subject: [Freeswitch-users] valet_park timeout and spot announcement
In-Reply-To: <AANLkTi=F=YV+5vG6OvLTNAGJ9pFvRO131w_pcDkjk7Ee@mail.gmail.com>
References: <4CAB6A30.7070608@lightnex.com>
	<AANLkTi=F=YV+5vG6OvLTNAGJ9pFvRO131w_pcDkjk7Ee@mail.gmail.com>
Message-ID: <4CB88DA0.6060101@lightnex.com>

Michael,

Thank you very much for your time and effort on this!  I had to make a 
few changes to your code to make it work.  In the "global" extension 
section I had to add a backslash to the end of the first condition tag 
otherwise FS complained about nested conditions or missing </extension> 
tag (you'll also notice that I changed it to work for our parking setup 
of valet at 20 and spaces at 21-29):

       <condition field="${ani}" expression="^(1\d\d)$"/> <!-- include 
only 'parker' extensions -->
       <condition field="destination_number" expression="^(20)$">
         <anti-action application="hash" 
data="insert/valet_recall/${destination_number}/${uuid}"/>
       </condition>

In the valet_park extension sections, on the sched_api action, I had to 
change api_res to api_result, I had to add parenthesis to the sched_api 
function call, and I had to swap uuid_transfer and ${valet_uuid} to 
match the arguments I understood:

     <extension name="valet_park_retrieve">
       <condition field="destination_number" expression="^(2[1-9])$">
         <action application="answer"/>
         <action application="lua" data="cancel_valet_recall.lua 
valet_lot $1"/>
         <action application="valet_park" data="valet_lot $1"/>
       </condition>
     </extension>

     <extension name="valet_park_with_recall_timer">
       <condition field="destination_number" expression="^(20)$">
         <action application="answer"/>
         <action application="set" 
data="valet_uuid=${hash(select/valet_recall/${caller_id_number})}"/>
         <action application="log" data="INFO Pulled [${valet_uuid}] 
from hash valet_recall/${caller_id_number}"/>
         <!-- the magic happens on the next line. The +30 means 30 
seconds later x-fer the call back to the parker -->
         <action application="set" data="api_result=${sched_api(+30 
${valet_uuid} uuid_transfer ${valet_uuid} ${ani})}"/>
         <action application="valet_park" data="valet_lot auto in 21 29"/>
       </condition>
     </extension>

Upon making these changes, the recall works great when the original 
answerer makes the park.

If the call is transferred somewhere else first or picked up from the 
first park on a different extension and then parked again, it doesn't 
recall the second parker.  The error it gives when it fires the 
scheduled task is:

2010-10-15 11:16:21.131565 [DEBUG] mod_commands.c:3092 Command 
uuid_transfer(aca7ca8d-c33d-457e-a453-9f1e0899dad3 102):
-ERR No Such Channel!

2010-10-15 11:16:21.131565 [DEBUG] switch_scheduler.c:138 Deleting task 
22 sched_api_function (aca7ca8d-c33d-457e-a453-9f1e0899dad3)

I'm guessing that a new uuid is assigned when the call is transferred, 
but since we have already set the uuid variable we are looking at, the 
function call fails.  To be honest though, I'm not very familiar with 
this part of FS yet.

I will continue to test and provide feedback, but I think you're on to 
something good here.  The way it works right now will solve a majority 
of the lost calls in the system I'm working with presently.

Thanks again!
Jeremy

On 10/12/2010 06:58 PM, Michael Collins wrote:
> Jeremy,
>
> My apologies for taking so long to reply. I was kicking around ways to
> do this. Comments inline.
>
> On Tue, Oct 5, 2010 at 11:10 AM, Jeremy Stricker <jstricker at lightnex.com
> <mailto:jstricker at lightnex.com>> wrote:
>
>     Hello all,
>
>     Is there anyway to set a call timeout when a call is valet parked so
>     that it will ring back to the extension that parked it if it isn't
>     answered by the intended recipient?  We have an user who parks calls and
>     then blindly announces the parking spot over an intercom.  With this
>     setup there is a great risk of a caller getting stuck in park.  If there
>     isn't a timeout method, can anyone recommend any alternate setup that
>     would allow at least two parking spots with visual indication of calls
>     parked on the handsets (Aastra 6730i and 6757i).
>
>
> I couldn't find a simple way to do this but the following method worked
> for me in a lab environment. Assume that 6100 is the park extension and
> 6101-6199 are the parking stalls. Add these <condition> blocks to the
> end of your "global" extension (it needs to be executed for every call):
>
> <condition field="${ani}" expression="^(10\d\d)$"> <!-- include only
> 'parker' extensions -->
> <condition field="destination_number" expression="^(6100)$">
> <anti-action application="hash"
> data="insert/valet_recall/${destination_number}/${uuid}"/>
> </condition>
>
> Then create the valet_park extensions:
>
> <extension name="valet_park_retrieve">
> <condition field="destination_number" expression="^(61\d[1-9])$">
> <action application="answer"/>
> <action application="lua" data="cancel_valet_recall.lua valet_lot $1"/>
> <action application="valet_park" data="valet_lot $1"/>
> </condition>
> </extension>
>
> <extension name="valet_park_with_recall_timer">
> <condition field="destination_number" expression="^(6100)$">
> <action application="answer"/>
> <action application="set"
> data="valet_uuid=${hash(select/valet_recall/${caller_id_number})}"/>
> <action application="log" data="INFO Pulled [${valet_uuid}] from hash
> valet_recall/${caller_id_number}"/>
> <!-- the magic happens on the next line. The +30 means 30 seconds later
> x-fer the call back to the parker -->
> <action application="set" data="api_res=${sched_api +30 uuid_transfer
> ${valet_uuid} ${valet_uuid} ${ani}}"/>
> <action application="valet_park" data="valet_lot auto in 6101 6199"/>
> </condition>
> </extension>
>
> The trick is to use "sched_api" API to schedule the other leg to be
> transferred back to the parker. We use the hash API to store some
> information, namely the parker's extension number and the uuid of the
> parked leg. Change the +30 to however many seconds you want the call to
> be parked before recalling. Here's the catch: if someone does come along
> and grab that call out of the parking stall then we need to remove the
> scheduled transfer! I thought a Lua script would be the easiest way to
> handle that, so when the person picking up the parked call dials 6001
> e.g. then it launches a quickie Lua script that removes the sched_api
> from the task list. Here's the Lua script:
>
> -- cancel_valet_recall.lua
> --
> -- parse the valet lot in question and find the uuid for the extension
> -- remove the uuid from the sched task list since this call is now being
> unparked
> --
>
> valet_lot = argv[1]
> valet_ext = argv[2]
> uuid      = session:getVariable('uuid')
>
> --freeswitch.consoleLog('INFO','Lot: ' .. valet_lot .. ' , Ext: ' ..
> valet_ext .. ' , uuid: ' .. uuid .. "\n")
>
> api = freeswitch.API()
>
> if ( valet_lot == nil and valet_ext == nil ) then
>      -- improper args... (feel free to do some better error handling)
> else
>      -- sched_del uuid for the uuid in valet_lot, valet_ext
>      valet_info = api:executeString('valet_info ' .. valet_lot)
>      -- freeswitch.consoleLog("INFO","\n" .. valet_info .. "\n\n")
>
>      valet_data = string.match(valet_info,"<extension uuid=.->" ..
> valet_ext .. "</extension>")
>      -- freeswitch.consoleLog("INFO","uuid data line is: " .. valet_data
> .. "\n\n")
>
>      valet_uuid = string.gsub(valet_data,'<extension uuid="(.-)">' ..
> valet_ext .. "</extension>","%1")
>      --freeswitch.consoleLog("INFO","valet uuid is: " .. valet_uuid ..
> "\n\n")
>
>      -- perform the sched_del
>      api_res = api:executeString('sched_del ' .. valet_uuid)
>      --freeswitch.consoleLog("INFO","result of sched_del " .. valet_uuid
> .. " is " .. api_res .. "\n\n")
> end
>
> Throw this into conf/scripts/cancel_valet_recall.lua and give it a shot.
> Let me know how it goes. If it works and if no one has any obvious
> improvements then I will toss it up on the wiki as an example of how to
> do a valet_park recall timer.
>
> If you want to learn more about how this works then uncomment the log
> statements in the Lua script. Also after parking but before picking up
> the call go to the fs_cli and do "show tasks" and you'll see what the
> task looks like. I used the parked leg's uuid as the 'group id' in
> sched_api so that it is easy to find and remove when unparking the call.
>
> Have fun and let me know if this helps or not.
> -MC
>
>
>     Secondly, is there anyway to stop read-back of the parking spot on the
>     parkee's side of the call?  Currently, the parker performs an attended
>     transfer, waits for and hears the spot read-back.  The parkee hears MOH
>     while the transfer is being performed, but then the spot announcement
>     themselves as soon as the parker finishes the attended transfer by
>     hanging up.
>
>
> The parker is hanging up too soon. If he/she waits until he/she hears
> music and then completes the transfer it should be okay. I just tested this.
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Jeremy Stricker
LightNex Communications

e: jstricker at lightnex.com
p: 877-342-3768
w: www.lightnex.com


From msc at freeswitch.org  Fri Oct 15 10:32:29 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Oct 2010 10:32:29 -0700
Subject: [Freeswitch-users] Radius AAA
In-Reply-To: <COL109-W528C7DF8FD0E4D203FD866F4570@phx.gbl>
References: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>
	<AANLkTi=GiQaO-48BP=T8c-A+09sJMKzQH7BX=ewBdB8R@mail.gmail.com>
	<AANLkTikw8LGDqQki16D30b8JL0eJid2UOTs-bEaZSc2k@mail.gmail.com>
	<4CADC2E4.90608@gmail.com>
	<COL109-W528C7DF8FD0E4D203FD866F4570@phx.gbl>
Message-ID: <AANLkTi=oXjajMBSn8tJCT8chm6mM8E4XVzXVynDda_+O@mail.gmail.com>

search to make sure that the libfreeradius-client.so.2 file exists somewhere
on your system. If it does not then the freeradius-client install was not
successful.

-MC

On Fri, Oct 15, 2010 at 6:39 AM, Abid Saleem <abid_freeswitch at live.com>wrote:

>  Hi,
>
> I have tried making and installing mod_rad_auth but it gives me the
> following error when loading the module.
>
> 2010-10-15 17:58:02.088712 [CRIT] switch_loadable_module.c:882 Error
> Loading module /usr/local/freeswitch/mod/mod_rad_auth.so
> **libfreeradius-client.so.2: cannot open shared object file: No such file
> or directory**
>
> Could you please help what is the issue and any resolution available. I
> installed freeradius-client-1.1.6 already.
>
> Regards
> ------------
> Abid Saleem
> Sr. Product Manager
> Terminus Technologies
>
> > Date: Thu, 7 Oct 2010 17:53:56 +0500
> > From: nazim.aghabayov at gmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Radius AAA
> >
> > Hello Tihomir,
> >
> > I've just updated the Mod_rad_auth wiki with installation instructions.
> > Thanks a lot for the mod. I use it in production and it works just great!
> >
> > Regards,
> > Nazim
> >
> > On 10/07/2010 01:48 PM, Tihomir Culjaga wrote:
> > > On Thu, Oct 7, 2010 at 1:15 AM, Michael Collins<msc at freeswitch.org>
> wrote:
> > >
> > >> I'm afraid no such comprehensive documentation exists. Submissions to
> our
> > >> wiki are most welcomed. This is all we have at the moment:
> > >>
> > >> http://wiki.freeswitch.org/wiki/Mod_rad_auth
> > >>
> > >> Be sure to install freeradius2 (server) and freeradius-client before
> trying
> > >> to install mod_auth_rad.
> > >>
> > >> -MC
> > >>
> > >>
> > >>
> > > correct,
> > >
> > > Im the author of the module and im going to provide the documentation
> and
> > > how-to.
> > >
> > >
> > > in brief, this module does radius auth (not accounting). It is based on
> > > freeradius-client library and as such this is the only dependency.
> > >
> > > you can specify your own list of VSAs to be included in the packet
> along
> > > with the standard ones that are being used.
> > >
> > >
> > > name: just a description
> > > value: direct input or variable
> > > pec: vendor ID (0 for default, 9 for cisco...)
> > > expr: 1 for channel variable, 2 for direct input (string)
> > > direction: in for radius-request, out for radius-response
> > >
> > >
> > > Im not going to describe it here... its better i do it on the wiki
> itself...
> > >
> > >
> > >
> > >
> > > T.
> > >
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From javieraristizabal at gmail.com  Fri Oct 15 10:55:54 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Fri, 15 Oct 2010 12:55:54 -0500
Subject: [Freeswitch-users] Trouble to start fail2ban
Message-ID: <AANLkTi=5nZqN5yE=9Yk80aJuw5h3-UOf6dJTx_x46jrA@mail.gmail.com>

Hi folks,

I installed fail2ban CentOS 5.5. And i followed the wiki instructions to
configure fail2ban with FreeSWITCH. After i edit jail.conf with this config:

[freeswitch-tcp]

enabled  = true
port     = 5060,5061,5080,5081
protocol = tcp
filter   = freeswitch
logpath  = /usr/local/freeswitch/log/freeswitch.log

[freeswitch-udp]

enabled  = true
port     = 5060,5061,5080,5081
protocol = udp
filter   = freeswitch
logpath  = /usr/local/freeswitch/log/freeswitch.log
######

The fail2ban does not start, and the logs can not get anything.. Did anyone
have the same thing?

The freeswtich.conf:

# Fail2Ban configuration file
#
# Author: Rupa SChomaker
#

[Definition]

# Option:  failregex
# Notes.:  regex to match the password failures messages in the logfile. The
#          host must be matched by a group named "host". The tag "<HOST>"
can
#          be used for standard IP/hostname matching and is only an alias
for
#          (?:::f{4,6}:)?(?P<host>[\w\-.^_]+)
# Values:  TEXT
#
failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(REGISTER\) on
sofia profile \'\w+\' for \[.*\] from ip <HOST>

# Option:  ignoreregex
# Notes.:  regex to ignore. If this regex matches, the line is ignored.
# Values:  TEXT
#
ignoreregex =


Thanks

-- 
Javier Aristiz?bal
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From tculjaga at gmail.com  Fri Oct 15 10:56:10 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 15 Oct 2010 19:56:10 +0200
Subject: [Freeswitch-users] Radius AAA
In-Reply-To: <AANLkTi=oXjajMBSn8tJCT8chm6mM8E4XVzXVynDda_+O@mail.gmail.com>
References: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>
	<AANLkTi=GiQaO-48BP=T8c-A+09sJMKzQH7BX=ewBdB8R@mail.gmail.com>
	<AANLkTikw8LGDqQki16D30b8JL0eJid2UOTs-bEaZSc2k@mail.gmail.com>
	<4CADC2E4.90608@gmail.com>
	<COL109-W528C7DF8FD0E4D203FD866F4570@phx.gbl>
	<AANLkTi=oXjajMBSn8tJCT8chm6mM8E4XVzXVynDda_+O@mail.gmail.com>
Message-ID: <AANLkTi=P1QH2L5B-HckXvqRgHsQEpEia2X7mA_DDT=DN@mail.gmail.com>

On Fri, Oct 15, 2010 at 7:32 PM, Michael Collins <msc at freeswitch.org> wrote:

> search to make sure that the libfreeradius-client.so.2 file exists
> somewhere on your system. If it does not then the freeradius-client install
> was not successful.
>
> -MC
>
> I was short on time .. sorry


so:

1. find your library ...

updatedb && locate libfreeradius-client.s | grep lib

by default it goes into /usr/local/lib/ but you never know.


2. create a link to the library

cd /usr/local/freeswitch/lib/

ln -sf /usr/local/lib/libfreeradius-client.so.2 libfreeradius-client.so

please make sure you input the correct path to the library in question.


when you list your freeswitch/lib/ directory, you should have something like
this:

$ ls -l /usr/local/freeswitch/lib/ | grep radius
lrwxrwxrwx 1 root root      40 Mar  7  2010 libfreeradius-client.so.2 ->
/usr/local/lib/libfreeradius-client.so.2


the link should be there.

Now you can load mod_rad_auth without any issues.
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From daniel.neubert at solomo.de  Fri Oct 15 11:54:43 2010
From: daniel.neubert at solomo.de (Daniel Neubert)
Date: Fri, 15 Oct 2010 20:54:43 +0200
Subject: [Freeswitch-users] mod_cepstral 5.1 known issues?
In-Reply-To: <0FF607C13B7F7A42B5B80DA9EB204C9109D44BDD8E@srv-ex01-dal.intervoice.int>
References: <AANLkTik8KZdxAnUUfv0nKF6EkTHe559PtC2bF74eng11@mail.gmail.com>	<AANLkTi=MvWh14MQ9wPEboXZmMgAtURAY5zBoobn32Pkq@mail.gmail.com>	<AANLkTimj=SwX7d1sot+NhOxMgD4F-PN=bM3hxrh3bF3O@mail.gmail.com>	<4CB7F0C9.2040708@solomo.de>
	<0FF607C13B7F7A42B5B80DA9EB204C9109D44BDD8E@srv-ex01-dal.intervoice.int>
Message-ID: <4CB8A373.40504@solomo.de>

  Sure - what exactly do you need?

Best regards / Mit freundlichen Gr??en,
Daniel Neubert

Am 15.10.2010 13:32, schrieb Esser, Holger:
>
> Hi Daniel,
>
> May I ask which guide you used to set it up?
>
> Thanks,
>
> Holger
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Daniel Neubert
> *Sent:* Friday, October 15, 2010 1:12 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] mod_cepstral 5.1 known issues?
>
> I've exactly this setup which is in productive use for a few months now.
>
> Using cepstral directly on our FreeSWITCH nodes did not work (since 
> they are running on FreeBSD and Cepstral is only available for GNU Linux).
>
> Current setup is using Voice Katrin (German) Version 5.1.0 on Gentoo 
> Linux 64Bit via UniMRCP Server 1.0.0.
>
> Best regards / Mit freundlichen Gr??en,
> Daniel Neubert
>
>
> On 14.10.2010 17:58, Marc Kellem wrote:
>
> Is anyone using mod_unimrcp to access Cepstral 5.1 voices in a UniMRCP 
> server?
>
> On Thu, Oct 14, 2010 at 11:26 AM, Anthony Minessale 
> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
>
> random deadlocks in the cepstral engine blocking all the speech
> generation calls.
>
>
>
> On Thu, Oct 14, 2010 at 9:56 AM, Marc Kellem <mkellem at vontoo.com 
> <mailto:mkellem at vontoo.com>> wrote:
> > The mod_cepstral wiki page says "Warning: the suggested version to use
> > is 4.x since there are known issues with 5.1 (which is closed source)".
> >  What exactly are the known issues?
> > http://wiki.freeswitch.org/wiki/Mod_cepstral
> > Thanks,
> > Marc Kellem
>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org 
> <mailto:FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org 
> <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> No virus found in this incoming message.
> Checked by AVG - www.avg.com
> Version: 9.0.862 / Virus Database: 271.1.1/3192 - Release Date: 
> 10/14/10 13:34:00
>
>
> ------------------------------------------------------------------------
> This e-mail transmission may contain information that is proprietary, 
> privileged and/or confidential and is intended exclusively for the 
> person(s) to whom it is addressed. Any use, copying, retention or 
> disclosure by any person other than the intended recipient or the 
> intended recipient's designees is strictly prohibited. If you are the 
> intended recipient, you must treat the information in confidence and 
> in accordance with all laws related to the privacy and confidentiality 
> of such information. If you are not the intended recipient or their 
> designee, please notify the sender immediately by return e-mail and 
> delete all copies of this email, including all attachments.
>
> Intervoice, Inc. 17811 Waterview Parkway Dallas, TX 75252 USA
>
> Intervoice Limited, Registered in England and Wales with number 
> 2601740, 50 Park Road, Gatley, Cheshire, SK8 4HZ. VAT Number: 560421375
>
> Intervoice GmbH, Hagenauer Stra?e 55, 65203 Wiesbaden, Sitz der 
> Gesellschaft: Wiesbaden, Handelsregister: HRB 8186 (Amtsgericht 
> Wiesbaden), Gesch?ftsf?hrer: Wayne Barclay, Steffen Selbmann
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From freeswitch at tlainvestments.com  Fri Oct 15 11:53:31 2010
From: freeswitch at tlainvestments.com (Troy Anderson)
Date: Fri, 15 Oct 2010 11:53:31 -0700
Subject: [Freeswitch-users] Read only filesystem
Message-ID: <0925E7C5-987F-4065-85E1-2B45528D4DF1@tlainvestments.com>

I certainly don't know if this is a freeswitch specific issue, but we have several systems in the field running freeswitch on top of CentOS 5.5 and others on Ubuntu 10.04.  Some of the systems, after a weeks of performance, go into a read only state on the hard drive, at which point, a restart is the only fix.  After the reboot, the drive seems OK - I've run smartctl to verify that all seems good.  Sometimes, in readonly mode, it even gives an I/O error when trying to execute the /sbin/shutdown and we have to manually reboot.

At this point, I am trying to figure out what may be the root cause.  Has anyone else experienced something this?  Any suggestions?

Thanks,
Troy




From trob at freemail.hu  Fri Oct 15 10:28:55 2010
From: trob at freemail.hu (trob)
Date: Fri, 15 Oct 2010 10:28:55 -0700 (PDT)
Subject: [Freeswitch-users] mod_limit does not reset count when call
	gets to voicemail?
In-Reply-To: <AANLkTinYM15iDznFVS9mmomf367QT4g0vb7NKzwrd-px@mail.gmail.com>
References: <AANLkTinYM15iDznFVS9mmomf367QT4g0vb7NKzwrd-px@mail.gmail.com>
Message-ID: <1287163735627-5639685.post@n2.nabble.com>


Hi

Have you got the solution for this problem?
It does not work for me too.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-limit-does-not-reset-count-when-call-gets-to-voicemail-tp5578428p5639685.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jerre at j-cope.com  Fri Oct 15 11:12:39 2010
From: jerre at j-cope.com (Jerre Cope)
Date: Fri, 15 Oct 2010 13:12:39 -0500
Subject: [Freeswitch-users] tls/ssl gateway insists on registering SIP
Message-ID: <4CB89997.80305@j-cope.com>

I'm attempting to implement the Hybrid model as shown on

http://wiki.freeswitch.org/wiki/Tls#Hybrid_Encryption

FS(Truck)=ssl/tls=>FS(Office)

My idea was to create a dial plan so that when FS(Truck) dials 999, a 
secure connection could be made to the office.

The pastebin shows my directory entry in FS(Office) and the external 
gateway in FS(truck)

http://pastebin.freeswitch.org/14245

The gateway insists on connecting on 5060, calls made through the 
gateway appear in tshark to be using tls.  I can block 5060 at the 
FS(office) firewall after registration and make calls through the 
gateway OK until FS(truck) tries to register again.

The FS(truck) default gateway is voip.ms, the idea being non-secured 
calls would go out the voip.ms gateway so as to limit the traffic to 
FS(office).

I'm operating way beyond my experience, so my attempt at this may be a 
goofy start.  I'm asking for a little direction/perspective on how to 
implement.  I'm using ssl because all my FS(trucks) will be dynamic IPs.

Thanks for all the really cool programming/documentation so far.  I'm 
having great fun!





From vzeljezn at pbf.hr  Fri Oct 15 12:29:02 2010
From: vzeljezn at pbf.hr (Vedran Zeljeznak)
Date: Fri, 15 Oct 2010 21:29:02 +0200
Subject: [Freeswitch-users] Java_ESL_Client - unhandled exception caught
In-Reply-To: <AANLkTikgEBB5c4AtVZKkSFTjD62e8s_ws8tapmkebp+D@mail.gmail.com>
References: <AANLkTik4J3bbgSGjF-pkUtx6Jxcewxk+P_5sf2RowEUS@mail.gmail.com>
	<AANLkTikgEBB5c4AtVZKkSFTjD62e8s_ws8tapmkebp+D@mail.gmail.com>
Message-ID: <AANLkTikw4POs8MhAOrAhdQP9CWYdBj=PccOw89-jURDC@mail.gmail.com>

On Fri, Oct 15, 2010 at 1:17 AM, david varnes <david.varnes at gmail.com> wrote:
> Vedran,
> Thanks for reporting the issue. ?As you can see from the stack trace
> this is just a call to the regular java URL decoder.

Yes i've seen that. I've reported the issue so you can add that
exception to the try/catch block and "bulletproof it" some more...
:-).

> Can you capture
> the content of the event or message that is causing this to fail ?
>
> Or at the least what the cause event is so that I can determine if
> it is still a problem on trunk ?

If it happens again i'll send it.

I've also noticed that in some cases thread or threads from executors
(set from NioClientSocketChannelFactory constructor) hang when i use
Java_ESL_Client to connect to FS event socket. This behavior happened
to me once before when i didn't call close() method on client channel,
sometimes bootstrap.releaseExternalResources() helped with this issue.

regards,
---
Vedran Zeljeznak


From msc at freeswitch.org  Fri Oct 15 14:12:36 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Oct 2010 14:12:36 -0700
Subject: [Freeswitch-users] mod_limit does not reset count when call
 gets to voicemail?
In-Reply-To: <1287163735627-5639685.post@n2.nabble.com>
References: <AANLkTinYM15iDznFVS9mmomf367QT4g0vb7NKzwrd-px@mail.gmail.com>
	<1287163735627-5639685.post@n2.nabble.com>
Message-ID: <AANLkTinAda368joMh+yDBEsGwZNEmLknxU8YuNZ5sVDx@mail.gmail.com>

Could you explain a bit more about your scenario?
-MC

On Fri, Oct 15, 2010 at 10:28 AM, trob <trob at freemail.hu> wrote:

>
> Hi
>
> Have you got the solution for this problem?
> It does not work for me too.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/mod-limit-does-not-reset-count-when-call-gets-to-voicemail-tp5578428p5639685.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Fri Oct 15 14:14:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 Oct 2010 16:14:30 -0500
Subject: [Freeswitch-users] UI Developer Job for CudaTel
Message-ID: <AANLkTi=GphssE-WvkaKN4HRW=c7afPf2QK73O6RY4sFo@mail.gmail.com>

We're expanding our horizons again over at CudaTel and hiring some
more developers.
Anyone interested, who lives in WI, OK, CA or MI or willing to move to
one of those should apply.

Looking for people with good HTML Design instincts and JavaScript /
Jquery skills.

Full-time job with challenging/rewarding work.

Send inquires to jobs at freeswitch.org with a resume and any example work.



-- 
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


From xyangni at gmail.com  Fri Oct 15 14:58:31 2010
From: xyangni at gmail.com (xuyan yang)
Date: Sat, 16 Oct 2010 05:58:31 +0800
Subject: [Freeswitch-users] iLBC codec generating only noise.
In-Reply-To: <1286988590331-5631742.post@n2.nabble.com>
References: <AANLkTinz4JEfyGnjU-4nneOFg=A0qXHjVw9xE=jp2qSo@mail.gmail.com>
	<AANLkTinvkxRaS+Wmso1vubuH3Pj4Mib96B32Bx+sqR7H@mail.gmail.com>
	<AANLkTimmMM8Ua9AP9BwXU0yDfP9FReK2e9w-iMRiMVFu@mail.gmail.com>
	<AANLkTimiSPgCzs31VjEp0x8g1qDe-kzppsDM7uig0YGU@mail.gmail.com>
	<AANLkTikFDu0xC=VcXpzJ0_8i=NL-QfgBm5ms7nMftRzu@mail.gmail.com>
	<AANLkTi=6Esu59uFuRMvxd-RQ6B_fiSAkXS_Qx4Zu-sfF@mail.gmail.com>
	<1286988590331-5631742.post@n2.nabble.com>
Message-ID: <AANLkTi==N1v9ASfz6CYqhY6+5zkWS14bQ=dJec4-X+rb@mail.gmail.com>

Unfortunately, after recent update, the git head began to produce noise on
iLBC for Linux platform too.

On Thu, Oct 14, 2010 at 12:49 AM, Jeff Lenk <jeff at jefflenk.com> wrote:

>
> Please open a bug report with the specific build environment and details of
> how to reproduce the problem on windows. - Thanks.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/iLBC-codec-generating-only-noise-tp5616210p5631742.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Fri Oct 15 15:27:34 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 15 Oct 2010 23:27:34 +0100
Subject: [Freeswitch-users] Read only filesystem
In-Reply-To: <0925E7C5-987F-4065-85E1-2B45528D4DF1@tlainvestments.com>
References: <0925E7C5-987F-4065-85E1-2B45528D4DF1@tlainvestments.com>
Message-ID: <AANLkTikU1Uuc8JLSsHO77hfKLjhvhpXn0EZXmn3mY0kH@mail.gmail.com>

Sounds like hardware level, not FS-specific.

Every time I've had this it's been a bad drive or bad motherboard/raid
controller.

Try checking the drive for bad blocks. Since it's multiple servers
perhaps it could also be a driver bug.

Regards,
-Steve



On 15 October 2010 19:53, Troy Anderson <freeswitch at tlainvestments.com> wrote:
> I certainly don't know if this is a freeswitch specific issue, but we have several systems in the field running freeswitch on top of CentOS 5.5 and others on Ubuntu 10.04. ?Some of the systems, after a weeks of performance, go into a read only state on the hard drive, at which point, a restart is the only fix. ?After the reboot, the drive seems OK - I've run smartctl to verify that all seems good. ?Sometimes, in readonly mode, it even gives an I/O error when trying to execute the /sbin/shutdown and we have to manually reboot.
>
> At this point, I am trying to figure out what may be the root cause. ?Has anyone else experienced something this? ?Any suggestions?
>
> Thanks,
> Troy
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mkellem at vontoo.com  Fri Oct 15 16:22:28 2010
From: mkellem at vontoo.com (Marc Kellem)
Date: Fri, 15 Oct 2010 19:22:28 -0400
Subject: [Freeswitch-users] mod_cepstral 5.1 known issues?
In-Reply-To: <0FF607C13B7F7A42B5B80DA9EB204C9109D44BDD8E@srv-ex01-dal.intervoice.int>
References: <AANLkTik8KZdxAnUUfv0nKF6EkTHe559PtC2bF74eng11@mail.gmail.com>
	<AANLkTi=MvWh14MQ9wPEboXZmMgAtURAY5zBoobn32Pkq@mail.gmail.com>
	<AANLkTimj=SwX7d1sot+NhOxMgD4F-PN=bM3hxrh3bF3O@mail.gmail.com>
	<4CB7F0C9.2040708@solomo.de>
	<0FF607C13B7F7A42B5B80DA9EB204C9109D44BDD8E@srv-ex01-dal.intervoice.int>
Message-ID: <AANLkTi=cAW7Ru3KPy4LH7bwy0eGKce0q5z-yztF-Nhd2@mail.gmail.com>

Holger, this JIRA issue might answer some of your questions.

http://jira.freeswitch.org/browse/FS-10



On Fri, Oct 15, 2010 at 7:32 AM, Esser, Holger
<Holger.Esser at convergys.com>wrote:

>  Hi Daniel,
>
>
>
> May I ask which guide you used to set it up?
>
>
>
> Thanks,
>
> Holger
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Daniel
> Neubert
> *Sent:* Friday, October 15, 2010 1:12 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] mod_cepstral 5.1 known issues?
>
>
>
> I've exactly this setup which is in productive use for a few months now.
>
> Using cepstral directly on our FreeSWITCH nodes did not work (since they
> are running on FreeBSD and Cepstral is only available for GNU Linux).
>
> Current setup is using Voice Katrin (German) Version 5.1.0 on Gentoo Linux
> 64Bit via UniMRCP Server 1.0.0.
>
>  Best regards / Mit freundlichen Gr??en,
> Daniel Neubert
>
>
> On 14.10.2010 17:58, Marc Kellem wrote:
>
> Is anyone using mod_unimrcp to access Cepstral 5.1 voices in a UniMRCP
> server?
>
>
>
> On Thu, Oct 14, 2010 at 11:26 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> random deadlocks in the cepstral engine blocking all the speech
> generation calls.
>
>
>
> On Thu, Oct 14, 2010 at 9:56 AM, Marc Kellem <mkellem at vontoo.com> wrote:
> > The mod_cepstral wiki page says "Warning: the suggested version to use
> > is 4.x since there are known issues with 5.1 (which is closed source)".
> >  What exactly are the known issues?
> > http://wiki.freeswitch.org/wiki/Mod_cepstral
> > Thanks,
> > Marc Kellem
>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> No virus found in this incoming message.
> Checked by AVG - www.avg.com
> Version: 9.0.862 / Virus Database: 271.1.1/3192 - Release Date: 10/14/10
> 13:34:00
>
> ------------------------------
> This e-mail transmission may contain information that is proprietary,
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> to whom it is addressed. Any use, copying, retention or disclosure by any
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> Gesellschaft: Wiesbaden, Handelsregister: HRB 8186 (Amtsgericht Wiesbaden),
> Gesch?ftsf?hrer: Wayne Barclay, Steffen Selbmann
>
> _______________________________________________
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From Nabble at slickdeals.endjunk.com  Fri Oct 15 16:34:25 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 15 Oct 2010 16:34:25 -0700 (PDT)
Subject: [Freeswitch-users] Read only filesystem
In-Reply-To: <0925E7C5-987F-4065-85E1-2B45528D4DF1@tlainvestments.com>
References: <0925E7C5-987F-4065-85E1-2B45528D4DF1@tlainvestments.com>
Message-ID: <1287185665565-5640809.post@n2.nabble.com>



Troy Anderson wrote:
> At this point, I am trying to figure out what may be the root cause.  Has
> anyone else experienced something this?  Any suggestions?
Your best bet on this is to look at the log file (/var/log/messages and
among others in the same sub-dir) and see if you can identify what triggered
the HD into a RO mode.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Read-only-filesystem-tp5640425p5640809.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From dujinfang at gmail.com  Fri Oct 15 20:12:40 2010
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 16 Oct 2010 11:12:40 +0800
Subject: [Freeswitch-users] FreeSWITCH HA question&talk
Message-ID: <AANLkTimG_V=pBbSzf2hr=3KUex_Jx9GB=E5sDWSqCmcS@mail.gmail.com>

Hi,

I tested FS HA, it works like a charm. however, I'd like to talk about
some more:

I tested on my local computer by enabling tack-calls.  make a call and
kill FS, restart FS and execute sofia recover.

1) originate user/1000 &record(/tmp/blah.wav)

Recover works, however, it overwrote the recording. Added
RECORD_APPEND=true made it work.

But for other APPs like socket, it might make a new socket to a
outbound socket, can it send a var like channel_recovered so a
outbound socket can aware of a recover?


2) call from 1000 to 1001

Recover works. If I also do uuid_record, the recording cannot recover.
Seems there are no infos in sip_recovery table.

Neither does record_session, though there's an entry in sip_recovery
table. I guess it might be possible to get this work.

If it can send a channel_recovered event, then it might be possible to
continue the record in outside logic(say, event socket).

3) call from 1000 to a gateway outside (NAT-ed), leg-a can recover,
leg-b failed with:

2010-10-16 10:28:17.162241 [ERR] sofia_glue.c:3190 AUDIO RTP REPORTS
ERROR: [Bind Error!]

I guess it is caused by NAT. (since I kill & restart FS manually, it
may take 10 seconds). Is it possible to change the router settings? Or
could FS send a re-Invite to re-establish media?


Below are some fantastic thoughts:

4) N + M redundancy

Default FS HA is designed to 1+1 redundancy, it might be interesting
for N(active)+M(standby) in large farms. If one active FS fails, bind
the float IP to a standby server and go.

5) N (N>2) servers without float IP

Let's say each client UA register to 2 (or more) servers, If one
server fails, a FS controller then re-call the client in an active
call through a pair-server and the client responsible to hangup the
originate call and automatically answer the new one. This need client
support of course.  :)


-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From b_ball_henry at hotmail.com  Sat Oct 16 00:41:46 2010
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Sat, 16 Oct 2010 15:41:46 +0800
Subject: [Freeswitch-users] FreeSWITCH HA question&talk
In-Reply-To: <AANLkTimG_V=pBbSzf2hr=3KUex_Jx9GB=E5sDWSqCmcS@mail.gmail.com>
References: <AANLkTimG_V=pBbSzf2hr=3KUex_Jx9GB=E5sDWSqCmcS@mail.gmail.com>
Message-ID: <AANLkTi=Z7q4n+tKMvBWiXWezNfLXy3-jQi--giKRuawB@mail.gmail.com>

For the FS HA, is there a wiki page you are referring to?

Henry

On Sat, Oct 16, 2010 at 11:12 AM, Seven Du <dujinfang at gmail.com> wrote:

> Hi,
>
> I tested FS HA, it works like a charm. however, I'd like to talk about
> some more:
>
> I tested on my local computer by enabling tack-calls.  make a call and
> kill FS, restart FS and execute sofia recover.
>
> 1) originate user/1000 &record(/tmp/blah.wav)
>
> Recover works, however, it overwrote the recording. Added
> RECORD_APPEND=true made it work.
>
> But for other APPs like socket, it might make a new socket to a
> outbound socket, can it send a var like channel_recovered so a
> outbound socket can aware of a recover?
>
>
> 2) call from 1000 to 1001
>
> Recover works. If I also do uuid_record, the recording cannot recover.
> Seems there are no infos in sip_recovery table.
>
> Neither does record_session, though there's an entry in sip_recovery
> table. I guess it might be possible to get this work.
>
> If it can send a channel_recovered event, then it might be possible to
> continue the record in outside logic(say, event socket).
>
> 3) call from 1000 to a gateway outside (NAT-ed), leg-a can recover,
> leg-b failed with:
>
> 2010-10-16 10:28:17.162241 [ERR] sofia_glue.c:3190 AUDIO RTP REPORTS
> ERROR: [Bind Error!]
>
> I guess it is caused by NAT. (since I kill & restart FS manually, it
> may take 10 seconds). Is it possible to change the router settings? Or
> could FS send a re-Invite to re-establish media?
>
>
> Below are some fantastic thoughts:
>
> 4) N + M redundancy
>
> Default FS HA is designed to 1+1 redundancy, it might be interesting
> for N(active)+M(standby) in large farms. If one active FS fails, bind
> the float IP to a standby server and go.
>
> 5) N (N>2) servers without float IP
>
> Let's say each client UA register to 2 (or more) servers, If one
> server fails, a FS controller then re-call the client in an active
> call through a pair-server and the client responsible to hangup the
> originate call and automatically answer the new one. This need client
> support of course.  :)
>
>
> --
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From stevendt at primrosebank.net  Sat Oct 16 02:10:14 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sat, 16 Oct 2010 10:10:14 +0100
Subject: [Freeswitch-users] Passing incoming Caller ID
Message-ID: <9C6BF151E8B54010923546BBF63C877E@bp1.ad.bp.com>

Hi,

can someone help me with passing caller ID to FreeSwitch extensions please?

I have a Linksys SPA-3102 setup to receive PSTN calls and Caller ID has just been configured on the line.
The Gateway is setup as Extension 1000 with Dialplan entries
      <variable name="effective_caller_id_name" value="PSTN Line"/>
      <variable name="effective_caller_id_number" value="1000"/>

I have a group name (primrose) setup (in directory\default.xml) to ring a number of phones
I have extension 100 configured with 
      <variable name="callgroup" value="primrose"/>

The Gateway is setup to dial extension 100 on an incoming call

When a call comes in, the Gateway sees the Caller ID, e.g, 07751234567

And FreeSwitch see the caller ID from the gateway :-

[NOTICE] switch_channel.c:669 New Channel sofia/internal/07751234567 at 192.168.1.181 [f863c03f-0f7e-4e4b-beb0-dfa57dad2b50]


When the group extension ring, the caller is shown as "PSTN Line" on the extensions.


Can someone tell me how & where I can modify the config to pick up the CID from the Gateway and pass it to the group extensions please ?

regards
Dave
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From mnhassan at usa.net  Sat Oct 16 03:28:47 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Sat, 16 Oct 2010 16:28:47 +0600
Subject: [Freeswitch-users] Passing incoming Caller ID
In-Reply-To: <9C6BF151E8B54010923546BBF63C877E@bp1.ad.bp.com>
References: <9C6BF151E8B54010923546BBF63C877E@bp1.ad.bp.com>
Message-ID: <AANLkTi=V7sAMxmDFmLUf_bEgrxz6nOwXjnXL9OYsk5qN@mail.gmail.com>

You have to use the variables in that line that sets caller name. I
don't remember on top of my head which. You can use the dialplan
application called "info" which will give you a nice list of useful
variables and other information on the call and the channel.

Regards
HASSAN


On 2010-10-16, Dave Stevenson <stevendt at primrosebank.net> wrote:
> Hi,
>
> can someone help me with passing caller ID to FreeSwitch extensions please?
>
> I have a Linksys SPA-3102 setup to receive PSTN calls and Caller ID has just
> been configured on the line.
> The Gateway is setup as Extension 1000 with Dialplan entries
>       <variable name="effective_caller_id_name" value="PSTN Line"/>
>       <variable name="effective_caller_id_number" value="1000"/>
>
> I have a group name (primrose) setup (in directory\default.xml) to ring a
> number of phones
> I have extension 100 configured with
>       <variable name="callgroup" value="primrose"/>
>
> The Gateway is setup to dial extension 100 on an incoming call
>
> When a call comes in, the Gateway sees the Caller ID, e.g, 07751234567
>
> And FreeSwitch see the caller ID from the gateway :-
>
> [NOTICE] switch_channel.c:669 New Channel
> sofia/internal/07751234567 at 192.168.1.181
> [f863c03f-0f7e-4e4b-beb0-dfa57dad2b50]
>
>
> When the group extension ring, the caller is shown as "PSTN Line" on the
> extensions.
>
>
> Can someone tell me how & where I can modify the config to pick up the CID
> from the Gateway and pass it to the group extensions please ?
>
> regards
> Dave
>

-- 
Sent from my mobile device


From stevendt at primrosebank.net  Sat Oct 16 04:24:59 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sat, 16 Oct 2010 12:24:59 +0100
Subject: [Freeswitch-users] Passing incoming Caller ID
References: <9C6BF151E8B54010923546BBF63C877E@bp1.ad.bp.com>
	<AANLkTi=V7sAMxmDFmLUf_bEgrxz6nOwXjnXL9OYsk5qN@mail.gmail.com>
Message-ID: <5CE1E7BE88EA4679851DBFFBEFE315F7@bp1.ad.bp.com>

Hi Nassan,

thanks for the pointer - I can see those variables now, including :-

Channel-Presence-ID: [07751234567 at 192.168.1.181]
Caller-Username: [07751234567]
Caller-Caller-ID-Number: [07751234567]
Caller-ANI: [07751234567]
Caller-Channel-Name: [sofia/internal/07751234567 at 192.168.1.181]
variable_sip_from_user: [07751234567]
variable_sip_from_uri: [07751234567 at 192.168.1.181]
variable_sip_from_user_stripped: [07751234567]
variable_sip_full_from: [PSTN Line 
<sip:07751234567 at 192.168.1.181>;tag=26bdee1e605c89ebo1]
variable_sip_contact_user: [07751234567]
variable_sip_contact_uri: [07751234567 at 192.168.1.183:5061]
variable_channel_name: [sofia/internal/07751234567 at 192.168.1.181]
variable_presence_id: [07751234567 at 192.168.1.181]

Lots of places where I can pick up the number from, but in the interests of 
"doing it right", I'd like to pick the "proper" (most reliable) one. Again, 
this call is coming from PSTN via a gateway, so it looks like the best one 
might be "Caller-Caller-ID-Number" ?

I'm not sure where to copy the relevant variable to though ?

regards
Dave


----- Original Message ----- 
From: "Nyamul Hassan" <mnhassan at usa.net>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, October 16, 2010 11:28 AM
Subject: Re: [Freeswitch-users] Passing incoming Caller ID


> You have to use the variables in that line that sets caller name. I
> don't remember on top of my head which. You can use the dialplan
> application called "info" which will give you a nice list of useful
> variables and other information on the call and the channel.
>
> Regards
> HASSAN
>
>
> On 2010-10-16, Dave Stevenson <stevendt at primrosebank.net> wrote:
>> Hi,
>>
>> can someone help me with passing caller ID to FreeSwitch extensions 
>> please?
>>
>> I have a Linksys SPA-3102 setup to receive PSTN calls and Caller ID has 
>> just
>> been configured on the line.
>> The Gateway is setup as Extension 1000 with Dialplan entries
>>       <variable name="effective_caller_id_name" value="PSTN Line"/>
>>       <variable name="effective_caller_id_number" value="1000"/>
>>
>> I have a group name (primrose) setup (in directory\default.xml) to ring a
>> number of phones
>> I have extension 100 configured with
>>       <variable name="callgroup" value="primrose"/>
>>
>> The Gateway is setup to dial extension 100 on an incoming call
>>
>> When a call comes in, the Gateway sees the Caller ID, e.g, 07751234567
>>
>> And FreeSwitch see the caller ID from the gateway :-
>>
>> [NOTICE] switch_channel.c:669 New Channel
>> sofia/internal/07751234567 at 192.168.1.181
>> [f863c03f-0f7e-4e4b-beb0-dfa57dad2b50]
>>
>>
>> When the group extension ring, the caller is shown as "PSTN Line" on the
>> extensions.
>>
>>
>> Can someone tell me how & where I can modify the config to pick up the 
>> CID
>> from the Gateway and pass it to the group extensions please ?
>>
>> regards
>> Dave
>>
>
> -- 
> Sent from my mobile device
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From mnhassan at usa.net  Sat Oct 16 05:07:43 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Sat, 16 Oct 2010 18:07:43 +0600
Subject: [Freeswitch-users] Passing incoming Caller ID
In-Reply-To: <5CE1E7BE88EA4679851DBFFBEFE315F7@bp1.ad.bp.com>
References: <9C6BF151E8B54010923546BBF63C877E@bp1.ad.bp.com>
	<AANLkTi=V7sAMxmDFmLUf_bEgrxz6nOwXjnXL9OYsk5qN@mail.gmail.com>
	<5CE1E7BE88EA4679851DBFFBEFE315F7@bp1.ad.bp.com>
Message-ID: <AANLkTi=+_b8vfiL_popFsyoxv0bKhfyoYyv2AYFw6dxu@mail.gmail.com>

The caller_id_number is the best bet for the Caller ID. But the
display on your extension shows what you set for caller name (PSTN
Line). So I guess you need to set the caller name to
${caller_id_number}.

Regards
HASSAN


On 2010-10-16, Dave Stevenson <stevendt at primrosebank.net> wrote:
> Hi Nassan,
>
> thanks for the pointer - I can see those variables now, including :-
>
> Channel-Presence-ID: [07751234567 at 192.168.1.181]
> Caller-Username: [07751234567]
> Caller-Caller-ID-Number: [07751234567]
> Caller-ANI: [07751234567]
> Caller-Channel-Name: [sofia/internal/07751234567 at 192.168.1.181]
> variable_sip_from_user: [07751234567]
> variable_sip_from_uri: [07751234567 at 192.168.1.181]
> variable_sip_from_user_stripped: [07751234567]
> variable_sip_full_from: [PSTN Line
> <sip:07751234567 at 192.168.1.181>;tag=26bdee1e605c89ebo1]
> variable_sip_contact_user: [07751234567]
> variable_sip_contact_uri: [07751234567 at 192.168.1.183:5061]
> variable_channel_name: [sofia/internal/07751234567 at 192.168.1.181]
> variable_presence_id: [07751234567 at 192.168.1.181]
>
> Lots of places where I can pick up the number from, but in the interests of
> "doing it right", I'd like to pick the "proper" (most reliable) one. Again,
> this call is coming from PSTN via a gateway, so it looks like the best one
> might be "Caller-Caller-ID-Number" ?
>
> I'm not sure where to copy the relevant variable to though ?
>
> regards
> Dave
>
>
> ----- Original Message -----
> From: "Nyamul Hassan" <mnhassan at usa.net>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Saturday, October 16, 2010 11:28 AM
> Subject: Re: [Freeswitch-users] Passing incoming Caller ID
>
>
>> You have to use the variables in that line that sets caller name. I
>> don't remember on top of my head which. You can use the dialplan
>> application called "info" which will give you a nice list of useful
>> variables and other information on the call and the channel.
>>
>> Regards
>> HASSAN
>>
>>
>> On 2010-10-16, Dave Stevenson <stevendt at primrosebank.net> wrote:
>>> Hi,
>>>
>>> can someone help me with passing caller ID to FreeSwitch extensions
>>> please?
>>>
>>> I have a Linksys SPA-3102 setup to receive PSTN calls and Caller ID has
>>> just
>>> been configured on the line.
>>> The Gateway is setup as Extension 1000 with Dialplan entries
>>>       <variable name="effective_caller_id_name" value="PSTN Line"/>
>>>       <variable name="effective_caller_id_number" value="1000"/>
>>>
>>> I have a group name (primrose) setup (in directory\default.xml) to ring a
>>> number of phones
>>> I have extension 100 configured with
>>>       <variable name="callgroup" value="primrose"/>
>>>
>>> The Gateway is setup to dial extension 100 on an incoming call
>>>
>>> When a call comes in, the Gateway sees the Caller ID, e.g, 07751234567
>>>
>>> And FreeSwitch see the caller ID from the gateway :-
>>>
>>> [NOTICE] switch_channel.c:669 New Channel
>>> sofia/internal/07751234567 at 192.168.1.181
>>> [f863c03f-0f7e-4e4b-beb0-dfa57dad2b50]
>>>
>>>
>>> When the group extension ring, the caller is shown as "PSTN Line" on the
>>> extensions.
>>>
>>>
>>> Can someone tell me how & where I can modify the config to pick up the
>>> CID
>>> from the Gateway and pass it to the group extensions please ?
>>>
>>> regards
>>> Dave
>>>
>>
>> --
>> Sent from my mobile device
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device


From stevendt at primrosebank.net  Sat Oct 16 06:11:43 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sat, 16 Oct 2010 14:11:43 +0100
Subject: [Freeswitch-users] Passing incoming Caller ID
References: <9C6BF151E8B54010923546BBF63C877E@bp1.ad.bp.com><AANLkTi=V7sAMxmDFmLUf_bEgrxz6nOwXjnXL9OYsk5qN@mail.gmail.com><5CE1E7BE88EA4679851DBFFBEFE315F7@bp1.ad.bp.com>
	<AANLkTi=+_b8vfiL_popFsyoxv0bKhfyoYyv2AYFw6dxu@mail.gmail.com>
Message-ID: <12B86B97B509492D88540BAD85842B7B@bp1.ad.bp.com>

Hi again Hassan

great ! - Works a treat - thanks a lot!

regards
Dave



----- Original Message ----- 
From: "Nyamul Hassan" <mnhassan at usa.net>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, October 16, 2010 1:07 PM
Subject: Re: [Freeswitch-users] Passing incoming Caller ID


> The caller_id_number is the best bet for the Caller ID. But the
> display on your extension shows what you set for caller name (PSTN
> Line). So I guess you need to set the caller name to
> ${caller_id_number}.
>
> Regards
> HASSAN
>
>
> On 2010-10-16, Dave Stevenson <stevendt at primrosebank.net> wrote:
>> Hi Nassan,
>>
>> thanks for the pointer - I can see those variables now, including :-
>>
>> Channel-Presence-ID: [07751234567 at 192.168.1.181]
>> Caller-Username: [07751234567]
>> Caller-Caller-ID-Number: [07751234567]
>> Caller-ANI: [07751234567]
>> Caller-Channel-Name: [sofia/internal/07751234567 at 192.168.1.181]
>> variable_sip_from_user: [07751234567]
>> variable_sip_from_uri: [07751234567 at 192.168.1.181]
>> variable_sip_from_user_stripped: [07751234567]
>> variable_sip_full_from: [PSTN Line
>> <sip:07751234567 at 192.168.1.181>;tag=26bdee1e605c89ebo1]
>> variable_sip_contact_user: [07751234567]
>> variable_sip_contact_uri: [07751234567 at 192.168.1.183:5061]
>> variable_channel_name: [sofia/internal/07751234567 at 192.168.1.181]
>> variable_presence_id: [07751234567 at 192.168.1.181]
>>
>> Lots of places where I can pick up the number from, but in the interests 
>> of
>> "doing it right", I'd like to pick the "proper" (most reliable) one. 
>> Again,
>> this call is coming from PSTN via a gateway, so it looks like the best 
>> one
>> might be "Caller-Caller-ID-Number" ?
>>
>> I'm not sure where to copy the relevant variable to though ?
>>
>> regards
>> Dave
>>
>>
>> ----- Original Message -----
>> From: "Nyamul Hassan" <mnhassan at usa.net>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Saturday, October 16, 2010 11:28 AM
>> Subject: Re: [Freeswitch-users] Passing incoming Caller ID
>>
>>
>>> You have to use the variables in that line that sets caller name. I
>>> don't remember on top of my head which. You can use the dialplan
>>> application called "info" which will give you a nice list of useful
>>> variables and other information on the call and the channel.
>>>
>>> Regards
>>> HASSAN
>>>
>>>
>>> On 2010-10-16, Dave Stevenson <stevendt at primrosebank.net> wrote:
>>>> Hi,
>>>>
>>>> can someone help me with passing caller ID to FreeSwitch extensions
>>>> please?
>>>>
>>>> I have a Linksys SPA-3102 setup to receive PSTN calls and Caller ID has
>>>> just
>>>> been configured on the line.
>>>> The Gateway is setup as Extension 1000 with Dialplan entries
>>>>       <variable name="effective_caller_id_name" value="PSTN Line"/>
>>>>       <variable name="effective_caller_id_number" value="1000"/>
>>>>
>>>> I have a group name (primrose) setup (in directory\default.xml) to ring 
>>>> a
>>>> number of phones
>>>> I have extension 100 configured with
>>>>       <variable name="callgroup" value="primrose"/>
>>>>
>>>> The Gateway is setup to dial extension 100 on an incoming call
>>>>
>>>> When a call comes in, the Gateway sees the Caller ID, e.g, 07751234567
>>>>
>>>> And FreeSwitch see the caller ID from the gateway :-
>>>>
>>>> [NOTICE] switch_channel.c:669 New Channel
>>>> sofia/internal/07751234567 at 192.168.1.181
>>>> [f863c03f-0f7e-4e4b-beb0-dfa57dad2b50]
>>>>
>>>>
>>>> When the group extension ring, the caller is shown as "PSTN Line" on 
>>>> the
>>>> extensions.
>>>>
>>>>
>>>> Can someone tell me how & where I can modify the config to pick up the
>>>> CID
>>>> from the Gateway and pass it to the group extensions please ?
>>>>
>>>> regards
>>>> Dave
>>>>
>>>
>>> --
>>> Sent from my mobile device
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> -- 
> Sent from my mobile device
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From freeswitch at tlainvestments.com  Fri Oct 15 22:59:28 2010
From: freeswitch at tlainvestments.com (Troy Anderson)
Date: Fri, 15 Oct 2010 22:59:28 -0700
Subject: [Freeswitch-users] Read only filesystem
In-Reply-To: <1287185665565-5640809.post@n2.nabble.com>
References: <0925E7C5-987F-4065-85E1-2B45528D4DF1@tlainvestments.com>
	<1287185665565-5640809.post@n2.nabble.com>
Message-ID: <148F4BF8-B475-4C2F-9823-1660597CB022@tlainvestments.com>

Many times there isn't anything suspect in any logs, but recently on one system, I saw the following (repeated many times over time):

Oct 13 19:40:13 localhost kernel: ata2.00: exception Emask 0x0 SAct 0x0 SErr 0x0 action 0x6 frozen
Oct 13 19:40:13 localhost kernel: ata2.00: cmd e5/00:00:00:00:00/00:00:00:00:00/00 tag 0
Oct 13 19:40:13 localhost kernel:          res 40/00:01:00:00:00/00:00:00:00:00/00 Emask 0x4 (timeout)
Oct 13 19:40:13 localhost kernel: ata2.00: status: { DRDY }
Oct 13 19:40:17 localhost kernel: ata2: soft resetting link
Oct 13 19:40:17 localhost kernel: ata2.00: configured for UDMA/33
Oct 13 19:40:17 localhost kernel: sd 1:0:0:0: timing out command, waited 7s
Oct 13 19:40:17 localhost kernel: ata2: EH complete
Oct 13 19:40:17 localhost kernel: SCSI device sda: 234441648 512-byte hdwr sectors (120034 MB)
Oct 13 19:40:17 localhost kernel: sda: Write Protect is off
Oct 13 19:40:18 localhost kernel: SCSI device sda: drive cache: write back

Can anyone help me with pointers on how to determine what they mean?


On Oct 15, 2010, at 4:34 PM, mazilo wrote:

> 
> 
> Troy Anderson wrote:
>> At this point, I am trying to figure out what may be the root cause.  Has
>> anyone else experienced something this?  Any suggestions?
> Your best bet on this is to look at the log file (/var/log/messages and
> among others in the same sub-dir) and see if you can identify what triggered
> the HD into a RO mode.
> 
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> -- 
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Read-only-filesystem-tp5640425p5640809.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jason at jasonjgw.net  Fri Oct 15 18:17:01 2010
From: jason at jasonjgw.net (Jason White)
Date: Sat, 16 Oct 2010 12:17:01 +1100
Subject: [Freeswitch-users] Binding to external IP address during system boot
Message-ID: <87wrpj0wdu.fsf@jdc.jasonjgw.net>


I am running FreeSWITCH under Debian, with ppp0 (an ADSL connection) as
the external interface. Often, FreeSWITCH successfully starts and binds
to the (dynamically assigned) external IPv4 address during the system
boot process. Sometimes, however, this fails and I get (from sofia
status profile external):
sip:mod_sofia at 127.0.0.1:5080

Clearly, it has bound to the loopback interface. Restarting the sip
profile doesn't fix it, but restarting FreeSWITCH does.

Is there a command that I can place in a networking script that will
force the binding to be dropped and a new binding to be created, without
restarting FreeSWITCH entirely?



From steveayre at gmail.com  Sat Oct 16 07:28:49 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 16 Oct 2010 15:28:49 +0100
Subject: [Freeswitch-users] Binding to external IP address during system
	boot
In-Reply-To: <87wrpj0wdu.fsf@jdc.jasonjgw.net>
References: <87wrpj0wdu.fsf@jdc.jasonjgw.net>
Message-ID: <686AA79D-D8FE-4A5B-8209-DE8EBF272AF2@gmail.com>

One if these would do it:
sofia reload
sofia profile <name> restart

Note you might (probably will) get dropped calls if you do it while calls are in progress.

Steve on iPhone

On 16 Oct 2010, at 02:17, Jason White <jason at jasonjgw.net> wrote:

> 
> I am running FreeSWITCH under Debian, with ppp0 (an ADSL connection) as
> the external interface. Often, FreeSWITCH successfully starts and binds
> to the (dynamically assigned) external IPv4 address during the system
> boot process. Sometimes, however, this fails and I get (from sofia
> status profile external):
> sip:mod_sofia at 127.0.0.1:5080
> 
> Clearly, it has bound to the loopback interface. Restarting the sip
> profile doesn't fix it, but restarting FreeSWITCH does.
> 
> Is there a command that I can place in a networking script that will
> force the binding to be dropped and a new binding to be created, without
> restarting FreeSWITCH entirely?
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From steveayre at gmail.com  Sat Oct 16 07:37:25 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 16 Oct 2010 15:37:25 +0100
Subject: [Freeswitch-users] Read only filesystem
In-Reply-To: <148F4BF8-B475-4C2F-9823-1660597CB022@tlainvestments.com>
References: <0925E7C5-987F-4065-85E1-2B45528D4DF1@tlainvestments.com>
	<1287185665565-5640809.post@n2.nabble.com>
	<148F4BF8-B475-4C2F-9823-1660597CB022@tlainvestments.com>
Message-ID: <63ECC286-C13A-4796-8E5F-43635B7CAA7E@gmail.com>

See below.

Steve on iPhone

On 16 Oct 2010, at 06:59, Troy Anderson <freeswitch at tlainvestments.com> wrote:

> Many times there isn't anything suspect in any logs, but recently on one system, I saw the following (repeated many times over time):
> 
> Oct 13 19:40:13 localhost kernel: ata2.00: exception Emask 0x0 SAct 0x0 SErr 0x0 action 0x6 frozen
> Oct 13 19:40:13 localhost kernel: ata2.00: cmd e5/00:00:00:00:00/00:00:00:00:00/00 tag 0
> Oct 13 19:40:13 localhost kernel:          res 40/00:01:00:00:00/00:00:00:00:00/00 Emask 0x4 (timeout)

The driver asked the drive to do something and got no response from the drive. The numbers tell you what, but is only really useful if you're a developer of the driver.

> Oct 13 19:40:13 localhost kernel: ata2.00: status: { DRDY }
> Oct 13 19:40:17 localhost kernel: ata2: soft resetting link

Chipset is disconnecting and reconnecting the drive, since that can sometimes fix a problem by resetting the drive.

> Oct 13 19:40:17 localhost kernel: ata2.00: configured for UDMA/33

Fallen speed back to UDMA 33 (slowest UDMA), since some problems are caused by the drive running faster than the drive/cable can cope with.

> Oct 13 19:40:17 localhost kernel: sd 1:0:0:0: timing out command, waited 7s

Something timed out - not sure if this is the reset or the failed action that triggered the reset.

> Oct 13 19:40:17 localhost kernel: ata2: EH complete
> Oct 13 19:40:17 localhost kernel: SCSI device sda: 234441648 512-byte hdwr sectors (120034 MB)
> Oct 13 19:40:17 localhost kernel: sda: Write Protect is off
> Oct 13 19:40:18 localhost kernel: SCSI device sda: drive cache: write back

Drive is back.

> 
> Can anyone help me with pointers on how to determine what they mean?
> 
> 
> On Oct 15, 2010, at 4:34 PM, mazilo wrote:
> 
>> 
>> 
>> Troy Anderson wrote:
>>> At this point, I am trying to figure out what may be the root cause.  Has
>>> anyone else experienced something this?  Any suggestions?
>> Your best bet on this is to look at the log file (/var/log/messages and
>> among others in the same sub-dir) and see if you can identify what triggered
>> the HD into a RO mode.
>> 
>> -----
>> don't and stop are the ONLY two 4-letter words considered offensive to men,
>> but not when used together.
>> -- 
>> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Read-only-filesystem-tp5640425p5640809.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From curriegrad2004 at gmail.com  Sat Oct 16 09:05:58 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Sat, 16 Oct 2010 09:05:58 -0700
Subject: [Freeswitch-users] Read only filesystem
In-Reply-To: <63ECC286-C13A-4796-8E5F-43635B7CAA7E@gmail.com>
References: <0925E7C5-987F-4065-85E1-2B45528D4DF1@tlainvestments.com>
	<1287185665565-5640809.post@n2.nabble.com>
	<148F4BF8-B475-4C2F-9823-1660597CB022@tlainvestments.com>
	<63ECC286-C13A-4796-8E5F-43635B7CAA7E@gmail.com>
Message-ID: <AANLkTimxYEvAbsb4YTsJi9BewB-YFoW5XRqw+rJB7so-@mail.gmail.com>

Sounds like a firmware issue with either the Drive Controller or the
hard drive itself.

On Sat, Oct 16, 2010 at 7:37 AM, Steven Ayre <steveayre at gmail.com> wrote:
> See below.
>
> Steve on iPhone
>
> On 16 Oct 2010, at 06:59, Troy Anderson <freeswitch at tlainvestments.com> wrote:
>
>> Many times there isn't anything suspect in any logs, but recently on one system, I saw the following (repeated many times over time):
>>
>> Oct 13 19:40:13 localhost kernel: ata2.00: exception Emask 0x0 SAct 0x0 SErr 0x0 action 0x6 frozen
>> Oct 13 19:40:13 localhost kernel: ata2.00: cmd e5/00:00:00:00:00/00:00:00:00:00/00 tag 0
>> Oct 13 19:40:13 localhost kernel: ? ? ? ? ?res 40/00:01:00:00:00/00:00:00:00:00/00 Emask 0x4 (timeout)
>
> The driver asked the drive to do something and got no response from the drive. The numbers tell you what, but is only really useful if you're a developer of the driver.
>
>> Oct 13 19:40:13 localhost kernel: ata2.00: status: { DRDY }
>> Oct 13 19:40:17 localhost kernel: ata2: soft resetting link
>
> Chipset is disconnecting and reconnecting the drive, since that can sometimes fix a problem by resetting the drive.
>
>> Oct 13 19:40:17 localhost kernel: ata2.00: configured for UDMA/33
>
> Fallen speed back to UDMA 33 (slowest UDMA), since some problems are caused by the drive running faster than the drive/cable can cope with.
>
>> Oct 13 19:40:17 localhost kernel: sd 1:0:0:0: timing out command, waited 7s
>
> Something timed out - not sure if this is the reset or the failed action that triggered the reset.
>
>> Oct 13 19:40:17 localhost kernel: ata2: EH complete
>> Oct 13 19:40:17 localhost kernel: SCSI device sda: 234441648 512-byte hdwr sectors (120034 MB)
>> Oct 13 19:40:17 localhost kernel: sda: Write Protect is off
>> Oct 13 19:40:18 localhost kernel: SCSI device sda: drive cache: write back
>
> Drive is back.
>
>>
>> Can anyone help me with pointers on how to determine what they mean?
>>
>>
>> On Oct 15, 2010, at 4:34 PM, mazilo wrote:
>>
>>>
>>>
>>> Troy Anderson wrote:
>>>> At this point, I am trying to figure out what may be the root cause. ?Has
>>>> anyone else experienced something this? ?Any suggestions?
>>> Your best bet on this is to look at the log file (/var/log/messages and
>>> among others in the same sub-dir) and see if you can identify what triggered
>>> the HD into a RO mode.
>>>
>>> -----
>>> don't and stop are the ONLY two 4-letter words considered offensive to men,
>>> but not when used together.
>>> --
>>> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Read-only-filesystem-tp5640425p5640809.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From timb0311 at hotmail.com  Sat Oct 16 08:40:58 2010
From: timb0311 at hotmail.com (Tim B)
Date: Sat, 16 Oct 2010 11:40:58 -0400
Subject: [Freeswitch-users] Compile on Windows
Message-ID: <BAY157-w52ACAB7DF808D795ADBB3BDE580@phx.gbl>


Anybody compiling freeswitch on windows?  I am running windows 7 x64 with VS2010.  Downloaded the latest git snapshot of freeswitch 1.0.6 and opened the vs2010 project, then tried to compile... getting like 700+ errors.
 
I was able to compile version 1.0.4 on Vista with VS2008 with no problems awhile back.
 
Just wondering if anyone else had any problems similiar and how they resolved them.
 
Tim
  		 	   		  
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From rupa at rupa.com  Sat Oct 16 09:51:01 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 16 Oct 2010 11:51:01 -0500
Subject: [Freeswitch-users] Read only filesystem
In-Reply-To: <148F4BF8-B475-4C2F-9823-1660597CB022@tlainvestments.com>
References: <0925E7C5-987F-4065-85E1-2B45528D4DF1@tlainvestments.com>
	<1287185665565-5640809.post@n2.nabble.com>
	<148F4BF8-B475-4C2F-9823-1660597CB022@tlainvestments.com>
Message-ID: <AANLkTim1Wg1-C4hE4HE-fAOejT97DDi08id4eUo_gsx+@mail.gmail.com>

Have you ruled out hard drive failure using smartctl?  Do a long selftest
and review the results.

Also, swap out your SATA cables or at least ensure they are fully seated.

Finally, check to see if there are known issues with the SATA interface you
are using (is it onboard, add-in card, raid card?) and the linux kernel you
have installed.

On Sat, Oct 16, 2010 at 12:59 AM, Troy Anderson <
freeswitch at tlainvestments.com> wrote:

> Many times there isn't anything suspect in any logs, but recently on one
> system, I saw the following (repeated many times over time):
>
> Oct 13 19:40:13 localhost kernel: ata2.00: exception Emask 0x0 SAct 0x0
> SErr 0x0 action 0x6 frozen
> Oct 13 19:40:13 localhost kernel: ata2.00: cmd
> e5/00:00:00:00:00/00:00:00:00:00/00 tag 0
> Oct 13 19:40:13 localhost kernel:          res
> 40/00:01:00:00:00/00:00:00:00:00/00 Emask 0x4 (timeout)
> Oct 13 19:40:13 localhost kernel: ata2.00: status: { DRDY }
> Oct 13 19:40:17 localhost kernel: ata2: soft resetting link
> Oct 13 19:40:17 localhost kernel: ata2.00: configured for UDMA/33
> Oct 13 19:40:17 localhost kernel: sd 1:0:0:0: timing out command, waited 7s
> Oct 13 19:40:17 localhost kernel: ata2: EH complete
> Oct 13 19:40:17 localhost kernel: SCSI device sda: 234441648 512-byte hdwr
> sectors (120034 MB)
> Oct 13 19:40:17 localhost kernel: sda: Write Protect is off
> Oct 13 19:40:18 localhost kernel: SCSI device sda: drive cache: write back
>
> Can anyone help me with pointers on how to determine what they mean?
>
>
> On Oct 15, 2010, at 4:34 PM, mazilo wrote:
>
> >
> >
> > Troy Anderson wrote:
> >> At this point, I am trying to figure out what may be the root cause.
>  Has
> >> anyone else experienced something this?  Any suggestions?
> > Your best bet on this is to look at the log file (/var/log/messages and
> > among others in the same sub-dir) and see if you can identify what
> triggered
> > the HD into a RO mode.
> >
> > -----
> > don't and stop are the ONLY two 4-letter words considered offensive to
> men,
> > but not when used together.
> > --
> > View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Read-only-filesystem-tp5640425p5640809.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From Russell.Mosemann at cune.org  Sat Oct 16 09:54:02 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sat, 16 Oct 2010 11:54:02 -0500
Subject: [Freeswitch-users] Read only filesystem
In-Reply-To: <AANLkTimxYEvAbsb4YTsJi9BewB-YFoW5XRqw+rJB7so-@mail.gmail.com>
References: <0925E7C5-987F-4065-85E1-2B45528D4DF1@tlainvestments.com><1287185665565-5640809.post@n2.nabble.com><148F4BF8-B475-4C2F-9823-1660597CB022@tlainvestments.com><63ECC286-C13A-4796-8E5F-43635B7CAA7E@gmail.com>
	<AANLkTimxYEvAbsb4YTsJi9BewB-YFoW5XRqw+rJB7so-@mail.gmail.com>
Message-ID: <6918DCF6CDFF44C8961FBC23C3D2A8F7@cune.pri>

Jeffrey Leung suggested:

> Sounds like a firmware issue with either the Drive Controller or the
> hard drive itself.

This can also happen with consumer-grade drives when they detect and remap a bad block (or chunk of bad blocks). This is especially problematic with RAID. The drive is busy fixing the problem, and it appears to the driver that the drive is no longer responding. If this happens frequently, it might be an indication that the drive is failing. The drive could also be overheating.

mose



From Russell.Mosemann at cune.org  Sat Oct 16 10:01:16 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sat, 16 Oct 2010 12:01:16 -0500
Subject: [Freeswitch-users] Read only filesystem
In-Reply-To: <AANLkTim1Wg1-C4hE4HE-fAOejT97DDi08id4eUo_gsx+@mail.gmail.com>
References: <0925E7C5-987F-4065-85E1-2B45528D4DF1@tlainvestments.com><1287185665565-5640809.post@n2.nabble.com><148F4BF8-B475-4C2F-9823-1660597CB022@tlainvestments.com>
	<AANLkTim1Wg1-C4hE4HE-fAOejT97DDi08id4eUo_gsx+@mail.gmail.com>
Message-ID: <42A8056AECDB40DA9C728F9D1C91A05A@cune.pri>

Rupa Schomaker wrote:

> Also, swap out your SATA cables or at least ensure they are fully seated.

That's a good suggestion. I had purchased some cheap SATA cables. They were fine at the beginning and over the next year or two they started failing. The drive would just drop out of a RAID array without warning, because it appeared to fail. Testing the drive didn't show any errors. Reseating the cables didn't help. Replacing the cable was the next thing to try, and it fixed the problem. I didn't track down why the cables failed, because it wasn't worth the time. The connector might have oxidized or something.

mose



From pjintheusa at gmail.com  Sat Oct 16 10:57:27 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sat, 16 Oct 2010 13:57:27 -0400
Subject: [Freeswitch-users] Compile on Windows
In-Reply-To: <BAY157-w52ACAB7DF808D795ADBB3BDE580@phx.gbl>
References: <BAY157-w52ACAB7DF808D795ADBB3BDE580@phx.gbl>
Message-ID: <AANLkTik88xyTdZdZeeN0jiEjRQ0Sn4H4Pa5aMWTQ4uM+@mail.gmail.com>

Its a GIT thing

Note from: http://wiki.freeswitch.org/wiki/Download_FreeSWITCH

"You must set autocrlf=false otherwise the gawk scripts will fail!"

On Sat, Oct 16, 2010 at 11:40 AM, Tim B <timb0311 at hotmail.com> wrote:

>  Anybody compiling freeswitch on windows?  I am running windows 7 x64 with
> VS2010.  Downloaded the latest git snapshot of freeswitch 1.0.6 and opened
> the vs2010 project, then tried to compile... getting like 700+ errors.
>
> I was able to compile version 1.0.4 on Vista with VS2008 with no problems
> awhile back.
>
> Just wondering if anyone else had any problems similiar and how they
> resolved them.
>
> Tim
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peter.olsson at visionutveckling.se  Sat Oct 16 11:00:13 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 16 Oct 2010 20:00:13 +0200
Subject: [Freeswitch-users] Compile on Windows
In-Reply-To: <BAY157-w52ACAB7DF808D795ADBB3BDE580@phx.gbl>
References: <BAY157-w52ACAB7DF808D795ADBB3BDE580@phx.gbl>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57E5ECDC2F@cooper>

Did you download the .tar.gz file for 1.0.6, or did you use git HEAD? I've not used VS2010 myself, but I know that Jeff has done quite some work on it last couple of months, so I think it should work ok. But it will require the latest git HEAD.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tim B [timb0311 at hotmail.com]
Skickat: den 16 oktober 2010 17:40
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Compile on Windows

Anybody compiling freeswitch on windows?  I am running windows 7 x64 with VS2010.  Downloaded the latest git snapshot of freeswitch 1.0.6 and opened the vs2010 project, then tried to compile... getting like 700+ errors.

I was able to compile version 1.0.4 on Vista with VS2008 with no problems awhile back.

Just wondering if anyone else had any problems similiar and how they resolved them.

Tim

!DSPAM:4cb9d8e632937089411596!


From curriegrad2004 at gmail.com  Sat Oct 16 18:11:08 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Sat, 16 Oct 2010 18:11:08 -0700
Subject: [Freeswitch-users] Compile on Windows
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57E5ECDC2F@cooper>
References: <BAY157-w52ACAB7DF808D795ADBB3BDE580@phx.gbl>
	<549CFEF87AEDE841A38E9D15EAB4C04C57E5ECDC2F@cooper>
Message-ID: <AANLkTi=_VoOu1EVsm6CZp+okOdfcVnvKdR=+iY9fOc8g@mail.gmail.com>

I'm able to compile the latest git build on VS2008/2010 without too
much issues either. It does sound like the auto crlf conversion issue
that's causing problems here.

On Sat, Oct 16, 2010 at 11:00 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> Did you download the .tar.gz file for 1.0.6, or did you use git HEAD? I've not used VS2010 myself, but I know that Jeff has done quite some work on it last couple of months, so I think it should work ok. But it will require the latest git HEAD.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tim B [timb0311 at hotmail.com]
> Skickat: den 16 oktober 2010 17:40
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: [Freeswitch-users] Compile on Windows
>
> Anybody compiling freeswitch on windows? ?I am running windows 7 x64 with VS2010. ?Downloaded the latest git snapshot of freeswitch 1.0.6 and opened the vs2010 project, then tried to compile... getting like 700+ errors.
>
> I was able to compile version 1.0.4 on Vista with VS2008 with no problems awhile back.
>
> Just wondering if anyone else had any problems similiar and how they resolved them.
>
> Tim
>
> !DSPAM:4cb9d8e632937089411596!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From curriegrad2004 at gmail.com  Sat Oct 16 18:13:23 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Sat, 16 Oct 2010 18:13:23 -0700
Subject: [Freeswitch-users] Read only filesystem
In-Reply-To: <42A8056AECDB40DA9C728F9D1C91A05A@cune.pri>
References: <0925E7C5-987F-4065-85E1-2B45528D4DF1@tlainvestments.com>
	<1287185665565-5640809.post@n2.nabble.com>
	<148F4BF8-B475-4C2F-9823-1660597CB022@tlainvestments.com>
	<AANLkTim1Wg1-C4hE4HE-fAOejT97DDi08id4eUo_gsx+@mail.gmail.com>
	<42A8056AECDB40DA9C728F9D1C91A05A@cune.pri>
Message-ID: <AANLkTinP3GBVwfne=8X0BuP+3fGzUHi8Y1N4=1vS0Vij@mail.gmail.com>

It could also be bad soldering on either ends of the connectors, or
lead-free solder growing 'legs' and causing a short somewhere. I
wished they stopped marketing RoHS products... There's a reason why to
this date, no sane avionics technician will ever use RoHS solder on an
airplane.

On Sat, Oct 16, 2010 at 10:01 AM, Russell Mosemann
<Russell.Mosemann at cune.org> wrote:
> Rupa Schomaker wrote:
>
>> Also, swap out your SATA cables or at least ensure they are fully seated.
>
> That's a good suggestion. I had purchased some cheap SATA cables. They were fine at the beginning and over the next year or two they started failing. The drive would just drop out of a RAID array without warning, because it appeared to fail. Testing the drive didn't show any errors. Reseating the cables didn't help. Replacing the cable was the next thing to try, and it fixed the problem. I didn't track down why the cables failed, because it wasn't worth the time. The connector might have oxidized or something.
>
> mose
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From edpimentl at gmail.com  Sat Oct 16 20:22:10 2010
From: edpimentl at gmail.com (EdPimentl)
Date: Sat, 16 Oct 2010 23:22:10 -0400
Subject: [Freeswitch-users] PHONO ...turns any web browser into a PHONE and
	IM Client
Message-ID: <AANLkTikVYGY5wHoTDpFRjf6UGVEyJfq1wbZH5u7MtNQs@mail.gmail.com>

http://phono.com/

Phono is a simple JQuery plugin and JavaScript SDK
that turns any web browser into a multi-channel communications platform
capable of making phone calls and sending IM messages. You can even connect
to SIP clients; all with a simple unified API.

-E
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From dujinfang at gmail.com  Sat Oct 16 23:27:18 2010
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 17 Oct 2010 14:27:18 +0800
Subject: [Freeswitch-users] mod_erlang_event core dump on load with
	shortname=false on Mac
Message-ID: <AANLkTikToP3n7rGHO3DgaW3C1cK-XpN7WcmzNdzd8KHr@mail.gmail.com>

Hi,

I experience this a few days ago, everything is ok with
shortname=true, but shortname=false causes immediately core dump on
load.

I also pulled the latest git. problem remains. It has nothing to do
with the latest change.

The last change does help. I set <param name="nodename"
value="freeswitch at 192.168.1.128"/> then no more crash.

So, the problem might be because it has trouble on get the fully
qualified domain name on my box.

Any other experience on this?  I can make a jira if confirmed a bug.

Thanks.
-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From andrew at hijacked.us  Sun Oct 17 09:42:57 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Sun, 17 Oct 2010 12:42:57 -0400
Subject: [Freeswitch-users] mod_erlang_event core dump on load with
	shortname=false on Mac
In-Reply-To: <AANLkTikToP3n7rGHO3DgaW3C1cK-XpN7WcmzNdzd8KHr@mail.gmail.com>
References: <AANLkTikToP3n7rGHO3DgaW3C1cK-XpN7WcmzNdzd8KHr@mail.gmail.com>
Message-ID: <20101017164257.GK15964@hijacked.us>

Or just email me privately the information. I've been trying to rework
that code to match how erlang infers long names (because the ones I
inferred weren't matching erlang's).

The email or the JIRA should include the backtrace, the output of
'hostname', 'domainname', and /etc/resolv.conf. Also, what erlang on the
box guesses for short/long names.

Andrew


From daniel.neubert at solomo.de  Sun Oct 17 11:32:06 2010
From: daniel.neubert at solomo.de (Daniel Neubert)
Date: Sun, 17 Oct 2010 20:32:06 +0200
Subject: [Freeswitch-users] mod_cepstral 5.1 known issues?
In-Reply-To: <AANLkTi=cAW7Ru3KPy4LH7bwy0eGKce0q5z-yztF-Nhd2@mail.gmail.com>
References: <AANLkTik8KZdxAnUUfv0nKF6EkTHe559PtC2bF74eng11@mail.gmail.com>	<AANLkTi=MvWh14MQ9wPEboXZmMgAtURAY5zBoobn32Pkq@mail.gmail.com>	<AANLkTimj=SwX7d1sot+NhOxMgD4F-PN=bM3hxrh3bF3O@mail.gmail.com>	<4CB7F0C9.2040708@solomo.de>	<0FF607C13B7F7A42B5B80DA9EB204C9109D44BDD8E@srv-ex01-dal.intervoice.int>
	<AANLkTi=cAW7Ru3KPy4LH7bwy0eGKce0q5z-yztF-Nhd2@mail.gmail.com>
Message-ID: <4CBB4126.9020208@solomo.de>

  I'm sure it will ;-) This one is from me and the answers given / 
linked in this jira ticket helped me a lot.

Best regards / Mit freundlichen Gr??en,
Daniel Neubert

Am 16.10.2010 01:22, schrieb Marc Kellem:
> Holger, this JIRA issue might answer some of your questions.
>
> http://jira.freeswitch.org/browse/FS-10
>
>
>
> On Fri, Oct 15, 2010 at 7:32 AM, Esser, Holger 
> <Holger.Esser at convergys.com <mailto:Holger.Esser at convergys.com>> wrote:
>
>     Hi Daniel,
>
>     May I ask which guide you used to set it up?
>
>     Thanks,
>
>     Holger
>
>     *From:* freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>     [mailto:freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf
>     Of *Daniel Neubert
>     *Sent:* Friday, October 15, 2010 1:12 AM
>     *To:* FreeSWITCH Users Help
>     *Subject:* Re: [Freeswitch-users] mod_cepstral 5.1 known issues?
>
>     I've exactly this setup which is in productive use for a few
>     months now.
>
>     Using cepstral directly on our FreeSWITCH nodes did not work
>     (since they are running on FreeBSD and Cepstral is only available
>     for GNU Linux).
>
>     Current setup is using Voice Katrin (German) Version 5.1.0 on
>     Gentoo Linux 64Bit via UniMRCP Server 1.0.0.
>
>     Best regards / Mit freundlichen Gr??en,
>     Daniel Neubert
>
>
>     On 14.10.2010 17:58, Marc Kellem wrote:
>
>     Is anyone using mod_unimrcp to access Cepstral 5.1 voices in a
>     UniMRCP server?
>
>     On Thu, Oct 14, 2010 at 11:26 AM, Anthony Minessale
>     <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>>
>     wrote:
>
>     random deadlocks in the cepstral engine blocking all the speech
>     generation calls.
>
>
>
>     On Thu, Oct 14, 2010 at 9:56 AM, Marc Kellem <mkellem at vontoo.com
>     <mailto:mkellem at vontoo.com>> wrote:
>     > The mod_cepstral wiki page says "Warning: the suggested version
>     to use
>     > is 4.x since there are known issues with 5.1 (which is closed
>     source)".
>     >  What exactly are the known issues?
>     > http://wiki.freeswitch.org/wiki/Mod_cepstral
>     > Thanks,
>     > Marc Kellem
>
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > 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
>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
>     FreeSWITCH Developer Conference
>     sip:888 at conference.freeswitch.org
>     <mailto:sip%3A888 at conference.freeswitch.org>
>     googletalk:conf+888 at conference.freeswitch.org
>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>     pstn:+19193869900
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>     No virus found in this incoming message.
>     Checked by AVG - www.avg.com <http://www.avg.com>
>     Version: 9.0.862 / Virus Database: 271.1.1/3192 - Release Date:
>     10/14/10 13:34:00
>
>
>     ------------------------------------------------------------------------
>     This e-mail transmission may contain information that is
>     proprietary, privileged and/or confidential and is intended
>     exclusively for the person(s) to whom it is addressed. Any use,
>     copying, retention or disclosure by any person other than the
>     intended recipient or the intended recipient's designees is
>     strictly prohibited. If you are the intended recipient, you must
>     treat the information in confidence and in accordance with all
>     laws related to the privacy and confidentiality of such
>     information. If you are not the intended recipient or their
>     designee, please notify the sender immediately by return e-mail
>     and delete all copies of this email, including all attachments.
>
>     Intervoice, Inc. 17811 Waterview Parkway Dallas, TX 75252 USA
>
>     Intervoice Limited, Registered in England and Wales with number
>     2601740, 50 Park Road, Gatley, Cheshire, SK8 4HZ. VAT Number:
>     560421375
>
>     Intervoice GmbH, Hagenauer Stra?e 55, 65203 Wiesbaden, Sitz der
>     Gesellschaft: Wiesbaden, Handelsregister: HRB 8186 (Amtsgericht
>     Wiesbaden), Gesch?ftsf?hrer: Wayne Barclay, Steffen Selbmann
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
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From jeff at jefflenk.com  Sun Oct 17 14:25:28 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sun, 17 Oct 2010 14:25:28 -0700 (PDT)
Subject: [Freeswitch-users] Compile on Windows
In-Reply-To: <AANLkTi=_VoOu1EVsm6CZp+okOdfcVnvKdR=+iY9fOc8g@mail.gmail.com>
References: <BAY157-w52ACAB7DF808D795ADBB3BDE580@phx.gbl>
	<549CFEF87AEDE841A38E9D15EAB4C04C57E5ECDC2F@cooper>
	<AANLkTi=_VoOu1EVsm6CZp+okOdfcVnvKdR=+iY9fOc8g@mail.gmail.com>
Message-ID: <1287350728991-5645270.post@n2.nabble.com>


Follow the  instructions here for Git on windows -
http://wiki.freeswitch.org/wiki/Git_Install

Visual Studio 2010 support for both Express and Pro+ builds is available in
Git Head for both x86 and x64.

Please feel free to contribute to the documentation or any other way you can
to help new windows users on the project.

Thanks
Jeff
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Compile-on-Windows-tp5642331p5645270.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From slim at thegreek.com  Sun Oct 17 15:33:09 2010
From: slim at thegreek.com (Jeroen C. van Gelderen)
Date: Sun, 17 Oct 2010 17:33:09 -0500
Subject: [Freeswitch-users] exec_after*
Message-ID: <0298143B64B347A996A1F1766DF3D7F8@mbnet.local>

Hi,

My helpdesk extensions are recorded. When the recording is complete (after
bridge) I want to run a script to create a ticket and attach said recording
to it.

Initially I tried to use

[...]
Dialplan: FreeTDM/3:1/ Action set(exec_after_bridge_app=system)
Dialplan: FreeTDM/3:1/ Action
set(exec_after_bridge_arg=/root/wav2ticket.php)
[...]

but this doesn't result in an invocation of my script and seemingly no
helpful diagnostic messages.

I changed to using

<action application="set" 
 data="api_after_bridge=system /root/wav2ticket.php ${recfile}"/>

which DOES work.

Can anyone clarify the difference between the two approaches?

Cheers,
-Slim
--
Jeroen C. "Slim" van Gelderen
Olympic Sports Data Services
Email: slim at thegreek.com
Phone: +1 876 953 6182 x128




From tear152 at hotmail.com  Sun Oct 17 17:37:00 2010
From: tear152 at hotmail.com (=?gb2312?B?va0gQ2hpYW5nINbB1tAgQ2hpaC1DaHVuZyAgIFd5Ymll?=)
Date: Mon, 18 Oct 2010 08:37:00 +0800
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <AD407430-9024-4E0D-A860-9026FD2837A3@ipeva.fr>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>, ,
	,,<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>, ,
	,,<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>, ,,,
	<SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>, ,
	,,<AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>,
	,,<SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>,
	, , <C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>, ,
	<SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>, ,
	<6A8D7D0E-4DF5-4F35-8A00-F52FECC4F826@ipeva.fr>,
	<SNT130-w6005F900786BDFBE48FF43FB570@phx.gbl>,
	<AD407430-9024-4E0D-A860-9026FD2837A3@ipeva.fr>
Message-ID: <SNT130-w445656F3A9C14ED384A1B0FB5A0@phx.gbl>


I tried to set different expression but it still can't work
 
debug log below
 
2010-10-18 08:34:19.462617 [DEBUG] sofia.c:5849 IP 60.248.175.43 Approved by acl "domains[]". Access Granted.
2010-10-18 08:34:19.462617 [NOTICE] switch_channel.c:675 New Channel sofia/internal/1013 at 60.248.175.38 [1f4277c1-9ded-49c5-b63f-a312bead9d5f]
2010-10-18 08:34:19.464814 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_NEW
2010-10-18 08:34:19.464814 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1013 at 60.248.175.38) State NEW
2010-10-18 08:34:19.475647 [DEBUG] sofia.c:4200 Channel sofia/internal/1013 at 60.248.175.38 entering state [received][100]
2010-10-18 08:34:19.475647 [DEBUG] sofia.c:4211 Remote SDP:
v=0
o=- 0 2 IN IP4 60.248.175.43
s=CounterPath X-Lite 3.0
c=IN IP4 60.248.175.43
t=0 0
m=audio 32692 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 3 : d7A5MyJF tN44o7uR 60.248.175.43 32692
a=alt:2 2 : cCI83KQt qU62M7Ia 192.168.182.1 32692
a=alt:3 1 : qtl4Jt3z ug+kYltk 192.168.60.1 32692
2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[G7221:115:32000:20]
2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[G7221:107:16000:20]
2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[G722:9:8000:20]
2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[PCMU:0:8000:20]
2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[PCMA:8:8000:20]
2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[GSM:3:8000:20]
2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [PCMU:0:8000:20]/[G7221:115:32000:20]
2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [PCMU:0:8000:20]/[G7221:107:16000:20]
2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [PCMU:0:8000:20]/[G722:9:8000:20]
2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20]
2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:2376 Set Codec sofia/internal/1013 at 60.248.175.38 PCMU/8000 20 ms 160 samples
2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3610 Set 2833 dtmf send/recv payload to 101
2010-10-18 08:34:19.476688 [DEBUG] sofia.c:4357 (sofia/internal/1013 at 60.248.175.38) State Change CS_NEW -> CS_INIT
2010-10-18 08:34:19.476688 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_INIT
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1013 at 60.248.175.38) State INIT
2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:83 sofia/internal/1013 at 60.248.175.38 SOFIA INIT
2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:117 (sofia/internal/1013 at 60.248.175.38) State Change CS_INIT -> CS_ROUTING
2010-10-18 08:34:19.476688 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1013 at 60.248.175.38) State INIT going to sleep
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_ROUTING
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38) State ROUTING
2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:140 sofia/internal/1013 at 60.248.175.38 SOFIA ROUTING
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1013 at 60.248.175.38 Standard ROUTING
2010-10-18 08:34:19.476688 [INFO] mod_dialplan_xml.c:331 Processing 1013->4001 in context public
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->unloop] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->outside_call] continue=true
Dialplan: sofia/internal/1013 at 60.248.175.38 Absolute Condition [outside_call]
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(outside_call=true)
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->call_debug] continue=true
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->public_extensions] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [public_extensions] destination_number(4001) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->Calls from 37] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [Calls from 37] destination_number(4001) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->public_did] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [public_did] destination_number(4001) =~ /^(5551212)$/ break=on-false
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1013 at 60.248.175.38) State Change CS_ROUTING -> CS_EXECUTE
2010-10-18 08:34:19.476688 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38) State ROUTING going to sleep
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_EXECUTE
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38) State EXECUTE
2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:233 sofia/internal/1013 at 60.248.175.38 SOFIA EXECUTE
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1013 at 60.248.175.38 Standard EXECUTE
EXECUTE sofia/internal/1013 at 60.248.175.38 set(outside_call=true)
2010-10-18 08:34:19.476688 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [outside_call]=[true]
2010-10-18 08:34:19.476688 [NOTICE] switch_core_state_machine.c:185 sofia/internal/1013 at 60.248.175.38 has executed the last dialplan instruction, hanging up.
2010-10-18 08:34:19.476688 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/1013 at 60.248.175.38 [CS_EXECUTE] [NORMAL_CLEARING]
2010-10-18 08:34:19.476688 [DEBUG] switch_channel.c:2145 Send signal sofia/internal/1013 at 60.248.175.38 [KILL]
2010-10-18 08:34:19.476688 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38) State EXECUTE going to sleep
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_HANGUP
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/1013 at 60.248.175.38) State HANGUP
2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:441 Channel sofia/internal/1013 at 60.248.175.38 hanging up, cause: NORMAL_CLEARING
2010-10-18 08:34:19.494277 [DEBUG] mod_sofia.c:503 Responding to INVITE with: 480
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1013 at 60.248.175.38 Standard HANGUP, cause: NORMAL_CLEARING
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/1013 at 60.248.175.38) State HANGUP going to sleep
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/1013 at 60.248.175.38) State Change CS_HANGUP -> CS_REPORTING
2010-10-18 08:34:19.494277 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_REPORTING
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:589 (sofia/internal/1013 at 60.248.175.38) State REPORTING
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1013 at 60.248.175.38 Standard REPORTING, cause: NORMAL_CLEARING
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:589 (sofia/internal/1013 at 60.248.175.38) State REPORTING going to sleep
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/1013 at 60.248.175.38) State Change CS_REPORTING -> CS_DESTROY
2010-10-18 08:34:19.494277 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 08:34:19.494277 [DEBUG] switch_core_session.c:1170 Session 61 (sofia/internal/1013 at 60.248.175.38) Locked, Waiting on external entities
2010-10-18 08:34:19.494277 [NOTICE] switch_core_session.c:1188 Session 61 (sofia/internal/1013 at 60.248.175.38) Ended
2010-10-18 08:34:19.494277 [NOTICE] switch_core_session.c:1190 Close Channel sofia/internal/1013 at 60.248.175.38 [CS_DESTROY]
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:428 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_DESTROY
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:438 (sofia/internal/1013 at 60.248.175.38) State DESTROY
2010-10-18 08:34:19.494277 [DEBUG] mod_sofia.c:350 sofia/internal/1013 at 60.248.175.38 SOFIA DESTROY
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1013 at 60.248.175.38 Standard DESTROY
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:438 (sofia/internal/1013 at 60.248.175.38) State DESTROY going to sleep

Thank you for your attension
 
Best Regards
 
Gary
 


From: david.ponzone at ipeva.fr
To: freeswitch-users at lists.freeswitch.org
Date: Fri, 15 Oct 2010 11:16:41 +0200
Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite

You need to understand that if you dial a number that you want to be routed to your 60.248.175.37, you need NOT TO HAVE this number intercepted by the default context on 60.248.175.38.
In the log you sent, the call is staying local to 60.248.175.38.
It never reaches 60.248.175.37.


You need to have different numbering plan (you can't have a 4001 on both machines and hope to reach both with the same number).
Or you keep the same numbering plan, but you use a prefix in the dialplan to reach the other host.
For instance you would dial 994001, but your dialplan will match it with a regexp like ^99(4\d{3})$ and bridge $1 (so only 4001) to the other FS.





David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34


Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr  -   www.ipeva-studio.com



Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 15/10/2010 ? 10:52, ? Chiang ?? Chih-Chung Wybie a ?crit :

Hello this is my debug log
 
I use 1013 at 60.248.175.38 call to 4001 at 60.248.175.37
 
EXECUTE sofia/internal/1013 at 60.248.175.38 transfer(4001 XML default)
2010-10-15 16:49:30.666644 [DEBUG] switch_ivr.c:1444 (sofia/internal/1013 at 60.248.175.38) State Change CS_EXECUTE -> CS_ROUTING
2010-10-15 16:49:30.666644 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:30.666644 [DEBUG] switch_core_session.c:642 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:30.666644 [NOTICE] switch_ivr.c:1450 Transfer sofia/internal/1013 at 60.248.175.38 to XML[4001 at default]
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38) State EXECUTE going to sleep
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_ROUTING
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38) State ROUTING
2010-10-15 16:49:30.666644 [DEBUG] mod_sofia.c:140 sofia/internal/1013 at 60.248.175.38 SOFIA ROUTING
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1013 at 60.248.175.38Standard ROUTING
2010-10-15 16:49:30.666644 [INFO] mod_dialplan_xml.c:331 Processing 1013->4001 in context default
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->unloop] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->tod_example] continue=true
Dialplan: sofia/internal/1013 at 60.248.175.38 Date/Time Match (PASS) [tod_example] break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(open=true)
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->holiday_example] continue=true
Dialplan: sofia/internal/1013 at 60.248.175.38 Date/Time Match (FAIL) [holiday_example] break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->global-intercept] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [global-intercept] destination_number(4001) =~ /^886$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->group-intercept] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [group-intercept] destination_number(4001) =~ /^\*8$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->intercept-ext] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [intercept-ext] destination_number(4001) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->redial] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [redial] destination_number(4001) =~ /^(redial|870)$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->global] continue=true
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: sofia/internal/1013 at 60.248.175.38 Absolute Condition [global]
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-last_dial/global/${uuid})
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->snom-demo-2] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [snom-demo-2] destination_number(4001) =~ /^9001$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->snom-demo-1] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [snom-demo-1] destination_number(4001) =~ /^9000$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->eavesdrop] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [eavesdrop] destination_number(4001) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->eavesdrop] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [eavesdrop] destination_number(4001) =~ /^779$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->call_return] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call_return] destination_number(4001) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->del-group] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [del-group] destination_number(4001) =~ /^80(\d{2})$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->add-group] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [add-group] destination_number(4001) =~ /^81(\d{2})$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->call-group-simo] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call-group-simo] destination_number(4001) =~ /^82(\d{2})$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->call-group-order] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call-group-order] destination_number(4001) =~ /^83(\d{2})$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->extension-intercom] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [extension-intercom] destination_number(4001) =~ /^8(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->Local_Extension] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (PASS) [Local_Extension] destination_number(4001) =~ /^([0-4]0[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_COPYRIGHT=(c) 2010)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_SOFTWARE=FreeSwitch)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_ARTIST=FreeSwitch)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_COMMENT=FreeSwitch)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_DATE=${strftime(%Y-%m-%d %H:%M)})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_STEREO=true)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(dialed_extension=4001)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action export(dialed_extension=4001)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set_user(${dialed_extension}@${domain_name})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action limit(60.248.175.38 ${dialed_extension} ${max_calls} ${fail_over})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action bind_meta_app(1 b s execute_extension::dx XML features)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action bind_meta_app(3 b s execute_extension::cf XML features)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(ringback=${us-ring})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(transfer_ringback=local_stream://moh)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(call_timeout=30)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(hangup_after_bridge=true)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(continue_on_fail=true)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action bridge(user/${dialed_extension}@${domain_name})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action answer()
Dialplan: sofia/internal/1013 at 60.248.175.38 Action sleep(1000)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension})
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1013 at 60.248.175.38) State Change CS_ROUTING -> CS_EXECUTE
2010-10-15 16:49:30.666644 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38) State ROUTING going to sleep
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_EXECUTE
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38) State EXECUTE
2010-10-15 16:49:30.666644 [DEBUG] mod_sofia.c:233 sofia/internal/1013 at 60.248.175.38 SOFIA EXECUTE
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1013 at 60.248.175.38Standard EXECUTE
EXECUTE sofia/internal/1013 at 60.248.175.38 set(open=true)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [open]=[true]
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-spymap/1013/fe66f80a-992b-43f7-b130-47f0c87f4c15)
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-last_dial/1013/4001)
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-last_dial/global/fe66f80a-992b-43f7-b130-47f0c87f4c15)
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_COPYRIGHT=(c) 2010)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_COPYRIGHT]=[(c) 2010]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_SOFTWARE=FreeSwitch)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_SOFTWARE]=[FreeSwitch]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_ARTIST=FreeSwitch)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_ARTIST]=[FreeSwitch]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_COMMENT=FreeSwitch)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_COMMENT]=[FreeSwitch]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_DATE=2010-10-15 16:49)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_DATE]=[2010-10-15 16:49]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_STEREO=true)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_STEREO]=[true]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(dialed_extension=4001)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [dialed_extension]=[4001]
EXECUTE sofia/internal/1013 at 60.248.175.38 export(dialed_extension=4001)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:901 EXPORT [dialed_extension]=[4001]
EXECUTE sofia/internal/1013 at 60.248.175.38 set_user(4001 at 60.248.175.38)
EXECUTE sofia/internal/1013 at 60.248.175.38 limit(60.248.175.38 4001  )
2010-10-15 16:49:30.666644 [WARNING] mod_limit.c:779 USAGE: limit <realm> <id> [<max> [number  [dialplan [context]]]]
EXECUTE sofia/internal/1013 at 60.248.175.38 bind_meta_app(1 b s execute_extension::dx XML features)
2010-10-15 16:49:30.666644 [INFO] switch_ivr_async.c:2372 Bound B-Leg: 1 execute_extension::dx XML features
EXECUTE sofia/internal/1013 at 60.248.175.38 bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/1013.2010-10-15-16-49-30.wav)
2010-10-15 16:49:30.666644 [INFO] switch_ivr_async.c:2372 Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/1013.2010-10-15-16-49-30.wav
EXECUTE sofia/internal/1013 at 60.248.175.38 bind_meta_app(3 b s execute_extension::cf XML features)
2010-10-15 16:49:30.666644 [INFO] switch_ivr_async.c:2372 Bound B-Leg: 3 execute_extension::cf XML features
EXECUTE sofia/internal/1013 at 60.248.175.38 set(ringback=%(2000,4000,440.0,480.0))
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [ringback]=[%(2000,4000,440.0,480.0)]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(transfer_ringback=local_stream://moh)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [transfer_ringback]=[local_stream://moh]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(call_timeout=30)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [call_timeout]=[30]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(hangup_after_bridge=true)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(continue_on_fail=true)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [continue_on_fail]=[true]
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-call_return/4001/1013)
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-last_dial_ext/4001/fe66f80a-992b-43f7-b130-47f0c87f4c15)
EXECUTE sofia/internal/1013 at 60.248.175.38 set(called_party_callgroup=techsupport)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [called_party_callgroup]=[techsupport]
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-last_dial/techsupport/fe66f80a-992b-43f7-b130-47f0c87f4c15)
EXECUTE sofia/internal/1013 at 60.248.175.38 bridge(user/4001 at 60.248.175.38)
2010-10-15 16:49:30.666644 [DEBUG] switch_ivr_originate.c:1946 variable string 0 = [presence_id=4001 at 60.248.175.38]
2010-10-15 16:49:30.666644 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
2010-10-15 16:49:30.666644 [DEBUG] switch_ivr_originate.c:3308 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2010-10-15 16:49:30.666644 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
2010-10-15 16:49:30.666644 [DEBUG] switch_ivr_originate.c:3308 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2010-10-15 16:49:30.666644 [INFO] mod_dptools.c:2356 Originate Failed.  Cause: USER_NOT_REGISTERED
EXECUTE sofia/internal/1013 at 60.248.175.38 answer()
2010-10-15 16:49:30.694130 [DEBUG] sofia_glue.c:2616 AUDIO RTP [sofia/internal/1013 at 60.248.175.38] 60.248.175.38 port 25606 -> 60.248.175.43 port 35070 codec: 0 ms: 20
2010-10-15 16:49:30.694130 [DEBUG] switch_rtp.c:1343 Starting timer [soft] 160 bytes per 20ms
2010-10-15 16:49:30.695467 [DEBUG] sofia_glue.c:2822 Set 2833 dtmf send payload to 101
2010-10-15 16:49:30.695467 [DEBUG] sofia_glue.c:2827 Set 2833 dtmf receive payload to 101
2010-10-15 16:49:30.695467 [DEBUG] mod_sofia.c:663 Local SDP sofia/internal/1013 at 60.248.175.38:
v=0
o=FreeSWITCH 1287106964 1287106965 IN IP4 60.248.175.38
s=FreeSWITCH
c=IN IP4 60.248.175.38
t=0 0
m=audio 25606 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
2010-10-15 16:49:30.695467 [DEBUG] switch_core_session.c:642 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:30.695467 [NOTICE] mod_dptools.c:720 Channel [sofia/internal/1013 at 60.248.175.38] has been answered
EXECUTE sofia/internal/1013 at 60.248.175.38 sleep(1000)
2010-10-15 16:49:30.698172 [DEBUG] sofia.c:4200 Channel sofia/internal/1013 at 60.248.175.38 entering state [completed][200]
2010-10-15 16:49:30.745378 [DEBUG] switch_rtp.c:2443 Correct ip/port confirmed.
2010-10-15 16:49:30.807388 [DEBUG] sofia.c:4200 Channel sofia/internal/1013 at 60.248.175.38 entering state [ready][200]
EXECUTE sofia/internal/1013 at 60.248.175.38 bridge(loopback/app=voicemail:default 60.248.175.38 4001)
2010-10-15 16:49:31.704837 [NOTICE] switch_channel.c:675 New Channel loopback/app=voicemail:default 60.248.175.38 4001-a [23f3d5de-f985-44bd-8b1a-118b98bb5eae]
2010-10-15 16:49:31.704837 [DEBUG] mod_loopback.c:127 loopback/app=voicemail:default 60.248.175.38 4001-a setup codec PCMU/8000/20
2010-10-15 16:49:31.704837 [NOTICE] switch_channel.c:673 Rename Channel loopback/app=voicemail:default 60.248.175.38 4001-a->loopback/voicemail-a [23f3d5de-f985-44bd-8b1a-118b98bb5eae]
2010-10-15 16:49:31.704837 [DEBUG] mod_loopback.c:911 (loopback/voicemail-a) State Change CS_NEW -> CS_INIT
2010-10-15 16:49:31.704837 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.704837 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.706218 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-a) Running State Change CS_INIT
2010-10-15 16:49:31.706218 [DEBUG] switch_core_state_machine.c:338 (loopback/voicemail-a) State INIT
2010-10-15 16:49:31.706218 [NOTICE] switch_channel.c:675 New Channel loopback/voicemail-b [1abc53a2-df93-40db-96ab-547f9ddd1b1d]
2010-10-15 16:49:31.706218 [DEBUG] mod_loopback.c:127 loopback/voicemail-b setup codec PCMU/8000/20
2010-10-15 16:49:31.706218 [DEBUG] mod_loopback.c:240 (loopback/voicemail-b) State Change CS_NEW -> CS_INIT
2010-10-15 16:49:31.706218 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-b [BREAK]
2010-10-15 16:49:31.706218 [DEBUG] mod_loopback.c:452 loopback/voicemail-b CHANNEL KILL
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-b) Running State Change CS_INIT
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:338 (loopback/voicemail-b) State INIT
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:281 (loopback/voicemail-b) State Change CS_INIT -> CS_ROUTING
2010-10-15 16:49:31.707132 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-b [BREAK]
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:452 loopback/voicemail-b CHANNEL KILL
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:338 (loopback/voicemail-b) State INIT going to sleep
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-b) Running State Change CS_ROUTING
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:341 (loopback/voicemail-b) State ROUTING
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:315 loopback/voicemail-b CHANNEL ROUTING
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:326 (loopback/voicemail-b) State Change CS_ROUTING -> CS_EXECUTE
2010-10-15 16:49:31.707132 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-b [BREAK]
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:452 loopback/voicemail-b CHANNEL KILL
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:341 (loopback/voicemail-b) State ROUTING going to sleep
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-b) Running State Change CS_EXECUTE
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:348 (loopback/voicemail-b) State EXECUTE
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:346 loopback/voicemail-b CHANNEL EXECUTE
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:157 loopback/voicemail-b Standard EXECUTE
EXECUTE loopback/voicemail-b pre_answer()
2010-10-15 16:49:31.707132 [NOTICE] mod_loopback.c:716 Pre-Answer loopback/voicemail-a!
2010-10-15 16:49:31.707132 [DEBUG] switch_channel.c:2254 Send signal sofia/internal/1013 at 60.248.175.38[BREAK]
2010-10-15 16:49:31.707132 [DEBUG] switch_core_session.c:642 Send signal loopback/voicemail-b [BREAK]
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:452 loopback/voicemail-b CHANNEL KILL
2010-10-15 16:49:31.707132 [NOTICE] mod_dptools.c:746 Pre-Answer loopback/voicemail-b!
2010-10-15 16:49:31.707132 [DEBUG] switch_channel.c:2254 Send signal sofia/internal/1013 at 60.248.175.38[BREAK]
EXECUTE loopback/voicemail-b voicemail(default 60.248.175.38 4001)
2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:281 (loopback/voicemail-a) State Change CS_INIT -> CS_ROUTING
2010-10-15 16:49:31.708787 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:338 (loopback/voicemail-a) State INIT going to sleep
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-a) Running State Change CS_ROUTING
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:341 (loopback/voicemail-a) State ROUTING
2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:315 loopback/voicemail-a CHANNEL ROUTING
2010-10-15 16:49:31.708787 [DEBUG] switch_ivr_originate.c:66 (loopback/voicemail-a) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2010-10-15 16:49:31.708787 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:341 (loopback/voicemail-a) State ROUTING going to sleep
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-a) Running State Change CS_CONSUME_MEDIA
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:360 (loopback/voicemail-a) State CONSUME_MEDIA
2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:512 CHANNEL CONSUME_MEDIA
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:360 (loopback/voicemail-a) State CONSUME_MEDIA going to sleep
2010-10-15 16:49:31.710698 [DEBUG] switch_ivr_originate.c:3228 Originate Resulted in Success: [loopback/voicemail-a]
2010-10-15 16:49:31.710698 [DEBUG] switch_core_session.c:642 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.710698 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.710698 [DEBUG] switch_core_session.c:642 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:31.710698 [DEBUG] switch_ivr_bridge.c:1182 (loopback/voicemail-a) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2010-10-15 16:49:31.710698 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.710698 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.710698 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-a) Running State Change CS_EXCHANGE_MEDIA
2010-10-15 16:49:31.710698 [DEBUG] switch_core_state_machine.c:351 (loopback/voicemail-a) State EXCHANGE_MEDIA
2010-10-15 16:49:31.710698 [DEBUG] mod_loopback.c:474 CHANNEL LOOPBACK
2010-10-15 16:49:31.712661 [DEBUG] switch_core_session.c:703 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.712661 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.712661 [DEBUG] switch_core_session.c:703 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:31.824953 [DEBUG] switch_ivr_play_say.c:63 No language specified - Using [en]
2010-10-15 16:49:31.824953 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-person.wav] (en:en)
2010-10-15 16:49:31.824953 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16 at 8000hz 1 channels 20ms
2010-10-15 16:49:33.185370 [DEBUG] switch_ivr_play_say.c:1444 done playing file
2010-10-15 16:49:33.305429 [DEBUG] switch_ivr_play_say.c:244 Handle say:[4001] (en:en)
2010-10-15 16:49:33.305429 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16 at 8000hz 1 channels 20ms
2010-10-15 16:49:33.744961 [DEBUG] switch_ivr_play_say.c:1444 done playing file
2010-10-15 16:49:33.744961 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16 at 8000hz 1 channels 20ms

 


From: david.ponzone at ipeva.fr
To: freeswitch-users at lists.freeswitch.org
Date: Thu, 14 Oct 2010 12:20:56 +0200
Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite

Don't be shy on the logs, send the full ones!


in fs_cli:
fsctl loglevel debug
/log 7


and then make a call





David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34


Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr  -   www.ipeva-studio.com



Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 14/10/2010 ? 11:53, ? Chiang ?? Chih-Chung Wybie a ?crit :

I tried use 1013 at 38 calls to 1019 at 37
 
show log below
 
2010-10-14 17:50:41.249876 [NOTICE] switch_channel.c:675 New Channel sofia/internal/1013 at 60.248.175.38[875fdeef-b94e-41fb-a621-ea005bbaedbd]
2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing 1013->1019 in context public
2010-10-14 17:50:41.263809 [NOTICE] switch_ivr.c:1450 Transfer sofia/internal/1013 at 60.248.175.38 to XML[1019 at default]
2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing 1013->1019 in context default
2010-10-14 17:50:41.283904 [INFO] switch_core_session.c:1764 Sending early media
2010-10-14 17:50:41.285023 [NOTICE] mod_sofia.c:1994 Pre-Answer sofia/internal/1013 at 60.248.175.38!

seems not work
 
Thank you for you attension
Best Regards
 
Gary
 

 


From: david.ponzone at ipeva.fr
To: freeswitch-users at lists.freeswitch.org
Date: Thu, 14 Oct 2010 10:16:28 +0200
Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite

I think we would need a complete log from the target FS, in order to provide you with any efficient help.


Thank





David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34


Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr  -   www.ipeva-studio.com



Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 14/10/2010 ? 09:55, ? Chiang ?? Chih-Chung Wybie a ?crit :

I tried Connect Two FreeSWITCH Boxes  http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
and my configuration :
 
60.248.175.37
 
(autoload_configs/act.conf.xml)
<list name="domains" default="deny">
      <node type="allow" cidr="60.248.175.38/28"/>
     <!-- <node type="allow" domain="$${domain}"/>-->
</list>

(dialplan/default.xml)
 
<extension name="Dial to 38">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="bridge" data="sofia/internal/$1 at 60.248.175.38"/>
      </condition>
 </extension>
 
(dialplan/public.xml)

<extension name="Calls from 38">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="transfer" data="$1 XML default"/>
      </condition>
 </extension>
 
 
 
60.248.175.38
 
(autoload_configs/act.conf.xml)
<list name="domains" default="deny">
      <node type="allow" cidr="60.248.175.37/28"/>
     <!-- <node type="allow" domain="$${domain}"/>-->
</list>

(dialplan/default.xml)
 
<extension name="Dial to 37">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="bridge" data="sofia/internal/$1 at 60.248.175.37"/>
      </condition>
 </extension>
 
(dialplan/public.xml)

<extension name="Calls from 37">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="transfer" data="$1 XML default"/>
      </condition>
 </extension>
 
when I use 1007 at 38 calls to 1007 at 37
it won't show errors but still not work
 
how should I change something ?
 
Thank you for your attension
 
Best Regards
Gary
 
 
 
 
 
> Date: Wed, 13 Oct 2010 22:17:13 -0700
> From: curriegrad2004 at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> 
> You can create a SIP trunk between the two servers so calls can be
> routed back and forth from the servers you've specified.
> 
> 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
> > I tried it
> > but not working
> > and show errors below
> > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493 Cannot create
> > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493 Cannot create
> > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate Failed.
> > Cause: USER_NOT_REGISTERED
> >
> > and for example
> >  I use SIP server 60.248.175.38  user id=1007
> > and I call to 1007 at 60.248.175.37    that will show these errors
> >
> > how should set more?
> >
> > thank you for your attension
> >
> > Best Regards
> > Gary
> >
> > ________________________________
> > Date: Thu, 14 Oct 2010 10:55:49 +0800
> > From: xyangni at gmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >
> > I think you can reg at server A and than call B as a normal SIP call to
> > external domain.
> >
> > 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
> >
> >
> >
> > Hello
> >
> > I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X-Lite to call
> >
> > my question is how can I call between these servers?
> >
> > if I can , how shoould I set in *.xml? or X-Lite?
> >
> > thank you for your attension
> >
> >
> > Best Regards
> > Gary
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________ FreeSWITCH-users mailing
> > list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


_______________________________________________ FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org_______________________________________________
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http://www.freeswitch.org


_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 		 	   		  
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From tear152 at hotmail.com  Sun Oct 17 23:05:57 2010
From: tear152 at hotmail.com (=?gb2312?B?va0gQ2hpYW5nINbB1tAgQ2hpaC1DaHVuZyAgIFd5Ymll?=)
Date: Mon, 18 Oct 2010 14:05:57 +0800
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <SNT130-w445656F3A9C14ED384A1B0FB5A0@phx.gbl>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>,
	,,,,<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>,
	,,,,<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>,
	,,,,<SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>,
	, , , , <AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>, ,
	, , <SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>, , , 
	, <C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>,
	,,<SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>,
	, , <6A8D7D0E-4DF5-4F35-8A00-F52FECC4F826@ipeva.fr>, ,
	<SNT130-w6005F900786BDFBE48FF43FB570@phx.gbl>, ,
	<AD407430-9024-4E0D-A860-9026FD2837A3@ipeva.fr>,
	<SNT130-w445656F3A9C14ED384A1B0FB5A0@phx.gbl>
Message-ID: <SNT130-w38B1E85477162CFADB0E04FB5A0@phx.gbl>


60.248.175.38
 
<extension name="Local_Extension">
      <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="set" data="RECORD_COPYRIGHT=(c) 2010"/>
        <action application="set" data="RECORD_SOFTWARE=FreeSwitch"/>
        <action application="set" data="RECORD_ARTIST=FreeSwitch"/>
        <action application="set" data="RECORD_COMMENT=FreeSwitch"/>
        <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/>
        <action application="set" data="RECORD_STEREO=true"/>
       <!-- <action application="record_session" data="$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>-->
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <!-- add for busy transfer -->
        <action application="set_user" data="${dialed_extension}@${domain_name}"/>
        <action application="limit" data="$${domain_name} ${dialed_extension} ${max_calls} ${fail_over}"/>
        <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
        <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
        <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
        <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
        <action application="set" data="ringback=${us-ring}"/>
        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="set" data="call_timeout=30"/>
        <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> -->
        <action application="set" data="hangup_after_bridge=true"/>
        <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
        <action application="set" data="continue_on_fail=true"/>
        <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
        <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
        <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
        <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>-->
        <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
        <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
        <!-- <action application="record_fsv" data="$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.fsv"/>-->
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
      </condition>
    </extension>
    <extension name="Dial to 37">
      <condition field="destination_number" expression="^(40[01][0-9])$">
        <action application="bridge" data="sofia/internal/$1 at 60.248.175.37"/>
      </condition>
    </extension>

 
60.248.175.37
 
<extension name="Local_Extension">
      <condition field="destination_number" expression="^(40[01][0-9])$">
        <action application="set" data="RECORD_COPYRIGHT=(c) 2010"/>
        <action application="set" data="RECORD_SOFTWARE=FreeSwitch"/>
        <action application="set" data="RECORD_ARTIST=FreeSwitch"/>
        <action application="set" data="RECORD_COMMENT=FreeSwitch"/>
        <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/>
        <action application="set" data="RECORD_STEREO=true"/>
        <action application="record_session" data="$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <!-- add for busy transfer -->
        <action application="set_user" data="${dialed_extension}@${domain_name}"/>
        <action application="limit" data="$${domain_name} ${dialed_extension} ${max_calls} ${fail_over}"/>
        <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
        <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
        <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
        <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
        <action application="set" data="ringback=${us-ring}"/>
        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="set" data="call_timeout=30"/>
        <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> -->
        <action application="set" data="hangup_after_bridge=true"/>
        <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
        <action application="set" data="continue_on_fail=true"/>
        <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
        <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
        <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
        <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>-->
        <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
        <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
        <!-- <action application="record_fsv" data="$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.fsv"/>-->
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
      </condition>
    </extension>
    <extension name="Dial to 38">
      <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="bridge" data="sofia/internal/$1 at 60.248.175.38"/>
      </condition>
    </extension>
 
 
it still can't call to each other I have no idea.....
What should I do??
 
Thank you for your attension
 
Best Regards
 
Gary
 


From: tear152 at hotmail.com
To: freeswitch-users at lists.freeswitch.org
Date: Mon, 18 Oct 2010 08:37:00 +0800
Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite




I tried to set different expression but it still can't work
 
debug log below
 
2010-10-18 08:34:19.462617 [DEBUG] sofia.c:5849 IP 60.248.175.43 Approved by acl "domains[]". Access Granted.
2010-10-18 08:34:19.462617 [NOTICE] switch_channel.c:675 New Channel sofia/internal/1013 at 60.248.175.38 [1f4277c1-9ded-49c5-b63f-a312bead9d5f]
2010-10-18 08:34:19.464814 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_NEW
2010-10-18 08:34:19.464814 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1013 at 60.248.175.38) State NEW
2010-10-18 08:34:19.475647 [DEBUG] sofia.c:4200 Channel sofia/internal/1013 at 60.248.175.38 entering state [received][100]
2010-10-18 08:34:19.475647 [DEBUG] sofia.c:4211 Remote SDP:
v=0
o=- 0 2 IN IP4 60.248.175.43
s=CounterPath X-Lite 3.0
c=IN IP4 60.248.175.43
t=0 0
m=audio 32692 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 3 : d7A5MyJF tN44o7uR 60.248.175.43 32692
a=alt:2 2 : cCI83KQt qU62M7Ia 192.168.182.1 32692
a=alt:3 1 : qtl4Jt3z ug+kYltk 192.168.60.1 32692
2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[G7221:115:32000:20]
2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[G7221:107:16000:20]
2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[G722:9:8000:20]
2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[PCMU:0:8000:20]
2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[PCMA:8:8000:20]
2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[GSM:3:8000:20]
2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [PCMU:0:8000:20]/[G7221:115:32000:20]
2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [PCMU:0:8000:20]/[G7221:107:16000:20]
2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [PCMU:0:8000:20]/[G722:9:8000:20]
2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20]
2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:2376 Set Codec sofia/internal/1013 at 60.248.175.38 PCMU/8000 20 ms 160 samples
2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3610 Set 2833 dtmf send/recv payload to 101
2010-10-18 08:34:19.476688 [DEBUG] sofia.c:4357 (sofia/internal/1013 at 60.248.175.38) State Change CS_NEW -> CS_INIT
2010-10-18 08:34:19.476688 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_INIT
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1013 at 60.248.175.38) State INIT
2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:83 sofia/internal/1013 at 60.248.175.38 SOFIA INIT
2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:117 (sofia/internal/1013 at 60.248.175.38) State Change CS_INIT -> CS_ROUTING
2010-10-18 08:34:19.476688 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1013 at 60.248.175.38) State INIT going to sleep
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_ROUTING
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38) State ROUTING
2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:140 sofia/internal/1013 at 60.248.175.38 SOFIA ROUTING
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1013 at 60.248.175.38 Standard ROUTING
2010-10-18 08:34:19.476688 [INFO] mod_dialplan_xml.c:331 Processing 1013->4001 in context public
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->unloop] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->outside_call] continue=true
Dialplan: sofia/internal/1013 at 60.248.175.38 Absolute Condition [outside_call]
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(outside_call=true)
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->call_debug] continue=true
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->public_extensions] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [public_extensions] destination_number(4001) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->Calls from 37] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [Calls from 37] destination_number(4001) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->public_did] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [public_did] destination_number(4001) =~ /^(5551212)$/ break=on-false
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1013 at 60.248.175.38) State Change CS_ROUTING -> CS_EXECUTE
2010-10-18 08:34:19.476688 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38) State ROUTING going to sleep
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_EXECUTE
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38) State EXECUTE
2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:233 sofia/internal/1013 at 60.248.175.38 SOFIA EXECUTE
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1013 at 60.248.175.38 Standard EXECUTE
EXECUTE sofia/internal/1013 at 60.248.175.38 set(outside_call=true)
2010-10-18 08:34:19.476688 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [outside_call]=[true]
2010-10-18 08:34:19.476688 [NOTICE] switch_core_state_machine.c:185 sofia/internal/1013 at 60.248.175.38 has executed the last dialplan instruction, hanging up.
2010-10-18 08:34:19.476688 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/1013 at 60.248.175.38 [CS_EXECUTE] [NORMAL_CLEARING]
2010-10-18 08:34:19.476688 [DEBUG] switch_channel.c:2145 Send signal sofia/internal/1013 at 60.248.175.38 [KILL]
2010-10-18 08:34:19.476688 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38) State EXECUTE going to sleep
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_HANGUP
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/1013 at 60.248.175.38) State HANGUP
2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:441 Channel sofia/internal/1013 at 60.248.175.38 hanging up, cause: NORMAL_CLEARING
2010-10-18 08:34:19.494277 [DEBUG] mod_sofia.c:503 Responding to INVITE with: 480
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1013 at 60.248.175.38 Standard HANGUP, cause: NORMAL_CLEARING
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/1013 at 60.248.175.38) State HANGUP going to sleep
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/1013 at 60.248.175.38) State Change CS_HANGUP -> CS_REPORTING
2010-10-18 08:34:19.494277 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_REPORTING
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:589 (sofia/internal/1013 at 60.248.175.38) State REPORTING
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1013 at 60.248.175.38 Standard REPORTING, cause: NORMAL_CLEARING
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:589 (sofia/internal/1013 at 60.248.175.38) State REPORTING going to sleep
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/1013 at 60.248.175.38) State Change CS_REPORTING -> CS_DESTROY
2010-10-18 08:34:19.494277 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 08:34:19.494277 [DEBUG] switch_core_session.c:1170 Session 61 (sofia/internal/1013 at 60.248.175.38) Locked, Waiting on external entities
2010-10-18 08:34:19.494277 [NOTICE] switch_core_session.c:1188 Session 61 (sofia/internal/1013 at 60.248.175.38) Ended
2010-10-18 08:34:19.494277 [NOTICE] switch_core_session.c:1190 Close Channel sofia/internal/1013 at 60.248.175.38 [CS_DESTROY]
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:428 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_DESTROY
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:438 (sofia/internal/1013 at 60.248.175.38) State DESTROY
2010-10-18 08:34:19.494277 [DEBUG] mod_sofia.c:350 sofia/internal/1013 at 60.248.175.38 SOFIA DESTROY
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1013 at 60.248.175.38 Standard DESTROY
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:438 (sofia/internal/1013 at 60.248.175.38) State DESTROY going to sleep

Thank you for your attension
 
Best Regards
 
Gary
 


From: david.ponzone at ipeva.fr
To: freeswitch-users at lists.freeswitch.org
Date: Fri, 15 Oct 2010 11:16:41 +0200
Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite

You need to understand that if you dial a number that you want to be routed to your 60.248.175.37, you need NOT TO HAVE this number intercepted by the default context on 60.248.175.38. 
In the log you sent, the call is staying local to 60.248.175.38.
It never reaches 60.248.175.37.


You need to have different numbering plan (you can't have a 4001 on both machines and hope to reach both with the same number).
Or you keep the same numbering plan, but you use a prefix in the dialplan to reach the other host.
For instance you would dial 994001, but your dialplan will match it with a regexp like ^99(4\d{3})$ and bridge $1 (so only 4001) to the other FS.





David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34


Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr  -   www.ipeva-studio.com



Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 15/10/2010 ? 10:52, ? Chiang ?? Chih-Chung Wybie a ?crit :

Hello this is my debug log
 
I use 1013 at 60.248.175.38 call to 4001 at 60.248.175.37
 
EXECUTE sofia/internal/1013 at 60.248.175.38 transfer(4001 XML default)
2010-10-15 16:49:30.666644 [DEBUG] switch_ivr.c:1444 (sofia/internal/1013 at 60.248.175.38) State Change CS_EXECUTE -> CS_ROUTING
2010-10-15 16:49:30.666644 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:30.666644 [DEBUG] switch_core_session.c:642 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:30.666644 [NOTICE] switch_ivr.c:1450 Transfer sofia/internal/1013 at 60.248.175.38 to XML[4001 at default]
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38) State EXECUTE going to sleep
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_ROUTING
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38) State ROUTING
2010-10-15 16:49:30.666644 [DEBUG] mod_sofia.c:140 sofia/internal/1013 at 60.248.175.38 SOFIA ROUTING
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1013 at 60.248.175.38Standard ROUTING
2010-10-15 16:49:30.666644 [INFO] mod_dialplan_xml.c:331 Processing 1013->4001 in context default
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->unloop] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->tod_example] continue=true
Dialplan: sofia/internal/1013 at 60.248.175.38 Date/Time Match (PASS) [tod_example] break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(open=true)
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->holiday_example] continue=true
Dialplan: sofia/internal/1013 at 60.248.175.38 Date/Time Match (FAIL) [holiday_example] break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->global-intercept] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [global-intercept] destination_number(4001) =~ /^886$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->group-intercept] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [group-intercept] destination_number(4001) =~ /^\*8$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->intercept-ext] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [intercept-ext] destination_number(4001) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->redial] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [redial] destination_number(4001) =~ /^(redial|870)$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->global] continue=true
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: sofia/internal/1013 at 60.248.175.38 Absolute Condition [global]
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-last_dial/global/${uuid})
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->snom-demo-2] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [snom-demo-2] destination_number(4001) =~ /^9001$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->snom-demo-1] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [snom-demo-1] destination_number(4001) =~ /^9000$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->eavesdrop] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [eavesdrop] destination_number(4001) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->eavesdrop] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [eavesdrop] destination_number(4001) =~ /^779$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->call_return] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call_return] destination_number(4001) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->del-group] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [del-group] destination_number(4001) =~ /^80(\d{2})$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->add-group] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [add-group] destination_number(4001) =~ /^81(\d{2})$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->call-group-simo] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call-group-simo] destination_number(4001) =~ /^82(\d{2})$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->call-group-order] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call-group-order] destination_number(4001) =~ /^83(\d{2})$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->extension-intercom] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [extension-intercom] destination_number(4001) =~ /^8(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->Local_Extension] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (PASS) [Local_Extension] destination_number(4001) =~ /^([0-4]0[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_COPYRIGHT=(c) 2010)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_SOFTWARE=FreeSwitch)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_ARTIST=FreeSwitch)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_COMMENT=FreeSwitch)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_DATE=${strftime(%Y-%m-%d %H:%M)})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_STEREO=true)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(dialed_extension=4001)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action export(dialed_extension=4001)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set_user(${dialed_extension}@${domain_name})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action limit(60.248.175.38 ${dialed_extension} ${max_calls} ${fail_over})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action bind_meta_app(1 b s execute_extension::dx XML features)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action bind_meta_app(3 b s execute_extension::cf XML features)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(ringback=${us-ring})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(transfer_ringback=local_stream://moh)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(call_timeout=30)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(hangup_after_bridge=true)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(continue_on_fail=true)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action bridge(user/${dialed_extension}@${domain_name})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action answer()
Dialplan: sofia/internal/1013 at 60.248.175.38 Action sleep(1000)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension})
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1013 at 60.248.175.38) State Change CS_ROUTING -> CS_EXECUTE
2010-10-15 16:49:30.666644 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38) State ROUTING going to sleep
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_EXECUTE
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38) State EXECUTE
2010-10-15 16:49:30.666644 [DEBUG] mod_sofia.c:233 sofia/internal/1013 at 60.248.175.38 SOFIA EXECUTE
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1013 at 60.248.175.38Standard EXECUTE
EXECUTE sofia/internal/1013 at 60.248.175.38 set(open=true)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [open]=[true]
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-spymap/1013/fe66f80a-992b-43f7-b130-47f0c87f4c15)
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-last_dial/1013/4001)
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-last_dial/global/fe66f80a-992b-43f7-b130-47f0c87f4c15)
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_COPYRIGHT=(c) 2010)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_COPYRIGHT]=[(c) 2010]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_SOFTWARE=FreeSwitch)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_SOFTWARE]=[FreeSwitch]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_ARTIST=FreeSwitch)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_ARTIST]=[FreeSwitch]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_COMMENT=FreeSwitch)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_COMMENT]=[FreeSwitch]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_DATE=2010-10-15 16:49)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_DATE]=[2010-10-15 16:49]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_STEREO=true)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_STEREO]=[true]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(dialed_extension=4001)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [dialed_extension]=[4001]
EXECUTE sofia/internal/1013 at 60.248.175.38 export(dialed_extension=4001)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:901 EXPORT [dialed_extension]=[4001]
EXECUTE sofia/internal/1013 at 60.248.175.38 set_user(4001 at 60.248.175.38)
EXECUTE sofia/internal/1013 at 60.248.175.38 limit(60.248.175.38 4001  )
2010-10-15 16:49:30.666644 [WARNING] mod_limit.c:779 USAGE: limit <realm> <id> [<max> [number  [dialplan [context]]]]
EXECUTE sofia/internal/1013 at 60.248.175.38 bind_meta_app(1 b s execute_extension::dx XML features)
2010-10-15 16:49:30.666644 [INFO] switch_ivr_async.c:2372 Bound B-Leg: 1 execute_extension::dx XML features
EXECUTE sofia/internal/1013 at 60.248.175.38 bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/1013.2010-10-15-16-49-30.wav)
2010-10-15 16:49:30.666644 [INFO] switch_ivr_async.c:2372 Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/1013.2010-10-15-16-49-30.wav
EXECUTE sofia/internal/1013 at 60.248.175.38 bind_meta_app(3 b s execute_extension::cf XML features)
2010-10-15 16:49:30.666644 [INFO] switch_ivr_async.c:2372 Bound B-Leg: 3 execute_extension::cf XML features
EXECUTE sofia/internal/1013 at 60.248.175.38 set(ringback=%(2000,4000,440.0,480.0))
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [ringback]=[%(2000,4000,440.0,480.0)]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(transfer_ringback=local_stream://moh)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [transfer_ringback]=[local_stream://moh]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(call_timeout=30)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [call_timeout]=[30]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(hangup_after_bridge=true)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(continue_on_fail=true)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [continue_on_fail]=[true]
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-call_return/4001/1013)
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-last_dial_ext/4001/fe66f80a-992b-43f7-b130-47f0c87f4c15)
EXECUTE sofia/internal/1013 at 60.248.175.38 set(called_party_callgroup=techsupport)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [called_party_callgroup]=[techsupport]
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-last_dial/techsupport/fe66f80a-992b-43f7-b130-47f0c87f4c15)
EXECUTE sofia/internal/1013 at 60.248.175.38 bridge(user/4001 at 60.248.175.38)
2010-10-15 16:49:30.666644 [DEBUG] switch_ivr_originate.c:1946 variable string 0 = [presence_id=4001 at 60.248.175.38]
2010-10-15 16:49:30.666644 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
2010-10-15 16:49:30.666644 [DEBUG] switch_ivr_originate.c:3308 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2010-10-15 16:49:30.666644 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
2010-10-15 16:49:30.666644 [DEBUG] switch_ivr_originate.c:3308 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2010-10-15 16:49:30.666644 [INFO] mod_dptools.c:2356 Originate Failed.  Cause: USER_NOT_REGISTERED
EXECUTE sofia/internal/1013 at 60.248.175.38 answer()
2010-10-15 16:49:30.694130 [DEBUG] sofia_glue.c:2616 AUDIO RTP [sofia/internal/1013 at 60.248.175.38] 60.248.175.38 port 25606 -> 60.248.175.43 port 35070 codec: 0 ms: 20
2010-10-15 16:49:30.694130 [DEBUG] switch_rtp.c:1343 Starting timer [soft] 160 bytes per 20ms
2010-10-15 16:49:30.695467 [DEBUG] sofia_glue.c:2822 Set 2833 dtmf send payload to 101
2010-10-15 16:49:30.695467 [DEBUG] sofia_glue.c:2827 Set 2833 dtmf receive payload to 101
2010-10-15 16:49:30.695467 [DEBUG] mod_sofia.c:663 Local SDP sofia/internal/1013 at 60.248.175.38:
v=0
o=FreeSWITCH 1287106964 1287106965 IN IP4 60.248.175.38
s=FreeSWITCH
c=IN IP4 60.248.175.38
t=0 0
m=audio 25606 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
2010-10-15 16:49:30.695467 [DEBUG] switch_core_session.c:642 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:30.695467 [NOTICE] mod_dptools.c:720 Channel [sofia/internal/1013 at 60.248.175.38] has been answered
EXECUTE sofia/internal/1013 at 60.248.175.38 sleep(1000)
2010-10-15 16:49:30.698172 [DEBUG] sofia.c:4200 Channel sofia/internal/1013 at 60.248.175.38 entering state [completed][200]
2010-10-15 16:49:30.745378 [DEBUG] switch_rtp.c:2443 Correct ip/port confirmed.
2010-10-15 16:49:30.807388 [DEBUG] sofia.c:4200 Channel sofia/internal/1013 at 60.248.175.38 entering state [ready][200]
EXECUTE sofia/internal/1013 at 60.248.175.38 bridge(loopback/app=voicemail:default 60.248.175.38 4001)
2010-10-15 16:49:31.704837 [NOTICE] switch_channel.c:675 New Channel loopback/app=voicemail:default 60.248.175.38 4001-a [23f3d5de-f985-44bd-8b1a-118b98bb5eae]
2010-10-15 16:49:31.704837 [DEBUG] mod_loopback.c:127 loopback/app=voicemail:default 60.248.175.38 4001-a setup codec PCMU/8000/20
2010-10-15 16:49:31.704837 [NOTICE] switch_channel.c:673 Rename Channel loopback/app=voicemail:default 60.248.175.38 4001-a->loopback/voicemail-a [23f3d5de-f985-44bd-8b1a-118b98bb5eae]
2010-10-15 16:49:31.704837 [DEBUG] mod_loopback.c:911 (loopback/voicemail-a) State Change CS_NEW -> CS_INIT
2010-10-15 16:49:31.704837 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.704837 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.706218 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-a) Running State Change CS_INIT
2010-10-15 16:49:31.706218 [DEBUG] switch_core_state_machine.c:338 (loopback/voicemail-a) State INIT
2010-10-15 16:49:31.706218 [NOTICE] switch_channel.c:675 New Channel loopback/voicemail-b [1abc53a2-df93-40db-96ab-547f9ddd1b1d]
2010-10-15 16:49:31.706218 [DEBUG] mod_loopback.c:127 loopback/voicemail-b setup codec PCMU/8000/20
2010-10-15 16:49:31.706218 [DEBUG] mod_loopback.c:240 (loopback/voicemail-b) State Change CS_NEW -> CS_INIT
2010-10-15 16:49:31.706218 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-b [BREAK]
2010-10-15 16:49:31.706218 [DEBUG] mod_loopback.c:452 loopback/voicemail-b CHANNEL KILL
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-b) Running State Change CS_INIT
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:338 (loopback/voicemail-b) State INIT
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:281 (loopback/voicemail-b) State Change CS_INIT -> CS_ROUTING
2010-10-15 16:49:31.707132 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-b [BREAK]
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:452 loopback/voicemail-b CHANNEL KILL
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:338 (loopback/voicemail-b) State INIT going to sleep
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-b) Running State Change CS_ROUTING
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:341 (loopback/voicemail-b) State ROUTING
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:315 loopback/voicemail-b CHANNEL ROUTING
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:326 (loopback/voicemail-b) State Change CS_ROUTING -> CS_EXECUTE
2010-10-15 16:49:31.707132 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-b [BREAK]
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:452 loopback/voicemail-b CHANNEL KILL
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:341 (loopback/voicemail-b) State ROUTING going to sleep
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-b) Running State Change CS_EXECUTE
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:348 (loopback/voicemail-b) State EXECUTE
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:346 loopback/voicemail-b CHANNEL EXECUTE
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:157 loopback/voicemail-b Standard EXECUTE
EXECUTE loopback/voicemail-b pre_answer()
2010-10-15 16:49:31.707132 [NOTICE] mod_loopback.c:716 Pre-Answer loopback/voicemail-a!
2010-10-15 16:49:31.707132 [DEBUG] switch_channel.c:2254 Send signal sofia/internal/1013 at 60.248.175.38[BREAK]
2010-10-15 16:49:31.707132 [DEBUG] switch_core_session.c:642 Send signal loopback/voicemail-b [BREAK]
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:452 loopback/voicemail-b CHANNEL KILL
2010-10-15 16:49:31.707132 [NOTICE] mod_dptools.c:746 Pre-Answer loopback/voicemail-b!
2010-10-15 16:49:31.707132 [DEBUG] switch_channel.c:2254 Send signal sofia/internal/1013 at 60.248.175.38[BREAK]
EXECUTE loopback/voicemail-b voicemail(default 60.248.175.38 4001)
2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:281 (loopback/voicemail-a) State Change CS_INIT -> CS_ROUTING
2010-10-15 16:49:31.708787 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:338 (loopback/voicemail-a) State INIT going to sleep
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-a) Running State Change CS_ROUTING
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:341 (loopback/voicemail-a) State ROUTING
2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:315 loopback/voicemail-a CHANNEL ROUTING
2010-10-15 16:49:31.708787 [DEBUG] switch_ivr_originate.c:66 (loopback/voicemail-a) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2010-10-15 16:49:31.708787 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:341 (loopback/voicemail-a) State ROUTING going to sleep
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-a) Running State Change CS_CONSUME_MEDIA
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:360 (loopback/voicemail-a) State CONSUME_MEDIA
2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:512 CHANNEL CONSUME_MEDIA
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:360 (loopback/voicemail-a) State CONSUME_MEDIA going to sleep
2010-10-15 16:49:31.710698 [DEBUG] switch_ivr_originate.c:3228 Originate Resulted in Success: [loopback/voicemail-a]
2010-10-15 16:49:31.710698 [DEBUG] switch_core_session.c:642 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.710698 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.710698 [DEBUG] switch_core_session.c:642 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:31.710698 [DEBUG] switch_ivr_bridge.c:1182 (loopback/voicemail-a) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2010-10-15 16:49:31.710698 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.710698 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.710698 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-a) Running State Change CS_EXCHANGE_MEDIA
2010-10-15 16:49:31.710698 [DEBUG] switch_core_state_machine.c:351 (loopback/voicemail-a) State EXCHANGE_MEDIA
2010-10-15 16:49:31.710698 [DEBUG] mod_loopback.c:474 CHANNEL LOOPBACK
2010-10-15 16:49:31.712661 [DEBUG] switch_core_session.c:703 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.712661 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.712661 [DEBUG] switch_core_session.c:703 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:31.824953 [DEBUG] switch_ivr_play_say.c:63 No language specified - Using [en]
2010-10-15 16:49:31.824953 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-person.wav] (en:en)
2010-10-15 16:49:31.824953 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16 at 8000hz 1 channels 20ms
2010-10-15 16:49:33.185370 [DEBUG] switch_ivr_play_say.c:1444 done playing file
2010-10-15 16:49:33.305429 [DEBUG] switch_ivr_play_say.c:244 Handle say:[4001] (en:en)
2010-10-15 16:49:33.305429 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16 at 8000hz 1 channels 20ms
2010-10-15 16:49:33.744961 [DEBUG] switch_ivr_play_say.c:1444 done playing file
2010-10-15 16:49:33.744961 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16 at 8000hz 1 channels 20ms

 


From: david.ponzone at ipeva.fr
To: freeswitch-users at lists.freeswitch.org
Date: Thu, 14 Oct 2010 12:20:56 +0200
Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite

Don't be shy on the logs, send the full ones! 


in fs_cli:
fsctl loglevel debug
/log 7


and then make a call





David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34


Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr  -   www.ipeva-studio.com



Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 14/10/2010 ? 11:53, ? Chiang ?? Chih-Chung Wybie a ?crit :

I tried use 1013 at 38 calls to 1019 at 37
 
show log below
 
2010-10-14 17:50:41.249876 [NOTICE] switch_channel.c:675 New Channel sofia/internal/1013 at 60.248.175.38[875fdeef-b94e-41fb-a621-ea005bbaedbd]
2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing 1013->1019 in context public
2010-10-14 17:50:41.263809 [NOTICE] switch_ivr.c:1450 Transfer sofia/internal/1013 at 60.248.175.38 to XML[1019 at default]
2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing 1013->1019 in context default
2010-10-14 17:50:41.283904 [INFO] switch_core_session.c:1764 Sending early media
2010-10-14 17:50:41.285023 [NOTICE] mod_sofia.c:1994 Pre-Answer sofia/internal/1013 at 60.248.175.38!

seems not work
 
Thank you for you attension
Best Regards
 
Gary
 

 


From: david.ponzone at ipeva.fr
To: freeswitch-users at lists.freeswitch.org
Date: Thu, 14 Oct 2010 10:16:28 +0200
Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite

I think we would need a complete log from the target FS, in order to provide you with any efficient help. 


Thank





David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34


Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr  -   www.ipeva-studio.com



Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 14/10/2010 ? 09:55, ? Chiang ?? Chih-Chung Wybie a ?crit :

I tried Connect Two FreeSWITCH Boxes  http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
and my configuration :
 
60.248.175.37
 
(autoload_configs/act.conf.xml)
<list name="domains" default="deny">
      <node type="allow" cidr="60.248.175.38/28"/>
     <!-- <node type="allow" domain="$${domain}"/>-->
</list>

(dialplan/default.xml)
 
<extension name="Dial to 38">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="bridge" data="sofia/internal/$1 at 60.248.175.38"/>
      </condition>
 </extension>
 
(dialplan/public.xml)

<extension name="Calls from 38">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="transfer" data="$1 XML default"/>
      </condition>
 </extension>
 
 
 
60.248.175.38
 
(autoload_configs/act.conf.xml)
<list name="domains" default="deny">
      <node type="allow" cidr="60.248.175.37/28"/>
     <!-- <node type="allow" domain="$${domain}"/>-->
</list>

(dialplan/default.xml)
 
<extension name="Dial to 37">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="bridge" data="sofia/internal/$1 at 60.248.175.37"/>
      </condition>
 </extension>
 
(dialplan/public.xml)

<extension name="Calls from 37">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="transfer" data="$1 XML default"/>
      </condition>
 </extension>
 
when I use 1007 at 38 calls to 1007 at 37
it won't show errors but still not work
 
how should I change something ?
 
Thank you for your attension
 
Best Regards
Gary
 
 
 
 
 
> Date: Wed, 13 Oct 2010 22:17:13 -0700
> From: curriegrad2004 at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> 
> You can create a SIP trunk between the two servers so calls can be
> routed back and forth from the servers you've specified.
> 
> 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
> > I tried it
> > but not working
> > and show errors below
> > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493 Cannot create
> > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493 Cannot create
> > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate Failed.
> > Cause: USER_NOT_REGISTERED
> >
> > and for example
> >  I use SIP server 60.248.175.38  user id=1007
> > and I call to 1007 at 60.248.175.37    that will show these errors
> >
> > how should set more?
> >
> > thank you for your attension
> >
> > Best Regards
> > Gary
> >
> > ________________________________
> > Date: Thu, 14 Oct 2010 10:55:49 +0800
> > From: xyangni at gmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >
> > I think you can reg at server A and than call B as a normal SIP call to
> > external domain.
> >
> > 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
> >
> >
> >
> > Hello
> >
> > I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X-Lite to call
> >
> > my question is how can I call between these servers?
> >
> > if I can , how shoould I set in *.xml? or X-Lite?
> >
> > thank you for your attension
> >
> >
> > Best Regards
> > Gary
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________ FreeSWITCH-users mailing
> > list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From elihayunfs at gmail.com  Mon Oct 18 00:15:45 2010
From: elihayunfs at gmail.com (Eli Hayun)
Date: Mon, 18 Oct 2010 09:15:45 +0200
Subject: [Freeswitch-users] How to change the language in conference
Message-ID: <AANLkTinAW1JpK4QjtOu_CNv==-2aEcj1vOPa=3FSsv73@mail.gmail.com>

Hi
1) I am trying to change the default language in the conference but no
success
I did : set default_language=ru but I still hear everything in English

2) How to change the language when I am running
api:execute("conference"....) in Lua?

Thanks
-- 
Eli Hayun
Hebrew University Jerusalem
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From david.ponzone at ipeva.fr  Mon Oct 18 00:25:25 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 18 Oct 2010 09:25:25 +0200
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <SNT130-w445656F3A9C14ED384A1B0FB5A0@phx.gbl>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>, , , ,
	<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>, , , ,
	<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>, ,
	, , <SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>, , , ,
	<AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>, ,
	, <SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>, , ,
	<C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>, ,
	<SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>, ,
	<6A8D7D0E-4DF5-4F35-8A00-F52FECC4F826@ipeva.fr>,
	<SNT130-w6005F900786BDFBE48FF43FB570@phx.gbl>,
	<AD407430-9024-4E0D-A860-9026FD2837A3@ipeva.fr>
	<SNT130-w445656F3A9C14ED384A1B0FB5A0@phx.gbl>
Message-ID: <056892B2-7226-4941-AF71-4CC5ABB01835@ipeva.fr>

You failed to tell us which server this log comes from.

In order to get a result, I think you need to try to understand what  
the logs mean and you need to understand that to help you, we need  
detailed logs and relevant config files.
 From the beginning of this thread, you failed to provide both at the  
same time.

Also, that's not a basic config.
Have you tried before to do a single server config ?
I would start there, so that you can understand how FreeSWITCH works,  
why the logs are meaning ful and how to read them, which dialplans are  
parsed and when.
If you don't do that, you're going to waste people's time because you  
want to avoid some required (and important) learning steps.

Thank you

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 18/10/2010 ? 02:37, ? Chiang ?? Chih-Chung Wybie a ?crit :

> I tried to set different expression but it still can't work
>
> debug log below
>
> 2010-10-18 08:34:19.462617 [DEBUG] sofia.c:5849 IP 60.248.175.43  
> Approved by acl "domains[]". Access Granted.
> 2010-10-18 08:34:19.462617 [NOTICE] switch_channel.c:675 New Channel sofia/internal/1013 at 60.248.175.38 
> [1f4277c1-9ded-49c5-b63f-a312bead9d5f]
> 2010-10-18 08:34:19.464814 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_NEW
> 2010-10-18 08:34:19.464814 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1013 at 60.248.175.38 
> ) State NEW
> 2010-10-18 08:34:19.475647 [DEBUG] sofia.c:4200 Channel sofia/internal/1013 at 60.248.175.38 
>  entering state [received][100]
> 2010-10-18 08:34:19.475647 [DEBUG] sofia.c:4211 Remote SDP:
> v=0
> o=- 0 2 IN IP4 60.248.175.43
> s=CounterPath X-Lite 3.0
> c=IN IP4 60.248.175.43
> t=0 0
> m=audio 32692 RTP/AVP 107 0 8 101
> a=rtpmap:107 BV32/16000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=alt:1 3 : d7A5MyJF tN44o7uR 60.248.175.43 32692
> a=alt:2 2 : cCI83KQt qU62M7Ia 192.168.182.1 32692
> a=alt:3 1 : qtl4Jt3z ug+kYltk 192.168.60.1 32692
> 2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[G7221:115:32000:20]
> 2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[G7221:107:16000:20]
> 2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[G722:9:8000:20]
> 2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[PCMU:0:8000:20]
> 2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[PCMA:8:8000:20]
> 2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[GSM:3:8000:20]
> 2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [PCMU:0:8000:20]/[G7221:115:32000:20]
> 2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [PCMU:0:8000:20]/[G7221:107:16000:20]
> 2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [PCMU:0:8000:20]/[G722:9:8000:20]
> 2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [PCMU:0:8000:20]/[PCMU:0:8000:20]
> 2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:2376 Set Codec sofia/internal/1013 at 60.248.175.38PCMU 
> /8000 20 ms 160 samples
> 2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3610 Set 2833 dtmf  
> send/recv payload to 101
> 2010-10-18 08:34:19.476688 [DEBUG] sofia.c:4357 (sofia/internal/1013 at 60.248.175.38 
> ) State Change CS_NEW -> CS_INIT
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_INIT
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1013 at 60.248.175.38 
> ) State INIT
> 2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:83 sofia/internal/1013 at 60.248.175.38 
>  SOFIA INIT
> 2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:117 (sofia/internal/1013 at 60.248.175.38 
> ) State Change CS_INIT -> CS_ROUTING
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1013 at 60.248.175.38 
> ) State INIT going to sleep
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_ROUTING
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38 
> ) State ROUTING
> 2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:140 sofia/internal/1013 at 60.248.175.38 
>  SOFIA ROUTING
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1013 at 60.248.175.38Standard 
>  ROUTING
> 2010-10-18 08:34:19.476688 [INFO] mod_dialplan_xml.c:331 Processing  
> 1013->4001 in context public
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->unloop]  
> continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (PASS) [unloop] $ 
> {unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [unloop] $ 
> {sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public- 
> >outside_call] continue=true
> Dialplan: sofia/internal/1013 at 60.248.175.38 Absolute Condition  
> [outside_call]
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(outside_call=true)
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public- 
> >call_debug] continue=true
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL)  
> [call_debug] ${call_debug}(false) =~ /^true$/ break=never
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public- 
> >public_extensions] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL)  
> [public_extensions] destination_number(4001) =~ /^(10[01][0-9])$/  
> break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->Calls  
> from 37] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [Calls from  
> 37] destination_number(4001) =~ /^(10[01][0-9])$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public- 
> >public_did] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL)  
> [public_did] destination_number(4001) =~ /^(5551212)$/ break=on-false
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1013 at 60.248.175.38 
> ) State Change CS_ROUTING -> CS_EXECUTE
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38 
> ) State ROUTING going to sleep
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_EXECUTE
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38 
> ) State EXECUTE
> 2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:233 sofia/internal/1013 at 60.248.175.38 
>  SOFIA EXECUTE
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1013 at 60.248.175.38Standard 
>  EXECUTE
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(outside_call=true)
> 2010-10-18 08:34:19.476688 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [outside_call]=[true]
> 2010-10-18 08:34:19.476688 [NOTICE] switch_core_state_machine.c:185 sofia/internal/1013 at 60.248.175.38has 
>  executed the last dialplan instruction, hanging up.
> 2010-10-18 08:34:19.476688 [NOTICE] switch_core_state_machine.c:187 Hangupsofia/internal/1013 at 60.248.175.38 
>  [CS_EXECUTE] [NORMAL_CLEARING]
> 2010-10-18 08:34:19.476688 [DEBUG] switch_channel.c:2145 Send signal sofia/internal/1013 at 60.248.175.38 
> [KILL]
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38 
> ) State EXECUTE going to sleep
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_HANGUP
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/1013 at 60.248.175.38 
> ) State HANGUP
> 2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:441 Channel sofia/internal/1013 at 60.248.175.38 
>  hanging up, cause: NORMAL_CLEARING
> 2010-10-18 08:34:19.494277 [DEBUG] mod_sofia.c:503 Responding to  
> INVITE with: 480
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1013 at 60.248.175.38Standard 
>  HANGUP, cause: NORMAL_CLEARING
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/1013 at 60.248.175.38 
> ) State HANGUP going to sleep
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/1013 at 60.248.175.38 
> ) State Change CS_HANGUP -> CS_REPORTING
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_REPORTING
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:589 (sofia/internal/1013 at 60.248.175.38 
> ) State REPORTING
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1013 at 60.248.175.38Standard 
>  REPORTING, cause: NORMAL_CLEARING
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:589 (sofia/internal/1013 at 60.248.175.38 
> ) State REPORTING going to sleep
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/1013 at 60.248.175.38 
> ) State Change CS_REPORTING -> CS_DESTROY
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_session.c:1170  
> Session 61 (sofia/internal/1013 at 60.248.175.38) Locked, Waiting on  
> external entities
> 2010-10-18 08:34:19.494277 [NOTICE] switch_core_session.c:1188  
> Session 61 (sofia/internal/1013 at 60.248.175.38) Ended
> 2010-10-18 08:34:19.494277 [NOTICE] switch_core_session.c:1190 Close Channelsofia/internal/1013 at 60.248.175.38 
>  [CS_DESTROY]
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:428 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_DESTROY
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:438 (sofia/internal/1013 at 60.248.175.38 
> ) State DESTROY
> 2010-10-18 08:34:19.494277 [DEBUG] mod_sofia.c:350 sofia/internal/1013 at 60.248.175.38 
>  SOFIA DESTROY
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1013 at 60.248.175.38Standard 
>  DESTROY
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:438 (sofia/internal/1013 at 60.248.175.38 
> ) State DESTROY going to sleep
>
> Thank you for your attension
>
> Best Regards
>
> Gary
>
> From: david.ponzone at ipeva.fr
> To: freeswitch-users at lists.freeswitch.org
> Date: Fri, 15 Oct 2010 11:16:41 +0200
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> You need to understand that if you dial a number that you want to be  
> routed to your 60.248.175.37, you need NOT TO HAVE this number  
> intercepted by the default context on 60.248.175.38.
> In the log you sent, the call is staying local to 60.248.175.38.
> It never reaches 60.248.175.37.
>
> You need to have different numbering plan (you can't have a 4001 on  
> both machines and hope to reach both with the same number).
> Or you keep the same numbering plan, but you use a prefix in the  
> dialplan to reach the other host.
> For instance you would dial 994001, but your dialplan will match it  
> with a regexp like ^99(4\d{3})$ and bridge $1 (so only 4001) to the  
> other FS.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
> message, merci de le d?truire imm?diatement et d'avertir  
> l'exp?diteur.
>
>
>
>
> Le 15/10/2010 ? 10:52, ? Chiang ?? Chih-Chung Wybie a ?crit :
>
> Hello this is my debug log
>
> I use 1013 at 60.248.175.38 call to 4001 at 60.248.175.37
>
> EXECUTE sofia/internal/1013 at 60.248.175.38 transfer(4001 XML default)
> 2010-10-15 16:49:30.666644 [DEBUG] switch_ivr.c:1444 (sofia/internal/1013 at 60.248.175.38 
> ) State Change CS_EXECUTE -> CS_ROUTING
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_session.c:642 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-15 16:49:30.666644 [NOTICE] switch_ivr.c:1450 Transfer sofia/internal/1013 at 60.248.175.38 
>  to XML[4001 at default]
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38 
> ) State EXECUTE going to sleep
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_ROUTING
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38 
> ) State ROUTING
> 2010-10-15 16:49:30.666644 [DEBUG] mod_sofia.c:140 sofia/internal/1013 at 60.248.175.38 
>  SOFIA ROUTING
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1013 at 60.248.175.38Standard 
>  ROUTING
> 2010-10-15 16:49:30.666644 [INFO] mod_dialplan_xml.c:331 Processing  
> 1013->4001 in context default
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >unloop] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (PASS) [unloop] $ 
> {unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [unloop] $ 
> {sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >tod_example] continue=true
> Dialplan: sofia/internal/1013 at 60.248.175.38 Date/Time Match (PASS)  
> [tod_example] break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(open=true)
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >holiday_example] continue=true
> Dialplan: sofia/internal/1013 at 60.248.175.38 Date/Time Match (FAIL)  
> [holiday_example] break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->global- 
> intercept] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [global- 
> intercept] destination_number(4001) =~ /^886$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->group- 
> intercept] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [group- 
> intercept] destination_number(4001) =~ /^\*8$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >intercept-ext] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [intercept- 
> ext] destination_number(4001) =~ /^\*\*(\d+)$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >redial] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [redial]  
> destination_number(4001) =~ /^(redial|870)$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >global] continue=true
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [global] $ 
> {call_debug}(false) =~ /^true$/ break=never
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [global] $ 
> {sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32| 
> AES_CM_128_HMAC_SHA1_80)$/ break=never
> Dialplan: sofia/internal/1013 at 60.248.175.38 Absolute Condition  
> [global]
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/$ 
> {domain_name}-spymap/${caller_id_number}/${uuid})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/$ 
> {domain_name}-last_dial/${caller_id_number}/${destination_number})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/$ 
> {domain_name}-last_dial/global/${uuid})
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->snom- 
> demo-2] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [snom- 
> demo-2] destination_number(4001) =~ /^9001$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->snom- 
> demo-1] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [snom- 
> demo-1] destination_number(4001) =~ /^9000$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >eavesdrop] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [eavesdrop]  
> destination_number(4001) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >eavesdrop] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [eavesdrop]  
> destination_number(4001) =~ /^779$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >call_return] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL)  
> [call_return] destination_number(4001) =~ /^\*69$|^869$|^lcr$/  
> break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->del- 
> group] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [del-group]  
> destination_number(4001) =~ /^80(\d{2})$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->add- 
> group] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [add-group]  
> destination_number(4001) =~ /^81(\d{2})$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->call- 
> group-simo] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call-group- 
> simo] destination_number(4001) =~ /^82(\d{2})$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->call- 
> group-order] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call-group- 
> order] destination_number(4001) =~ /^83(\d{2})$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >extension-intercom] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [extension- 
> intercom] destination_number(4001) =~ /^8(10[01][0-9])$/ break=on- 
> false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >Local_Extension] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (PASS)  
> [Local_Extension] destination_number(4001) =~ /^([0-4]0[01][0-9])$/  
> break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(RECORD_COPYRIGHT=(c) 2010)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(RECORD_SOFTWARE=FreeSwitch)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(RECORD_ARTIST=FreeSwitch)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(RECORD_COMMENT=FreeSwitch)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_DATE=$ 
> {strftime(%Y-%m-%d %H:%M)})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(RECORD_STEREO=true)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(dialed_extension=4001)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> export(dialed_extension=4001)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action set_user($ 
> {dialed_extension}@${domain_name})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> limit(60.248.175.38 ${dialed_extension} ${max_calls} ${fail_over})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action bind_meta_app(1 b  
> s execute_extension::dx XML features)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action bind_meta_app(2 b  
> s record_session::/usr/local/freeswitch/recordings/$ 
> {caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action bind_meta_app(3 b  
> s execute_extension::cf XML features)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(ringback=${us- 
> ring})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(transfer_ringback=local_stream://moh)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(call_timeout=30)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(hangup_after_bridge=true)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(continue_on_fail=true)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/$ 
> {domain_name}-call_return/${dialed_extension}/${caller_id_number})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/$ 
> {domain_name}-last_dial_ext/${dialed_extension}/${uuid})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(called_party_callgroup=${user_data(${dialed_extension}@$ 
> {domain_name} var callgroup)})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/$ 
> {domain_name}-last_dial/${called_party_callgroup}/${uuid})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action bridge(user/$ 
> {dialed_extension}@${domain_name})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action answer()
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action sleep(1000)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action bridge(loopback/ 
> app=voicemail:default ${domain_name} ${dialed_extension})
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1013 at 60.248.175.38 
> ) State Change CS_ROUTING -> CS_EXECUTE
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38 
> ) State ROUTING going to sleep
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_EXECUTE
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38 
> ) State EXECUTE
> 2010-10-15 16:49:30.666644 [DEBUG] mod_sofia.c:233 sofia/internal/1013 at 60.248.175.38 
>  SOFIA EXECUTE
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1013 at 60.248.175.38Standard 
>  EXECUTE
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(open=true)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [open]=[true]
> EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38- 
> spymap/1013/fe66f80a-992b-43f7-b130-47f0c87f4c15)
> EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38- 
> last_dial/1013/4001)
> EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38- 
> last_dial/global/fe66f80a-992b-43f7-b130-47f0c87f4c15)
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_COPYRIGHT=(c)  
> 2010)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [RECORD_COPYRIGHT]=[(c) 2010]
> EXECUTE sofia/internal/1013 at 60.248.175.38  
> set(RECORD_SOFTWARE=FreeSwitch)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [RECORD_SOFTWARE]=[FreeSwitch]
> EXECUTE sofia/internal/1013 at 60.248.175.38  
> set(RECORD_ARTIST=FreeSwitch)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [RECORD_ARTIST]=[FreeSwitch]
> EXECUTE sofia/internal/1013 at 60.248.175.38  
> set(RECORD_COMMENT=FreeSwitch)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [RECORD_COMMENT]=[FreeSwitch]
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_DATE=2010-10-15  
> 16:49)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [RECORD_DATE]=[2010-10-15 16:49]
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_STEREO=true)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [RECORD_STEREO]=[true]
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(dialed_extension=4001)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [dialed_extension]=[4001]
> EXECUTE sofia/internal/1013 at 60.248.175.38  
> export(dialed_extension=4001)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:901 EXPORT  
> [dialed_extension]=[4001]
> EXECUTE sofia/internal/1013 at 60.248.175.38 set_user(4001 at 60.248.175.38)
> EXECUTE sofia/internal/1013 at 60.248.175.38 limit(60.248.175.38 4001  )
> 2010-10-15 16:49:30.666644 [WARNING] mod_limit.c:779 USAGE: limit  
> <realm> <id> [<max> [number  [dialplan [context]]]]
> EXECUTE sofia/internal/1013 at 60.248.175.38 bind_meta_app(1 b s  
> execute_extension::dx XML features)
> 2010-10-15 16:49:30.666644 [INFO] switch_ivr_async.c:2372 Bound B- 
> Leg: 1 execute_extension::dx XML features
> EXECUTE sofia/internal/1013 at 60.248.175.38 bind_meta_app(2 b s  
> record_session::/usr/local/freeswitch/recordings/ 
> 1013.2010-10-15-16-49-30.wav)
> 2010-10-15 16:49:30.666644 [INFO] switch_ivr_async.c:2372 Bound B- 
> Leg: 2 record_session::/usr/local/freeswitch/recordings/ 
> 1013.2010-10-15-16-49-30.wav
> EXECUTE sofia/internal/1013 at 60.248.175.38 bind_meta_app(3 b s  
> execute_extension::cf XML features)
> 2010-10-15 16:49:30.666644 [INFO] switch_ivr_async.c:2372 Bound B- 
> Leg: 3 execute_extension::cf XML features
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(ringback= 
> %(2000,4000,440.0,480.0))
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [ringback]=[%(2000,4000,440.0,480.0)]
> EXECUTE sofia/internal/1013 at 60.248.175.38  
> set(transfer_ringback=local_stream://moh)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [transfer_ringback]=[local_stream://moh]
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(call_timeout=30)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [call_timeout]=[30]
> EXECUTE sofia/internal/1013 at 60.248.175.38  
> set(hangup_after_bridge=true)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [hangup_after_bridge]=[true]
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(continue_on_fail=true)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [continue_on_fail]=[true]
> EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38- 
> call_return/4001/1013)
> EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38- 
> last_dial_ext/4001/fe66f80a-992b-43f7-b130-47f0c87f4c15)
> EXECUTE sofia/internal/1013 at 60.248.175.38  
> set(called_party_callgroup=techsupport)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [called_party_callgroup]=[techsupport]
> EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38- 
> last_dial/techsupport/fe66f80a-992b-43f7-b130-47f0c87f4c15)
> EXECUTE sofia/internal/1013 at 60.248.175.38 bridge(user/4001 at 60.248.175.38 
> )
> 2010-10-15 16:49:30.666644 [DEBUG] switch_ivr_originate.c:1946  
> variable string 0 = [presence_id=4001 at 60.248.175.38]
> 2010-10-15 16:49:30.666644 [ERR] switch_ivr_originate.c:2493 Cannot  
> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> 2010-10-15 16:49:30.666644 [DEBUG] switch_ivr_originate.c:3308  
> Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
> 2010-10-15 16:49:30.666644 [ERR] switch_ivr_originate.c:2493 Cannot  
> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> 2010-10-15 16:49:30.666644 [DEBUG] switch_ivr_originate.c:3308  
> Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
> 2010-10-15 16:49:30.666644 [INFO] mod_dptools.c:2356 Originate  
> Failed.  Cause: USER_NOT_REGISTERED
> EXECUTE sofia/internal/1013 at 60.248.175.38 answer()
> 2010-10-15 16:49:30.694130 [DEBUG] sofia_glue.c:2616 AUDIO RTP [sofia/internal/1013 at 60.248.175.38 
> ] 60.248.175.38 port 25606 -> 60.248.175.43 port 35070 codec: 0 ms: 20
> 2010-10-15 16:49:30.694130 [DEBUG] switch_rtp.c:1343 Starting timer  
> [soft] 160 bytes per 20ms
> 2010-10-15 16:49:30.695467 [DEBUG] sofia_glue.c:2822 Set 2833 dtmf  
> send payload to 101
> 2010-10-15 16:49:30.695467 [DEBUG] sofia_glue.c:2827 Set 2833 dtmf  
> receive payload to 101
> 2010-10-15 16:49:30.695467 [DEBUG] mod_sofia.c:663 Local SDP sofia/internal/1013 at 60.248.175.38 
> :
> v=0
> o=FreeSWITCH 1287106964 1287106965 IN IP4 60.248.175.38
> s=FreeSWITCH
> c=IN IP4 60.248.175.38
> t=0 0
> m=audio 25606 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> 2010-10-15 16:49:30.695467 [DEBUG] switch_core_session.c:642 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-15 16:49:30.695467 [NOTICE] mod_dptools.c:720 Channel [sofia/internal/1013 at 60.248.175.38 
> ] has been answered
> EXECUTE sofia/internal/1013 at 60.248.175.38 sleep(1000)
> 2010-10-15 16:49:30.698172 [DEBUG] sofia.c:4200 Channel sofia/internal/1013 at 60.248.175.38 
>  entering state [completed][200]
> 2010-10-15 16:49:30.745378 [DEBUG] switch_rtp.c:2443 Correct ip/port  
> confirmed.
> 2010-10-15 16:49:30.807388 [DEBUG] sofia.c:4200 Channel sofia/internal/1013 at 60.248.175.38 
>  entering state [ready][200]
> EXECUTE sofia/internal/1013 at 60.248.175.38 bridge(loopback/ 
> app=voicemail:default 60.248.175.38 4001)
> 2010-10-15 16:49:31.704837 [NOTICE] switch_channel.c:675 New Channel  
> loopback/app=voicemail:default 60.248.175.38 4001-a [23f3d5de- 
> f985-44bd-8b1a-118b98bb5eae]
> 2010-10-15 16:49:31.704837 [DEBUG] mod_loopback.c:127 loopback/ 
> app=voicemail:default 60.248.175.38 4001-a setup codec PCMU/8000/20
> 2010-10-15 16:49:31.704837 [NOTICE] switch_channel.c:673 Rename  
> Channel loopback/app=voicemail:default 60.248.175.38 4001-a- 
> >loopback/voicemail-a [23f3d5de-f985-44bd-8b1a-118b98bb5eae]
> 2010-10-15 16:49:31.704837 [DEBUG] mod_loopback.c:911 (loopback/ 
> voicemail-a) State Change CS_NEW -> CS_INIT
> 2010-10-15 16:49:31.704837 [DEBUG] switch_core_session.c:1022 Send  
> signal loopback/voicemail-a [BREAK]
> 2010-10-15 16:49:31.704837 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-a CHANNEL KILL
> 2010-10-15 16:49:31.706218 [DEBUG] switch_core_state_machine.c:314  
> (loopback/voicemail-a) Running State Change CS_INIT
> 2010-10-15 16:49:31.706218 [DEBUG] switch_core_state_machine.c:338  
> (loopback/voicemail-a) State INIT
> 2010-10-15 16:49:31.706218 [NOTICE] switch_channel.c:675 New Channel  
> loopback/voicemail-b [1abc53a2-df93-40db-96ab-547f9ddd1b1d]
> 2010-10-15 16:49:31.706218 [DEBUG] mod_loopback.c:127 loopback/ 
> voicemail-b setup codec PCMU/8000/20
> 2010-10-15 16:49:31.706218 [DEBUG] mod_loopback.c:240 (loopback/ 
> voicemail-b) State Change CS_NEW -> CS_INIT
> 2010-10-15 16:49:31.706218 [DEBUG] switch_core_session.c:1022 Send  
> signal loopback/voicemail-b [BREAK]
> 2010-10-15 16:49:31.706218 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-b CHANNEL KILL
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:314  
> (loopback/voicemail-b) Running State Change CS_INIT
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:338  
> (loopback/voicemail-b) State INIT
> 2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:281 (loopback/ 
> voicemail-b) State Change CS_INIT -> CS_ROUTING
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_session.c:1022 Send  
> signal loopback/voicemail-b [BREAK]
> 2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-b CHANNEL KILL
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:338  
> (loopback/voicemail-b) State INIT going to sleep
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:314  
> (loopback/voicemail-b) Running State Change CS_ROUTING
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:341  
> (loopback/voicemail-b) State ROUTING
> 2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:315 loopback/ 
> voicemail-b CHANNEL ROUTING
> 2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:326 (loopback/ 
> voicemail-b) State Change CS_ROUTING -> CS_EXECUTE
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_session.c:1022 Send  
> signal loopback/voicemail-b [BREAK]
> 2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-b CHANNEL KILL
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:341  
> (loopback/voicemail-b) State ROUTING going to sleep
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:314  
> (loopback/voicemail-b) Running State Change CS_EXECUTE
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:348  
> (loopback/voicemail-b) State EXECUTE
> 2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:346 loopback/ 
> voicemail-b CHANNEL EXECUTE
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:157  
> loopback/voicemail-b Standard EXECUTE
> EXECUTE loopback/voicemail-b pre_answer()
> 2010-10-15 16:49:31.707132 [NOTICE] mod_loopback.c:716 Pre-Answer  
> loopback/voicemail-a!
> 2010-10-15 16:49:31.707132 [DEBUG] switch_channel.c:2254 Send signal sofia/internal/1013 at 60.248.175.38 
> [BREAK]
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_session.c:642 Send  
> signal loopback/voicemail-b [BREAK]
> 2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-b CHANNEL KILL
> 2010-10-15 16:49:31.707132 [NOTICE] mod_dptools.c:746 Pre-Answer  
> loopback/voicemail-b!
> 2010-10-15 16:49:31.707132 [DEBUG] switch_channel.c:2254 Send signal sofia/internal/1013 at 60.248.175.38 
> [BREAK]
> EXECUTE loopback/voicemail-b voicemail(default 60.248.175.38 4001)
> 2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:281 (loopback/ 
> voicemail-a) State Change CS_INIT -> CS_ROUTING
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_session.c:1022 Send  
> signal loopback/voicemail-a [BREAK]
> 2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-a CHANNEL KILL
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:338  
> (loopback/voicemail-a) State INIT going to sleep
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:314  
> (loopback/voicemail-a) Running State Change CS_ROUTING
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:341  
> (loopback/voicemail-a) State ROUTING
> 2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:315 loopback/ 
> voicemail-a CHANNEL ROUTING
> 2010-10-15 16:49:31.708787 [DEBUG] switch_ivr_originate.c:66  
> (loopback/voicemail-a) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_session.c:1022 Send  
> signal loopback/voicemail-a [BREAK]
> 2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-a CHANNEL KILL
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:341  
> (loopback/voicemail-a) State ROUTING going to sleep
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:314  
> (loopback/voicemail-a) Running State Change CS_CONSUME_MEDIA
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:360  
> (loopback/voicemail-a) State CONSUME_MEDIA
> 2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:512 CHANNEL  
> CONSUME_MEDIA
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:360  
> (loopback/voicemail-a) State CONSUME_MEDIA going to sleep
> 2010-10-15 16:49:31.710698 [DEBUG] switch_ivr_originate.c:3228  
> Originate Resulted in Success: [loopback/voicemail-a]
> 2010-10-15 16:49:31.710698 [DEBUG] switch_core_session.c:642 Send  
> signal loopback/voicemail-a [BREAK]
> 2010-10-15 16:49:31.710698 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-a CHANNEL KILL
> 2010-10-15 16:49:31.710698 [DEBUG] switch_core_session.c:642 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-15 16:49:31.710698 [DEBUG] switch_ivr_bridge.c:1182  
> (loopback/voicemail-a) State Change CS_CONSUME_MEDIA ->  
> CS_EXCHANGE_MEDIA
> 2010-10-15 16:49:31.710698 [DEBUG] switch_core_session.c:1022 Send  
> signal loopback/voicemail-a [BREAK]
> 2010-10-15 16:49:31.710698 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-a CHANNEL KILL
> 2010-10-15 16:49:31.710698 [DEBUG] switch_core_state_machine.c:314  
> (loopback/voicemail-a) Running State Change CS_EXCHANGE_MEDIA
> 2010-10-15 16:49:31.710698 [DEBUG] switch_core_state_machine.c:351  
> (loopback/voicemail-a) State EXCHANGE_MEDIA
> 2010-10-15 16:49:31.710698 [DEBUG] mod_loopback.c:474 CHANNEL LOOPBACK
> 2010-10-15 16:49:31.712661 [DEBUG] switch_core_session.c:703 Send  
> signal loopback/voicemail-a [BREAK]
> 2010-10-15 16:49:31.712661 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-a CHANNEL KILL
> 2010-10-15 16:49:31.712661 [DEBUG] switch_core_session.c:703 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-15 16:49:31.824953 [DEBUG] switch_ivr_play_say.c:63 No  
> language specified - Using [en]
> 2010-10-15 16:49:31.824953 [DEBUG] switch_ivr_play_say.c:244 Handle  
> play-file:[voicemail/vm-person.wav] (en:en)
> 2010-10-15 16:49:31.824953 [DEBUG] switch_ivr_play_say.c:1152 Codec  
> Activated L16 at 8000hz 1 channels 20ms
> 2010-10-15 16:49:33.185370 [DEBUG] switch_ivr_play_say.c:1444 done  
> playing file
> 2010-10-15 16:49:33.305429 [DEBUG] switch_ivr_play_say.c:244 Handle  
> say:[4001] (en:en)
> 2010-10-15 16:49:33.305429 [DEBUG] switch_ivr_play_say.c:1152 Codec  
> Activated L16 at 8000hz 1 channels 20ms
> 2010-10-15 16:49:33.744961 [DEBUG] switch_ivr_play_say.c:1444 done  
> playing file
> 2010-10-15 16:49:33.744961 [DEBUG] switch_ivr_play_say.c:1152 Codec  
> Activated L16 at 8000hz 1 channels 20ms
>
>
> From: david.ponzone at ipeva.fr
> To: freeswitch-users at lists.freeswitch.org
> Date: Thu, 14 Oct 2010 12:20:56 +0200
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> Don't be shy on the logs, send the full ones!
>
> in fs_cli:
> fsctl loglevel debug
> /log 7
>
> and then make a call
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
> message, merci de le d?truire imm?diatement et d'avertir  
> l'exp?diteur.
>
>
>
>
> Le 14/10/2010 ? 11:53, ? Chiang ?? Chih-Chung Wybie a ?crit :
>
> I tried use 1013 at 38 calls to 1019 at 37
>
> show log below
>
> 2010-10-14 17:50:41.249876 [NOTICE] switch_channel.c:675 New Channel sofia/internal/1013 at 60.248.175.38 
> [875fdeef-b94e-41fb-a621-ea005bbaedbd]
> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing  
> 1013->1019 in context public
> 2010-10-14 17:50:41.263809 [NOTICE] switch_ivr.c:1450 Transfer sofia/internal/1013 at 60.248.175.38 
>  to XML[1019 at default]
> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing  
> 1013->1019 in context default
> 2010-10-14 17:50:41.283904 [INFO] switch_core_session.c:1764 Sending  
> early media
> 2010-10-14 17:50:41.285023 [NOTICE] mod_sofia.c:1994 Pre-Answer sofia/internal/1013 at 60.248.175.38 
> !
>
> seems not work
>
> Thank you for you attension
> Best Regards
>
> Gary
>
>
>
> From: david.ponzone at ipeva.fr
> To: freeswitch-users at lists.freeswitch.org
> Date: Thu, 14 Oct 2010 10:16:28 +0200
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> I think we would need a complete log from the target FS, in order to  
> provide you with any efficient help.
>
> Thank
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
> message, merci de le d?truire imm?diatement et d'avertir  
> l'exp?diteur.
>
>
>
>
> Le 14/10/2010 ? 09:55, ? Chiang ?? Chih-Chung Wybie a ?crit :
>
> I tried Connect Two FreeSWITCH Boxes  http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
> and my configuration :
>
> 60.248.175.37
>
> (autoload_configs/act.conf.xml)
> <list name="domains" default="deny">
>       <node type="allow" cidr="60.248.175.38/28"/>
>      <!-- <node type="allow" domain="$${domain}"/>-->
> </list>
>
> (dialplan/default.xml)
>
> <extension name="Dial to 38">
>       <condition field="destination_number" expression="^([0-4]0[01] 
> [0-9])$">
>         <action application="bridge" data="sofia/internal/$1 at 60.248.175.38 
> "/>
>       </condition>
>  </extension>
>
> (dialplan/public.xml)
>
> <extension name="Calls from 38">
>       <condition field="destination_number" expression="^([0-4]0[01] 
> [0-9])$">
>         <action application="transfer" data="$1 XML default"/>
>       </condition>
>  </extension>
>
>
>
> 60.248.175.38
>
> (autoload_configs/act.conf.xml)
> <list name="domains" default="deny">
>       <node type="allow" cidr="60.248.175.37/28"/>
>      <!-- <node type="allow" domain="$${domain}"/>-->
> </list>
>
> (dialplan/default.xml)
>
> <extension name="Dial to 37">
>       <condition field="destination_number" expression="^([0-4]0[01] 
> [0-9])$">
>         <action application="bridge" data="sofia/internal/$1 at 60.248.175.37 
> "/>
>       </condition>
>  </extension>
>
> (dialplan/public.xml)
>
> <extension name="Calls from 37">
>       <condition field="destination_number" expression="^([0-4]0[01] 
> [0-9])$">
>         <action application="transfer" data="$1 XML default"/>
>       </condition>
>  </extension>
>
> when I use 1007 at 38 calls to 1007 at 37
> it won't show errors but still not work
>
> how should I change something ?
>
> Thank you for your attension
>
> Best Regards
> Gary
>
>
>
>
>
> > Date: Wed, 13 Oct 2010 22:17:13 -0700
> > From: curriegrad2004 at gmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X- 
> Lite
> >
> > You can create a SIP trunk between the two servers so calls can be
> > routed back and forth from the servers you've specified.
> >
> > 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
> > > I tried it
> > > but not working
> > > and show errors below
> > > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493  
> Cannot create
> > > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> > > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493  
> Cannot create
> > > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> > > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate  
> Failed.
> > > Cause: USER_NOT_REGISTERED
> > >
> > > and for example
> > >  I use SIP server 60.248.175.38  user id=1007
> > > and I call to 1007 at 60.248.175.37    that will show these errors
> > >
> > > how should set more?
> > >
> > > thank you for your attension
> > >
> > > Best Regards
> > > Gary
> > >
> > > ________________________________
> > > Date: Thu, 14 Oct 2010 10:55:49 +0800
> > > From: xyangni at gmail.com
> > > To: freeswitch-users at lists.freeswitch.org
> > > Subject: Re: [Freeswitch-users] some questions of freeswitch & X- 
> Lite
> > >
> > > I think you can reg at server A and than call B as a normal SIP  
> call to
> > > external domain.
> > >
> > > 2010/10/14 ? Chiang ?? Chih-Chung Wybie  
> <tear152 at hotmail.com>
> > >
> > >
> > >
> > > Hello
> > >
> > > I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X- 
> Lite to call
> > >
> > > my question is how can I call between these servers?
> > >
> > > if I can , how shoould I set in *.xml? or X-Lite?
> > >
> > > thank you for your attension
> > >
> > >
> > > Best Regards
> > > Gary
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> > >
> > > _______________________________________________ FreeSWITCH-users  
> mailing
> > > list FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________ FreeSWITCH-users  
> mailing listFreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
>  http:// 
> www.freeswitch.org_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________ FreeSWITCH-users  
> mailing listFreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
>  http:// 
> www.freeswitch.org_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________ FreeSWITCH-users  
> mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
>  http:// 
> www.freeswitch.org_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From tear152 at hotmail.com  Mon Oct 18 01:05:53 2010
From: tear152 at hotmail.com (=?gb2312?B?va0gQ2hpYW5nINbB1tAgQ2hpaC1DaHVuZyAgIFd5Ymll?=)
Date: Mon, 18 Oct 2010 16:05:53 +0800
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <056892B2-7226-4941-AF71-4CC5ABB01835@ipeva.fr>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>, , ,
	,,<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>, , ,
	, , <AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>, , , ,
	, <SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>, , ,
	,,<AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>, ,,,
	<SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>, ,
	,,<C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>,
	,,<SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>,
	, , <6A8D7D0E-4DF5-4F35-8A00-F52FECC4F826@ipeva.fr>, ,
	<SNT130-w6005F900786BDFBE48FF43FB570@phx.gbl>, ,
	<AD407430-9024-4E0D-A860-9026FD2837A3@ipeva.fr>,
	<SNT130-w445656F3A9C14ED384A1B0FB5A0@phx.gbl>,
	<056892B2-7226-4941-AF71-4CC5ABB01835@ipeva.fr>
Message-ID: <SNT130-w60DD13A21F1F843BC143E6FB5A0@phx.gbl>


sorry
 
but I tried it on a single server it can work
 
and I refer to http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes to set connect two FSs
 
and my all configs modified below
1, acl.conf.xml
60.248.175.37
<list name="domains" default="deny">
     <node type="allow" cidr="60.248.175.38/28"/>
     <!--<node type="allow" domain="$${domain}"/>-->
    </list>

60.248.175.38
<list name="domains" default="deny">
     <node type="allow" cidr="60.248.175.37/28"/>
     <!--<node type="allow" domain="$${domain}"/>-->
    </list>

2, dialplan/default.xml
60.248.175.37
<extension name="Local_Extension">
      <condition field="destination_number" expression="^(40[01][0-9])$">
        <action application="set" data="RECORD_COPYRIGHT=(c) 2010"/>
        <action application="set" data="RECORD_SOFTWARE=FreeSwitch"/>
        <action application="set" data="RECORD_ARTIST=FreeSwitch"/>
        <action application="set" data="RECORD_COMMENT=FreeSwitch"/>
        <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/>
        <action application="set" data="RECORD_STEREO=true"/>
        <action application="record_session" data="$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <!-- add for busy transfer -->
        <action application="set_user" data="${dialed_extension}@${domain_name}"/>
        <action application="limit" data="$${domain_name} ${dialed_extension} ${max_calls} ${fail_over}"/>
        <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
        <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
        <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
        <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
        <action application="set" data="ringback=${us-ring}"/>
        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="set" data="call_timeout=30"/>
        <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> -->
        <action application="set" data="hangup_after_bridge=true"/>
        <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
        <action application="set" data="continue_on_fail=true"/>
        <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
        <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
        <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
        <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>-->
        <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
        <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
        <!-- <action application="record_fsv" data="$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.fsv"/>-->
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
      </condition>
    </extension>
    <extension name="Dial to 38">
      <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="bridge" data="sofia/internal/$1 at 60.248.175.38"/>
      </condition>
    </extension>

 60.248.175.38
<extension name="Local_Extension">
      <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="set" data="RECORD_COPYRIGHT=(c) 2010"/>
        <action application="set" data="RECORD_SOFTWARE=FreeSwitch"/>
        <action application="set" data="RECORD_ARTIST=FreeSwitch"/>
        <action application="set" data="RECORD_COMMENT=FreeSwitch"/>
        <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/>
        <action application="set" data="RECORD_STEREO=true"/>
       <!-- <action application="record_session" data="$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>-->
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <!-- add for busy transfer -->
        <action application="set_user" data="${dialed_extension}@${domain_name}"/>
        <action application="limit" data="$${domain_name} ${dialed_extension} ${max_calls} ${fail_over}"/>
        <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
        <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
        <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
        <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
        <action application="set" data="ringback=${us-ring}"/>
        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="set" data="call_timeout=30"/>
        <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> -->
        <action application="set" data="hangup_after_bridge=true"/>
        <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
        <action application="set" data="continue_on_fail=true"/>
        <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
        <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
        <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
        <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>-->
        <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
        <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
        <!-- <action application="record_fsv" data="$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.fsv"/>-->
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
      </condition>
    </extension>
    <extension name="Dial to 37">
      <condition field="destination_number" expression="^(40[01][0-9])$">
        <action application="bridge" data="sofia/internal/$1 at 60.248.175.37"/>
      </condition>
    </extension>

3, dialplan/public.xml
60.248.175.37
<extension name="Calls from 38">
      <condition field="destination_number" expression="^(40[01][0-9])$">
        <action application="transfer" data="$1 XML default"/>
      </condition>
    </extension>

60.248.175.38
<extension name="Calls from 37">
      <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="transfer" data="$1 XML default"/>
      </condition>
    </extension>


4,DEBUG log
I use 1013 at 64.248.175.38 calls to 4001 at 64.248.175.37
log from 38 (that's all)
 
freeswitch at ubuntu> 2010-10-18 15:56:15.221328 [DEBUG] sofia.c:5849 IP 60.248.175.43 Approved by acl "domains[]". Access Granted.
2010-10-18 15:56:15.221328 [NOTICE] switch_channel.c:675 New Channel sofia/internal/1013 at 60.248.175.38 [bfc190d6-9371-4bfb-9822-2cbfaf1d6ebc]
2010-10-18 15:56:15.224062 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_NEW
2010-10-18 15:56:15.224062 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1013 at 60.248.175.38) State NEW
2010-10-18 15:56:15.236344 [DEBUG] sofia.c:4200 Channel sofia/internal/1013 at 60.248.175.38 entering state [received][100]
2010-10-18 15:56:15.236344 [DEBUG] sofia.c:4211 Remote SDP:
v=0
o=- 6 2 IN IP4 60.248.175.43
s=CounterPath X-Lite 3.0
c=IN IP4 60.248.175.43
t=0 0
m=audio 41530 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 3 : KOojNg3s EbMN6/84 60.248.175.43 41530
a=alt:2 2 : FZp3J381 HADuy4Pa 192.168.182.1 41530
a=alt:3 1 : Zq7Ue+wV Zj2BjStZ 192.168.60.1 41530
2010-10-18 15:56:15.236344 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[G7221:115:32000:20]
2010-10-18 15:56:15.236344 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[G7221:107:16000:20]
2010-10-18 15:56:15.236344 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[G722:9:8000:20]
2010-10-18 15:56:15.236344 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[PCMU:0:8000:20]
2010-10-18 15:56:15.236344 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[PCMA:8:8000:20]
2010-10-18 15:56:15.236344 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[GSM:3:8000:20]
2010-10-18 15:56:15.236344 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [PCMU:0:8000:20]/[G7221:115:32000:20]
2010-10-18 15:56:15.236344 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [PCMU:0:8000:20]/[G7221:107:16000:20]
2010-10-18 15:56:15.236344 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [PCMU:0:8000:20]/[G722:9:8000:20]
2010-10-18 15:56:15.236344 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20]
2010-10-18 15:56:15.237437 [DEBUG] sofia_glue.c:2376 Set Codec sofia/internal/1013 at 60.248.175.38 PCMU/8000 20 ms 160 samples
2010-10-18 15:56:15.237437 [DEBUG] sofia_glue.c:3610 Set 2833 dtmf send/recv payload to 101
2010-10-18 15:56:15.237437 [DEBUG] sofia.c:4357 (sofia/internal/1013 at 60.248.175.38) State Change CS_NEW -> CS_INIT
2010-10-18 15:56:15.237437 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_INIT
2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1013 at 60.248.175.38) State INIT
2010-10-18 15:56:15.237437 [DEBUG] mod_sofia.c:83 sofia/internal/1013 at 60.248.175.38 SOFIA INIT
2010-10-18 15:56:15.237437 [DEBUG] mod_sofia.c:117 (sofia/internal/1013 at 60.248.175.38) State Change CS_INIT -> CS_ROUTING
2010-10-18 15:56:15.237437 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1013 at 60.248.175.38) State INIT going to sleep
2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_ROUTING
2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38) State ROUTING
2010-10-18 15:56:15.237437 [DEBUG] mod_sofia.c:140 sofia/internal/1013 at 60.248.175.38 SOFIA ROUTING
2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1013 at 60.248.175.38 Standard ROUTING
2010-10-18 15:56:15.237437 [INFO] mod_dialplan_xml.c:331 Processing 1013->4001 in context public
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->unloop] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->outside_call] continue=true
Dialplan: sofia/internal/1013 at 60.248.175.38 Absolute Condition [outside_call]
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(outside_call=true)
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->call_debug] continue=true
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->public_extensions] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [public_extensions] destination_number(4001) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->Calls from 37] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [Calls from 37] destination_number(4001) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->public_did] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [public_did] destination_number(4001) =~ /^(5551212)$/ break=on-false
2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1013 at 60.248.175.38) State Change CS_ROUTING -> CS_EXECUTE
2010-10-18 15:56:15.237437 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38) State ROUTING going to sleep
2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_EXECUTE
2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38) State EXECUTE
2010-10-18 15:56:15.237437 [DEBUG] mod_sofia.c:233 sofia/internal/1013 at 60.248.175.38 SOFIA EXECUTE
2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1013 at 60.248.175.38 Standard EXECUTE
EXECUTE sofia/internal/1013 at 60.248.175.38 set(outside_call=true)
2010-10-18 15:56:15.237437 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [outside_call]=[true]
2010-10-18 15:56:15.237437 [NOTICE] switch_core_state_machine.c:185 sofia/internal/1013 at 60.248.175.38 has executed the last dialplan instruction, hanging up.
2010-10-18 15:56:15.237437 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/1013 at 60.248.175.38 [CS_EXECUTE] [NORMAL_CLEARING]
2010-10-18 15:56:15.237437 [DEBUG] switch_channel.c:2145 Send signal sofia/internal/1013 at 60.248.175.38 [KILL]
2010-10-18 15:56:15.237437 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38) State EXECUTE going to sleep
2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_HANGUP
2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/1013 at 60.248.175.38) State HANGUP
2010-10-18 15:56:15.237437 [DEBUG] mod_sofia.c:441 Channel sofia/internal/1013 at 60.248.175.38 hanging up, cause: NORMAL_CLEARING
2010-10-18 15:56:15.263831 [DEBUG] mod_sofia.c:503 Responding to INVITE with: 480
2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1013 at 60.248.175.38 Standard HANGUP, cause: NORMAL_CLEARING
2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/1013 at 60.248.175.38) State HANGUP going to sleep
2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/1013 at 60.248.175.38) State Change CS_HANGUP -> CS_REPORTING
2010-10-18 15:56:15.263831 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_REPORTING
2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:589 (sofia/internal/1013 at 60.248.175.38) State REPORTING
2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1013 at 60.248.175.38 Standard REPORTING, cause: NORMAL_CLEARING
2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:589 (sofia/internal/1013 at 60.248.175.38) State REPORTING going to sleep
2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/1013 at 60.248.175.38) State Change CS_REPORTING -> CS_DESTROY
2010-10-18 15:56:15.263831 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 15:56:15.263831 [DEBUG] switch_core_session.c:1170 Session 4 (sofia/internal/1013 at 60.248.175.38) Locked, Waiting on external entities
2010-10-18 15:56:15.263831 [NOTICE] switch_core_session.c:1188 Session 4 (sofia/internal/1013 at 60.248.175.38) Ended
2010-10-18 15:56:15.263831 [NOTICE] switch_core_session.c:1190 Close Channel sofia/internal/1013 at 60.248.175.38 [CS_DESTROY]
2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:428 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_DESTROY
2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:438 (sofia/internal/1013 at 60.248.175.38) State DESTROY
2010-10-18 15:56:15.263831 [DEBUG] mod_sofia.c:350 sofia/internal/1013 at 60.248.175.38 SOFIA DESTROY
2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1013 at 60.248.175.38 Standard DESTROY
2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:438 (sofia/internal/1013 at 60.248.175.38) State DESTROY going to sleep

and I use 4001 at 64.248.175.37 calls to 1013 at 64.248.175.38
log from 37 (that's all)
 
2010-10-18 15:59:04.730021 [DEBUG] sofia.c:5849 IP 60.248.175.42 Approved by acl "domains[]". Access Granted.
2010-10-18 15:59:04.730021 [NOTICE] switch_channel.c:675 New Channel sofia/internal/4001 at 60.248.175.37 [0a71af92-1564-49e1-9a8e-9f5432b1653b]
2010-10-18 15:59:04.735619 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/4001 at 60.248.175.37) Running State Change CS_NEW
2010-10-18 15:59:04.735619 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/4001 at 60.248.175.37) State NEW
2010-10-18 15:59:04.877644 [DEBUG] sofia.c:4200 Channel sofia/internal/4001 at 60.248.175.37 entering state [received][100]
2010-10-18 15:59:04.877644 [DEBUG] sofia.c:4211 Remote SDP:
v=0
o=- 6 2 IN IP4 60.248.175.42
s=CounterPath X-Lite 3.0
c=IN IP4 60.248.175.42
t=0 0
m=audio 51406 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 3 : s7WKw7mB W35tlaoK 60.248.175.42 51406
a=alt:2 2 : SfvGIlEF m2IrZtjW 192.168.24.1 51406
a=alt:3 1 : fQ/a57gE kxcr2NP4 192.168.137.1 51406
2010-10-18 15:59:04.880340 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[G7221:115:32000:20]
2010-10-18 15:59:04.881429 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[G7221:107:16000:20]
2010-10-18 15:59:04.881429 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[G722:9:8000:20]
2010-10-18 15:59:04.881429 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[PCMU:0:8000:20]
2010-10-18 15:59:04.883611 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[PCMA:8:8000:20]
2010-10-18 15:59:04.883611 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[GSM:3:8000:20]
2010-10-18 15:59:04.884891 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [PCMU:0:8000:20]/[G7221:115:32000:20]
2010-10-18 15:59:04.886125 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [PCMU:0:8000:20]/[G7221:107:16000:20]
2010-10-18 15:59:04.886125 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [PCMU:0:8000:20]/[G722:9:8000:20]
2010-10-18 15:59:04.886125 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20]
2010-10-18 15:59:04.887562 [DEBUG] sofia_glue.c:2376 Set Codec sofia/internal/4001 at 60.248.175.37 PCMU/8000 20 ms 160 samples
2010-10-18 15:59:04.888900 [DEBUG] sofia_glue.c:3610 Set 2833 dtmf send/recv payload to 101
2010-10-18 15:59:04.888900 [DEBUG] sofia.c:4357 (sofia/internal/4001 at 60.248.175.37) State Change CS_NEW -> CS_INIT
2010-10-18 15:59:04.890190 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/4001 at 60.248.175.37 [BREAK]
2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/4001 at 60.248.175.37) Running State Change CS_INIT
2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/4001 at 60.248.175.37) State INIT
2010-10-18 15:59:04.891645 [DEBUG] mod_sofia.c:83 sofia/internal/4001 at 60.248.175.37 SOFIA INIT
2010-10-18 15:59:04.891645 [DEBUG] mod_sofia.c:117 (sofia/internal/4001 at 60.248.175.37) State Change CS_INIT -> CS_ROUTING
2010-10-18 15:59:04.891645 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/4001 at 60.248.175.37 [BREAK]
2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/4001 at 60.248.175.37) State INIT going to sleep
2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/4001 at 60.248.175.37) Running State Change CS_ROUTING
2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/4001 at 60.248.175.37) State ROUTING
2010-10-18 15:59:04.891645 [DEBUG] mod_sofia.c:140 sofia/internal/4001 at 60.248.175.37 SOFIA ROUTING
2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:77 sofia/internal/4001 at 60.248.175.37 Standard ROUTING
2010-10-18 15:59:04.891645 [INFO] mod_dialplan_xml.c:331 Processing 4001->1013 in context public
Dialplan: sofia/internal/4001 at 60.248.175.37 parsing [public->unloop] continue=false
Dialplan: sofia/internal/4001 at 60.248.175.37 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/4001 at 60.248.175.37 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/4001 at 60.248.175.37 parsing [public->outside_call] continue=true
Dialplan: sofia/internal/4001 at 60.248.175.37 Absolute Condition [outside_call]
Dialplan: sofia/internal/4001 at 60.248.175.37 Action set(outside_call=true)
Dialplan: sofia/internal/4001 at 60.248.175.37 parsing [public->call_debug] continue=true
Dialplan: sofia/internal/4001 at 60.248.175.37 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/4001 at 60.248.175.37 parsing [public->public_extensions] continue=false
Dialplan: sofia/internal/4001 at 60.248.175.37 Regex (FAIL) [public_extensions] destination_number(1013) =~ /^(40[01][0-9])$/ break=on-false
Dialplan: sofia/internal/4001 at 60.248.175.37 parsing [public->Calls from 38] continue=false
Dialplan: sofia/internal/4001 at 60.248.175.37 Regex (FAIL) [Calls from 38] destination_number(1013) =~ /^(40[01][0-9])$/ break=on-false
Dialplan: sofia/internal/4001 at 60.248.175.37 parsing [public->public_did] continue=false
Dialplan: sofia/internal/4001 at 60.248.175.37 Regex (FAIL) [public_did] destination_number(1013) =~ /^(5551212)$/ break=on-false
2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/4001 at 60.248.175.37) State Change CS_ROUTING -> CS_EXECUTE
2010-10-18 15:59:04.891645 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/4001 at 60.248.175.37 [BREAK]
2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/4001 at 60.248.175.37) State ROUTING going to sleep
2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/4001 at 60.248.175.37) Running State Change CS_EXECUTE
2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/4001 at 60.248.175.37) State EXECUTE
2010-10-18 15:59:04.891645 [DEBUG] mod_sofia.c:233 sofia/internal/4001 at 60.248.175.37 SOFIA EXECUTE
2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:157 sofia/internal/4001 at 60.248.175.37 Standard EXECUTE
EXECUTE sofia/internal/4001 at 60.248.175.37 set(outside_call=true)
2010-10-18 15:59:04.891645 [DEBUG] mod_dptools.c:817 sofia/internal/4001 at 60.248.175.37 SET [outside_call]=[true]
2010-10-18 15:59:04.891645 [NOTICE] switch_core_state_machine.c:185 sofia/internal/4001 at 60.248.175.37 has executed the last dialplan instruction, hanging up.
2010-10-18 15:59:04.891645 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/4001 at 60.248.175.37 [CS_EXECUTE] [NORMAL_CLEARING]
2010-10-18 15:59:04.891645 [DEBUG] switch_channel.c:2145 Send signal sofia/internal/4001 at 60.248.175.37 [KILL]
2010-10-18 15:59:04.891645 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/4001 at 60.248.175.37 [BREAK]
2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/4001 at 60.248.175.37) State EXECUTE going to sleep
2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/4001 at 60.248.175.37) Running State Change CS_HANGUP
2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/4001 at 60.248.175.37) State HANGUP
2010-10-18 15:59:04.891645 [DEBUG] mod_sofia.c:441 Channel sofia/internal/4001 at 60.248.175.37 hanging up, cause: NORMAL_CLEARING
2010-10-18 15:59:04.928639 [DEBUG] mod_sofia.c:503 Responding to INVITE with: 480
2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:46 sofia/internal/4001 at 60.248.175.37 Standard HANGUP, cause: NORMAL_CLEARING
2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/4001 at 60.248.175.37) State HANGUP going to sleep
2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/4001 at 60.248.175.37) State Change CS_HANGUP -> CS_REPORTING
2010-10-18 15:59:04.928639 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/4001 at 60.248.175.37 [BREAK]
2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/4001 at 60.248.175.37) Running State Change CS_REPORTING
2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:589 (sofia/internal/4001 at 60.248.175.37) State REPORTING
2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:53 sofia/internal/4001 at 60.248.175.37 Standard REPORTING, cause: NORMAL_CLEARING
2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:589 (sofia/internal/4001 at 60.248.175.37) State REPORTING going to sleep
2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/4001 at 60.248.175.37) State Change CS_REPORTING -> CS_DESTROY
2010-10-18 15:59:04.928639 [DEBUG] switch_core_session.c:1022 Send signal sofia/internal/4001 at 60.248.175.37 [BREAK]
2010-10-18 15:59:04.928639 [DEBUG] switch_core_session.c:1170 Session 3 (sofia/internal/4001 at 60.248.175.37) Locked, Waiting on external entities
2010-10-18 15:59:04.928639 [NOTICE] switch_core_session.c:1188 Session 3 (sofia/internal/4001 at 60.248.175.37) Ended
2010-10-18 15:59:04.928639 [NOTICE] switch_core_session.c:1190 Close Channel sofia/internal/4001 at 60.248.175.37 [CS_DESTROY]
2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:428 (sofia/internal/4001 at 60.248.175.37) Running State Change CS_DESTROY
2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:438 (sofia/internal/4001 at 60.248.175.37) State DESTROY
2010-10-18 15:59:04.928639 [DEBUG] mod_sofia.c:350 sofia/internal/4001 at 60.248.175.37 SOFIA DESTROY
2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:60 sofia/internal/4001 at 60.248.175.37 Standard DESTROY
2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:438 (sofia/internal/4001 at 60.248.175.37) State DESTROY going to sleep
 
that's all my information I really hope that I can solve this problem but I have no idea about this function 
 
Thank you for your attsnsion
 
Best Regards
 
Gary 


From: david.ponzone at ipeva.fr
To: freeswitch-users at lists.freeswitch.org
Date: Mon, 18 Oct 2010 09:25:25 +0200
Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite

You failed to tell us which server this log comes from.


In order to get a result, I think you need to try to understand what the logs mean and you need to understand that to help you, we need detailed logs and relevant config files.
>From the beginning of this thread, you failed to provide both at the same time.


Also, that's not a basic config.
Have you tried before to do a single server config ?
I would start there, so that you can understand how FreeSWITCH works, why the logs are meaning ful and how to read them, which dialplans are parsed and when.
If you don't do that, you're going to waste people's time because you want to avoid some required (and important) learning steps.


Thank you






David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34


Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr  -   www.ipeva-studio.com



Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 18/10/2010 ? 02:37, ? Chiang ?? Chih-Chung Wybie a ?crit :

I tried to set different expression but it still can't work
 
debug log below
 
2010-10-18 08:34:19.462617 [DEBUG] sofia.c:5849 IP 60.248.175.43 Approved by acl "domains[]". Access Granted.
2010-10-18 08:34:19.462617 [NOTICE] switch_channel.c:675 New Channel sofia/internal/1013 at 60.248.175.38[1f4277c1-9ded-49c5-b63f-a312bead9d5f]
2010-10-18 08:34:19.464814 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_NEW
2010-10-18 08:34:19.464814 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1013 at 60.248.175.38) State NEW
2010-10-18 08:34:19.475647 [DEBUG] sofia.c:4200 Channel sofia/internal/1013 at 60.248.175.38 entering state [received][100]
2010-10-18 08:34:19.475647 [DEBUG] sofia.c:4211 Remote SDP:
v=0
o=- 0 2 IN IP4 60.248.175.43
s=CounterPath X-Lite 3.0
c=IN IP4 60.248.175.43
t=0 0
m=audio 32692 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 3 : d7A5MyJF tN44o7uR 60.248.175.43 32692
a=alt:2 2 : cCI83KQt qU62M7Ia 192.168.182.1 32692
a=alt:3 1 : qtl4Jt3z ug+kYltk 192.168.60.1 32692
2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[G7221:115:32000:20]
2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[G7221:107:16000:20]
2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[G722:9:8000:20]
2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[PCMU:0:8000:20]
2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[PCMA:8:8000:20]
2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [BV32:107:16000:20]/[GSM:3:8000:20]
2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [PCMU:0:8000:20]/[G7221:115:32000:20]
2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [PCMU:0:8000:20]/[G7221:107:16000:20]
2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [PCMU:0:8000:20]/[G722:9:8000:20]
2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3671 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20]
2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:2376 Set Codec sofia/internal/1013 at 60.248.175.38PCMU/8000 20 ms 160 samples
2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3610 Set 2833 dtmf send/recv payload to 101
2010-10-18 08:34:19.476688 [DEBUG] sofia.c:4357 (sofia/internal/1013 at 60.248.175.38) State Change CS_NEW -> CS_INIT
2010-10-18 08:34:19.476688 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_INIT
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1013 at 60.248.175.38) State INIT
2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:83 sofia/internal/1013 at 60.248.175.38 SOFIA INIT
2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:117 (sofia/internal/1013 at 60.248.175.38) State Change CS_INIT -> CS_ROUTING
2010-10-18 08:34:19.476688 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1013 at 60.248.175.38) State INIT going to sleep
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_ROUTING
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38) State ROUTING
2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:140 sofia/internal/1013 at 60.248.175.38 SOFIA ROUTING
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1013 at 60.248.175.38Standard ROUTING
2010-10-18 08:34:19.476688 [INFO] mod_dialplan_xml.c:331 Processing 1013->4001 in context public
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->unloop] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->outside_call] continue=true
Dialplan: sofia/internal/1013 at 60.248.175.38 Absolute Condition [outside_call]
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(outside_call=true)
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->call_debug] continue=true
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->public_extensions] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [public_extensions] destination_number(4001) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->Calls from 37] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [Calls from 37] destination_number(4001) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->public_did] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [public_did] destination_number(4001) =~ /^(5551212)$/ break=on-false
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1013 at 60.248.175.38) State Change CS_ROUTING -> CS_EXECUTE
2010-10-18 08:34:19.476688 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38) State ROUTING going to sleep
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_EXECUTE
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38) State EXECUTE
2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:233 sofia/internal/1013 at 60.248.175.38 SOFIA EXECUTE
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1013 at 60.248.175.38Standard EXECUTE
EXECUTE sofia/internal/1013 at 60.248.175.38 set(outside_call=true)
2010-10-18 08:34:19.476688 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [outside_call]=[true]
2010-10-18 08:34:19.476688 [NOTICE] switch_core_state_machine.c:185 sofia/internal/1013 at 60.248.175.38has executed the last dialplan instruction, hanging up.
2010-10-18 08:34:19.476688 [NOTICE] switch_core_state_machine.c:187 Hangupsofia/internal/1013 at 60.248.175.38 [CS_EXECUTE] [NORMAL_CLEARING]
2010-10-18 08:34:19.476688 [DEBUG] switch_channel.c:2145 Send signal sofia/internal/1013 at 60.248.175.38[KILL]
2010-10-18 08:34:19.476688 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38) State EXECUTE going to sleep
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_HANGUP
2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/1013 at 60.248.175.38) State HANGUP
2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:441 Channel sofia/internal/1013 at 60.248.175.38 hanging up, cause: NORMAL_CLEARING
2010-10-18 08:34:19.494277 [DEBUG] mod_sofia.c:503 Responding to INVITE with: 480
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1013 at 60.248.175.38Standard HANGUP, cause: NORMAL_CLEARING
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/1013 at 60.248.175.38) State HANGUP going to sleep
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/1013 at 60.248.175.38) State Change CS_HANGUP -> CS_REPORTING
2010-10-18 08:34:19.494277 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_REPORTING
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:589 (sofia/internal/1013 at 60.248.175.38) State REPORTING
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1013 at 60.248.175.38Standard REPORTING, cause: NORMAL_CLEARING
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:589 (sofia/internal/1013 at 60.248.175.38) State REPORTING going to sleep
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/1013 at 60.248.175.38) State Change CS_REPORTING -> CS_DESTROY
2010-10-18 08:34:19.494277 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-18 08:34:19.494277 [DEBUG] switch_core_session.c:1170 Session 61 (sofia/internal/1013 at 60.248.175.38) Locked, Waiting on external entities
2010-10-18 08:34:19.494277 [NOTICE] switch_core_session.c:1188 Session 61 (sofia/internal/1013 at 60.248.175.38) Ended
2010-10-18 08:34:19.494277 [NOTICE] switch_core_session.c:1190 Close Channelsofia/internal/1013 at 60.248.175.38 [CS_DESTROY]
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:428 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_DESTROY
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:438 (sofia/internal/1013 at 60.248.175.38) State DESTROY
2010-10-18 08:34:19.494277 [DEBUG] mod_sofia.c:350 sofia/internal/1013 at 60.248.175.38 SOFIA DESTROY
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1013 at 60.248.175.38Standard DESTROY
2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:438 (sofia/internal/1013 at 60.248.175.38) State DESTROY going to sleep

Thank you for your attension
 
Best Regards
 
Gary
 


From: david.ponzone at ipeva.fr
To: freeswitch-users at lists.freeswitch.org
Date: Fri, 15 Oct 2010 11:16:41 +0200
Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite

You need to understand that if you dial a number that you want to be routed to your 60.248.175.37, you need NOT TO HAVE this number intercepted by the default context on 60.248.175.38.
In the log you sent, the call is staying local to 60.248.175.38.
It never reaches 60.248.175.37.


You need to have different numbering plan (you can't have a 4001 on both machines and hope to reach both with the same number).
Or you keep the same numbering plan, but you use a prefix in the dialplan to reach the other host.
For instance you would dial 994001, but your dialplan will match it with a regexp like ^99(4\d{3})$ and bridge $1 (so only 4001) to the other FS.





David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34


Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr  -   www.ipeva-studio.com



Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 15/10/2010 ? 10:52, ? Chiang ?? Chih-Chung Wybie a ?crit :

Hello this is my debug log
 
I use 1013 at 60.248.175.38 call to 4001 at 60.248.175.37
 
EXECUTE sofia/internal/1013 at 60.248.175.38 transfer(4001 XML default)
2010-10-15 16:49:30.666644 [DEBUG] switch_ivr.c:1444 (sofia/internal/1013 at 60.248.175.38) State Change CS_EXECUTE -> CS_ROUTING
2010-10-15 16:49:30.666644 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:30.666644 [DEBUG] switch_core_session.c:642 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:30.666644 [NOTICE] switch_ivr.c:1450 Transfer sofia/internal/1013 at 60.248.175.38 to XML[4001 at default]
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38) State EXECUTE going to sleep
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_ROUTING
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38) State ROUTING
2010-10-15 16:49:30.666644 [DEBUG] mod_sofia.c:140 sofia/internal/1013 at 60.248.175.38 SOFIA ROUTING
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1013 at 60.248.175.38Standard ROUTING
2010-10-15 16:49:30.666644 [INFO] mod_dialplan_xml.c:331 Processing 1013->4001 in context default
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->unloop] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->tod_example] continue=true
Dialplan: sofia/internal/1013 at 60.248.175.38 Date/Time Match (PASS) [tod_example] break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(open=true)
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->holiday_example] continue=true
Dialplan: sofia/internal/1013 at 60.248.175.38 Date/Time Match (FAIL) [holiday_example] break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->global-intercept] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [global-intercept] destination_number(4001) =~ /^886$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->group-intercept] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [group-intercept] destination_number(4001) =~ /^\*8$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->intercept-ext] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [intercept-ext] destination_number(4001) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->redial] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [redial] destination_number(4001) =~ /^(redial|870)$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->global] continue=true
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: sofia/internal/1013 at 60.248.175.38 Absolute Condition [global]
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-last_dial/global/${uuid})
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->snom-demo-2] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [snom-demo-2] destination_number(4001) =~ /^9001$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->snom-demo-1] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [snom-demo-1] destination_number(4001) =~ /^9000$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->eavesdrop] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [eavesdrop] destination_number(4001) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->eavesdrop] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [eavesdrop] destination_number(4001) =~ /^779$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->call_return] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call_return] destination_number(4001) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->del-group] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [del-group] destination_number(4001) =~ /^80(\d{2})$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->add-group] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [add-group] destination_number(4001) =~ /^81(\d{2})$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->call-group-simo] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call-group-simo] destination_number(4001) =~ /^82(\d{2})$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->call-group-order] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call-group-order] destination_number(4001) =~ /^83(\d{2})$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->extension-intercom] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [extension-intercom] destination_number(4001) =~ /^8(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->Local_Extension] continue=false
Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (PASS) [Local_Extension] destination_number(4001) =~ /^([0-4]0[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_COPYRIGHT=(c) 2010)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_SOFTWARE=FreeSwitch)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_ARTIST=FreeSwitch)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_COMMENT=FreeSwitch)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_DATE=${strftime(%Y-%m-%d %H:%M)})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_STEREO=true)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(dialed_extension=4001)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action export(dialed_extension=4001)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set_user(${dialed_extension}@${domain_name})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action limit(60.248.175.38 ${dialed_extension} ${max_calls} ${fail_over})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action bind_meta_app(1 b s execute_extension::dx XML features)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action bind_meta_app(3 b s execute_extension::cf XML features)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(ringback=${us-ring})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(transfer_ringback=local_stream://moh)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(call_timeout=30)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(hangup_after_bridge=true)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(continue_on_fail=true)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action bridge(user/${dialed_extension}@${domain_name})
Dialplan: sofia/internal/1013 at 60.248.175.38 Action answer()
Dialplan: sofia/internal/1013 at 60.248.175.38 Action sleep(1000)
Dialplan: sofia/internal/1013 at 60.248.175.38 Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension})
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1013 at 60.248.175.38) State Change CS_ROUTING -> CS_EXECUTE
2010-10-15 16:49:30.666644 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38) State ROUTING going to sleep
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38) Running State Change CS_EXECUTE
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38) State EXECUTE
2010-10-15 16:49:30.666644 [DEBUG] mod_sofia.c:233 sofia/internal/1013 at 60.248.175.38 SOFIA EXECUTE
2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1013 at 60.248.175.38Standard EXECUTE
EXECUTE sofia/internal/1013 at 60.248.175.38 set(open=true)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [open]=[true]
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-spymap/1013/fe66f80a-992b-43f7-b130-47f0c87f4c15)
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-last_dial/1013/4001)
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-last_dial/global/fe66f80a-992b-43f7-b130-47f0c87f4c15)
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_COPYRIGHT=(c) 2010)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_COPYRIGHT]=[(c) 2010]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_SOFTWARE=FreeSwitch)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_SOFTWARE]=[FreeSwitch]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_ARTIST=FreeSwitch)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_ARTIST]=[FreeSwitch]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_COMMENT=FreeSwitch)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_COMMENT]=[FreeSwitch]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_DATE=2010-10-15 16:49)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_DATE]=[2010-10-15 16:49]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_STEREO=true)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [RECORD_STEREO]=[true]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(dialed_extension=4001)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [dialed_extension]=[4001]
EXECUTE sofia/internal/1013 at 60.248.175.38 export(dialed_extension=4001)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:901 EXPORT [dialed_extension]=[4001]
EXECUTE sofia/internal/1013 at 60.248.175.38 set_user(4001 at 60.248.175.38)
EXECUTE sofia/internal/1013 at 60.248.175.38 limit(60.248.175.38 4001  )
2010-10-15 16:49:30.666644 [WARNING] mod_limit.c:779 USAGE: limit <realm> <id> [<max> [number  [dialplan [context]]]]
EXECUTE sofia/internal/1013 at 60.248.175.38 bind_meta_app(1 b s execute_extension::dx XML features)
2010-10-15 16:49:30.666644 [INFO] switch_ivr_async.c:2372 Bound B-Leg: 1 execute_extension::dx XML features
EXECUTE sofia/internal/1013 at 60.248.175.38 bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/1013.2010-10-15-16-49-30.wav)
2010-10-15 16:49:30.666644 [INFO] switch_ivr_async.c:2372 Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/1013.2010-10-15-16-49-30.wav
EXECUTE sofia/internal/1013 at 60.248.175.38 bind_meta_app(3 b s execute_extension::cf XML features)
2010-10-15 16:49:30.666644 [INFO] switch_ivr_async.c:2372 Bound B-Leg: 3 execute_extension::cf XML features
EXECUTE sofia/internal/1013 at 60.248.175.38 set(ringback=%(2000,4000,440.0,480.0))
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [ringback]=[%(2000,4000,440.0,480.0)]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(transfer_ringback=local_stream://moh)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [transfer_ringback]=[local_stream://moh]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(call_timeout=30)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [call_timeout]=[30]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(hangup_after_bridge=true)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1013 at 60.248.175.38 set(continue_on_fail=true)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [continue_on_fail]=[true]
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-call_return/4001/1013)
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-last_dial_ext/4001/fe66f80a-992b-43f7-b130-47f0c87f4c15)
EXECUTE sofia/internal/1013 at 60.248.175.38 set(called_party_callgroup=techsupport)
2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 SET [called_party_callgroup]=[techsupport]
EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38-last_dial/techsupport/fe66f80a-992b-43f7-b130-47f0c87f4c15)
EXECUTE sofia/internal/1013 at 60.248.175.38 bridge(user/4001 at 60.248.175.38)
2010-10-15 16:49:30.666644 [DEBUG] switch_ivr_originate.c:1946 variable string 0 = [presence_id=4001 at 60.248.175.38]
2010-10-15 16:49:30.666644 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
2010-10-15 16:49:30.666644 [DEBUG] switch_ivr_originate.c:3308 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2010-10-15 16:49:30.666644 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
2010-10-15 16:49:30.666644 [DEBUG] switch_ivr_originate.c:3308 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2010-10-15 16:49:30.666644 [INFO] mod_dptools.c:2356 Originate Failed.  Cause: USER_NOT_REGISTERED
EXECUTE sofia/internal/1013 at 60.248.175.38 answer()
2010-10-15 16:49:30.694130 [DEBUG] sofia_glue.c:2616 AUDIO RTP [sofia/internal/1013 at 60.248.175.38] 60.248.175.38 port 25606 -> 60.248.175.43 port 35070 codec: 0 ms: 20
2010-10-15 16:49:30.694130 [DEBUG] switch_rtp.c:1343 Starting timer [soft] 160 bytes per 20ms
2010-10-15 16:49:30.695467 [DEBUG] sofia_glue.c:2822 Set 2833 dtmf send payload to 101
2010-10-15 16:49:30.695467 [DEBUG] sofia_glue.c:2827 Set 2833 dtmf receive payload to 101
2010-10-15 16:49:30.695467 [DEBUG] mod_sofia.c:663 Local SDP sofia/internal/1013 at 60.248.175.38:
v=0
o=FreeSWITCH 1287106964 1287106965 IN IP4 60.248.175.38
s=FreeSWITCH
c=IN IP4 60.248.175.38
t=0 0
m=audio 25606 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
2010-10-15 16:49:30.695467 [DEBUG] switch_core_session.c:642 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:30.695467 [NOTICE] mod_dptools.c:720 Channel [sofia/internal/1013 at 60.248.175.38] has been answered
EXECUTE sofia/internal/1013 at 60.248.175.38 sleep(1000)
2010-10-15 16:49:30.698172 [DEBUG] sofia.c:4200 Channel sofia/internal/1013 at 60.248.175.38 entering state [completed][200]
2010-10-15 16:49:30.745378 [DEBUG] switch_rtp.c:2443 Correct ip/port confirmed.
2010-10-15 16:49:30.807388 [DEBUG] sofia.c:4200 Channel sofia/internal/1013 at 60.248.175.38 entering state [ready][200]
EXECUTE sofia/internal/1013 at 60.248.175.38 bridge(loopback/app=voicemail:default 60.248.175.38 4001)
2010-10-15 16:49:31.704837 [NOTICE] switch_channel.c:675 New Channel loopback/app=voicemail:default 60.248.175.38 4001-a [23f3d5de-f985-44bd-8b1a-118b98bb5eae]
2010-10-15 16:49:31.704837 [DEBUG] mod_loopback.c:127 loopback/app=voicemail:default 60.248.175.38 4001-a setup codec PCMU/8000/20
2010-10-15 16:49:31.704837 [NOTICE] switch_channel.c:673 Rename Channel loopback/app=voicemail:default 60.248.175.38 4001-a->loopback/voicemail-a [23f3d5de-f985-44bd-8b1a-118b98bb5eae]
2010-10-15 16:49:31.704837 [DEBUG] mod_loopback.c:911 (loopback/voicemail-a) State Change CS_NEW -> CS_INIT
2010-10-15 16:49:31.704837 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.704837 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.706218 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-a) Running State Change CS_INIT
2010-10-15 16:49:31.706218 [DEBUG] switch_core_state_machine.c:338 (loopback/voicemail-a) State INIT
2010-10-15 16:49:31.706218 [NOTICE] switch_channel.c:675 New Channel loopback/voicemail-b [1abc53a2-df93-40db-96ab-547f9ddd1b1d]
2010-10-15 16:49:31.706218 [DEBUG] mod_loopback.c:127 loopback/voicemail-b setup codec PCMU/8000/20
2010-10-15 16:49:31.706218 [DEBUG] mod_loopback.c:240 (loopback/voicemail-b) State Change CS_NEW -> CS_INIT
2010-10-15 16:49:31.706218 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-b [BREAK]
2010-10-15 16:49:31.706218 [DEBUG] mod_loopback.c:452 loopback/voicemail-b CHANNEL KILL
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-b) Running State Change CS_INIT
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:338 (loopback/voicemail-b) State INIT
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:281 (loopback/voicemail-b) State Change CS_INIT -> CS_ROUTING
2010-10-15 16:49:31.707132 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-b [BREAK]
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:452 loopback/voicemail-b CHANNEL KILL
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:338 (loopback/voicemail-b) State INIT going to sleep
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-b) Running State Change CS_ROUTING
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:341 (loopback/voicemail-b) State ROUTING
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:315 loopback/voicemail-b CHANNEL ROUTING
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:326 (loopback/voicemail-b) State Change CS_ROUTING -> CS_EXECUTE
2010-10-15 16:49:31.707132 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-b [BREAK]
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:452 loopback/voicemail-b CHANNEL KILL
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:341 (loopback/voicemail-b) State ROUTING going to sleep
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-b) Running State Change CS_EXECUTE
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:348 (loopback/voicemail-b) State EXECUTE
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:346 loopback/voicemail-b CHANNEL EXECUTE
2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:157 loopback/voicemail-b Standard EXECUTE
EXECUTE loopback/voicemail-b pre_answer()
2010-10-15 16:49:31.707132 [NOTICE] mod_loopback.c:716 Pre-Answer loopback/voicemail-a!
2010-10-15 16:49:31.707132 [DEBUG] switch_channel.c:2254 Send signal sofia/internal/1013 at 60.248.175.38[BREAK]
2010-10-15 16:49:31.707132 [DEBUG] switch_core_session.c:642 Send signal loopback/voicemail-b [BREAK]
2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:452 loopback/voicemail-b CHANNEL KILL
2010-10-15 16:49:31.707132 [NOTICE] mod_dptools.c:746 Pre-Answer loopback/voicemail-b!
2010-10-15 16:49:31.707132 [DEBUG] switch_channel.c:2254 Send signal sofia/internal/1013 at 60.248.175.38[BREAK]
EXECUTE loopback/voicemail-b voicemail(default 60.248.175.38 4001)
2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:281 (loopback/voicemail-a) State Change CS_INIT -> CS_ROUTING
2010-10-15 16:49:31.708787 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:338 (loopback/voicemail-a) State INIT going to sleep
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-a) Running State Change CS_ROUTING
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:341 (loopback/voicemail-a) State ROUTING
2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:315 loopback/voicemail-a CHANNEL ROUTING
2010-10-15 16:49:31.708787 [DEBUG] switch_ivr_originate.c:66 (loopback/voicemail-a) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2010-10-15 16:49:31.708787 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:341 (loopback/voicemail-a) State ROUTING going to sleep
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-a) Running State Change CS_CONSUME_MEDIA
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:360 (loopback/voicemail-a) State CONSUME_MEDIA
2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:512 CHANNEL CONSUME_MEDIA
2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:360 (loopback/voicemail-a) State CONSUME_MEDIA going to sleep
2010-10-15 16:49:31.710698 [DEBUG] switch_ivr_originate.c:3228 Originate Resulted in Success: [loopback/voicemail-a]
2010-10-15 16:49:31.710698 [DEBUG] switch_core_session.c:642 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.710698 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.710698 [DEBUG] switch_core_session.c:642 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:31.710698 [DEBUG] switch_ivr_bridge.c:1182 (loopback/voicemail-a) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2010-10-15 16:49:31.710698 [DEBUG] switch_core_session.c:1022 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.710698 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.710698 [DEBUG] switch_core_state_machine.c:314 (loopback/voicemail-a) Running State Change CS_EXCHANGE_MEDIA
2010-10-15 16:49:31.710698 [DEBUG] switch_core_state_machine.c:351 (loopback/voicemail-a) State EXCHANGE_MEDIA
2010-10-15 16:49:31.710698 [DEBUG] mod_loopback.c:474 CHANNEL LOOPBACK
2010-10-15 16:49:31.712661 [DEBUG] switch_core_session.c:703 Send signal loopback/voicemail-a [BREAK]
2010-10-15 16:49:31.712661 [DEBUG] mod_loopback.c:452 loopback/voicemail-a CHANNEL KILL
2010-10-15 16:49:31.712661 [DEBUG] switch_core_session.c:703 Send signalsofia/internal/1013 at 60.248.175.38 [BREAK]
2010-10-15 16:49:31.824953 [DEBUG] switch_ivr_play_say.c:63 No language specified - Using [en]
2010-10-15 16:49:31.824953 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-person.wav] (en:en)
2010-10-15 16:49:31.824953 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16 at 8000hz 1 channels 20ms
2010-10-15 16:49:33.185370 [DEBUG] switch_ivr_play_say.c:1444 done playing file
2010-10-15 16:49:33.305429 [DEBUG] switch_ivr_play_say.c:244 Handle say:[4001] (en:en)
2010-10-15 16:49:33.305429 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16 at 8000hz 1 channels 20ms
2010-10-15 16:49:33.744961 [DEBUG] switch_ivr_play_say.c:1444 done playing file
2010-10-15 16:49:33.744961 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16 at 8000hz 1 channels 20ms

 


From: david.ponzone at ipeva.fr
To: freeswitch-users at lists.freeswitch.org
Date: Thu, 14 Oct 2010 12:20:56 +0200
Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite

Don't be shy on the logs, send the full ones!


in fs_cli:
fsctl loglevel debug
/log 7


and then make a call





David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34


Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr  -   www.ipeva-studio.com



Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 14/10/2010 ? 11:53, ? Chiang ?? Chih-Chung Wybie a ?crit :

I tried use 1013 at 38 calls to 1019 at 37
 
show log below
 
2010-10-14 17:50:41.249876 [NOTICE] switch_channel.c:675 New Channel sofia/internal/1013 at 60.248.175.38[875fdeef-b94e-41fb-a621-ea005bbaedbd]
2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing 1013->1019 in context public
2010-10-14 17:50:41.263809 [NOTICE] switch_ivr.c:1450 Transfer sofia/internal/1013 at 60.248.175.38 to XML[1019 at default]
2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing 1013->1019 in context default
2010-10-14 17:50:41.283904 [INFO] switch_core_session.c:1764 Sending early media
2010-10-14 17:50:41.285023 [NOTICE] mod_sofia.c:1994 Pre-Answer sofia/internal/1013 at 60.248.175.38!

seems not work
 
Thank you for you attension
Best Regards
 
Gary
 

 


From: david.ponzone at ipeva.fr
To: freeswitch-users at lists.freeswitch.org
Date: Thu, 14 Oct 2010 10:16:28 +0200
Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite

I think we would need a complete log from the target FS, in order to provide you with any efficient help.


Thank





David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34


Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr  -   www.ipeva-studio.com



Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 14/10/2010 ? 09:55, ? Chiang ?? Chih-Chung Wybie a ?crit :

I tried Connect Two FreeSWITCH Boxes  http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
and my configuration :
 
60.248.175.37
 
(autoload_configs/act.conf.xml)
<list name="domains" default="deny">
      <node type="allow" cidr="60.248.175.38/28"/>
     <!-- <node type="allow" domain="$${domain}"/>-->
</list>

(dialplan/default.xml)
 
<extension name="Dial to 38">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="bridge" data="sofia/internal/$1 at 60.248.175.38"/>
      </condition>
 </extension>
 
(dialplan/public.xml)

<extension name="Calls from 38">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="transfer" data="$1 XML default"/>
      </condition>
 </extension>
 
 
 
60.248.175.38
 
(autoload_configs/act.conf.xml)
<list name="domains" default="deny">
      <node type="allow" cidr="60.248.175.37/28"/>
     <!-- <node type="allow" domain="$${domain}"/>-->
</list>

(dialplan/default.xml)
 
<extension name="Dial to 37">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="bridge" data="sofia/internal/$1 at 60.248.175.37"/>
      </condition>
 </extension>
 
(dialplan/public.xml)

<extension name="Calls from 37">
      <condition field="destination_number" expression="^([0-4]0[01][0-9])$">
        <action application="transfer" data="$1 XML default"/>
      </condition>
 </extension>
 
when I use 1007 at 38 calls to 1007 at 37
it won't show errors but still not work
 
how should I change something ?
 
Thank you for your attension
 
Best Regards
Gary
 
 
 
 
 
> Date: Wed, 13 Oct 2010 22:17:13 -0700
> From: curriegrad2004 at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> 
> You can create a SIP trunk between the two servers so calls can be
> routed back and forth from the servers you've specified.
> 
> 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
> > I tried it
> > but not working
> > and show errors below
> > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493 Cannot create
> > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493 Cannot create
> > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate Failed.
> > Cause: USER_NOT_REGISTERED
> >
> > and for example
> >  I use SIP server 60.248.175.38  user id=1007
> > and I call to 1007 at 60.248.175.37    that will show these errors
> >
> > how should set more?
> >
> > thank you for your attension
> >
> > Best Regards
> > Gary
> >
> > ________________________________
> > Date: Thu, 14 Oct 2010 10:55:49 +0800
> > From: xyangni at gmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
> >
> > I think you can reg at server A and than call B as a normal SIP call to
> > external domain.
> >
> > 2010/10/14 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>
> >
> >
> >
> > Hello
> >
> > I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X-Lite to call
> >
> > my question is how can I call between these servers?
> >
> > if I can , how shoould I set in *.xml? or X-Lite?
> >
> > thank you for your attension
> >
> >
> > Best Regards
> > Gary
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________ FreeSWITCH-users mailing
> > list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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_______________________________________________ FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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_______________________________________________ FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org_______________________________________________
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_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org_______________________________________________
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From david.ponzone at ipeva.fr  Mon Oct 18 01:52:31 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 18 Oct 2010 10:52:31 +0200
Subject: [Freeswitch-users] some questions of freeswitch & X-Lite
In-Reply-To: <SNT130-w60DD13A21F1F843BC143E6FB5A0@phx.gbl>
References: <SNT130-w309C1C50F08EBBFD06F2AEFB560@phx.gbl>, , , , ,
	<SNT130-w4479843CA9BA7F78DCC8A4FB560@phx.gbl>, , , , ,
	<AANLkTi=5Y=N_TnWrSzRS-N++_5boaJVQZbXCCpwTSfS9@mail.gmail.com>, ,
	, , , <SNT130-w3803D906A1C0DF02775212FB560@phx.gbl>, , , , ,
	<AANLkTi=+fp_rptm997+MJOXq9MwK9nZZ8c0XwBWd88iW@mail.gmail.com>, ,
	, , <SNT130-w19BC5B122DF6FF3378E4A1FB560@phx.gbl>, , , ,
	<C243AB2B-E936-4AFD-AAB6-DA2A8B3FEB0E@ipeva.fr>, , ,
	<SNT130-w45AD76AD0E8700744A3844FB560@phx.gbl>, , ,
	<6A8D7D0E-4DF5-4F35-8A00-F52FECC4F826@ipeva.fr>, ,
	<SNT130-w6005F900786BDFBE48FF43FB570@phx.gbl>, ,
	<AD407430-9024-4E0D-A860-9026FD2837A3@ipeva.fr>,
	<SNT130-w445656F3A9C14ED384A1B0FB5A0@phx.gbl>,
	<056892B2-7226-4941-AF71-4CC5ABB01835@ipeva.fr>
	<SNT130-w60DD13A21F1F843BC143E6FB5A0@phx.gbl>
Message-ID: <92674D12-BDE9-4CB8-8A49-B6D70377F574@ipeva.fr>

I would reverse back the modifications you made to acl.conf.xml.
At the moment, the call is hitting the box as if it was not from a  
registered user, so it's hitting public dialplan instead of default.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 18/10/2010 ? 10:05, ? Chiang ?? Chih-Chung Wybie a ?crit :

> sorry
>
> but I tried it on a single server it can work
>
> and I refer to http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes 
>  to set connect two FSs
>
> and my all configs modified below
> 1, acl.conf.xml
> 60.248.175.37
> <list name="domains" default="deny">
>      <node type="allow" cidr="60.248.175.38/28"/>
>      <!--<node type="allow" domain="$${domain}"/>-->
>     </list>
>
> 60.248.175.38
> <list name="domains" default="deny">
>      <node type="allow" cidr="60.248.175.37/28"/>
>      <!--<node type="allow" domain="$${domain}"/>-->
>     </list>
>
> 2, dialplan/default.xml
> 60.248.175.37
> <extension name="Local_Extension">
>       <condition field="destination_number" expression="^(40[01] 
> [0-9])$">
>         <action application="set" data="RECORD_COPYRIGHT=(c) 2010"/>
>         <action application="set" data="RECORD_SOFTWARE=FreeSwitch"/>
>         <action application="set" data="RECORD_ARTIST=FreeSwitch"/>
>         <action application="set" data="RECORD_COMMENT=FreeSwitch"/>
>         <action application="set" data="RECORD_DATE=${strftime(%Y-%m- 
> %d %H:%M)}"/>
>         <action application="set" data="RECORD_STEREO=true"/>
>         <action application="record_session" data="$${base_dir}/ 
> recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_$ 
> {destination_number}_${caller_id_number}.wav"/>
>         <action application="set" data="dialed_extension=$1"/>
>         <action application="export" data="dialed_extension=$1"/>
>         <!-- add for busy transfer -->
>         <action application="set_user" data="${dialed_extension}@$ 
> {domain_name}"/>
>         <action application="limit" data="$${domain_name} $ 
> {dialed_extension} ${max_calls} ${fail_over}"/>
>         <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o| 
> s] <app> -->
>         <action application="bind_meta_app" data="1 b s  
> execute_extension::dx XML features"/>
>         <action application="bind_meta_app" data="2 b s  
> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y- 
> %m-%d-%H-%M-%S)}.wav"/>
>         <action application="bind_meta_app" data="3 b s  
> execute_extension::cf XML features"/>
>         <action application="set" data="ringback=${us-ring}"/>
>         <action application="set" data="transfer_ringback=$$ 
> {hold_music}"/>
>         <action application="set" data="call_timeout=30"/>
>         <!-- <action application="set" data="sip_exclude_contact=$ 
> {network_addr}"/> -->
>         <action application="set" data="hangup_after_bridge=true"/>
>         <!--<action application="set"  
> data 
> = 
> "continue_on_fail 
> = 
> NORMAL_TEMPORARY_FAILURE 
> ,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
>         <action application="set" data="continue_on_fail=true"/>
>         <action application="hash" data="insert/${domain_name}- 
> call_return/${dialed_extension}/${caller_id_number}"/>
>         <action application="hash" data="insert/${domain_name}- 
> last_dial_ext/${dialed_extension}/${uuid}"/>
>         <action application="set" data="called_party_callgroup=$ 
> {user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
>         <!--<action application="export"  
> data="nolocal:sip_secure_media=${user_data(${dialed_extension}@$ 
> {domain_name} var sip_secure_media)}"/>-->
>         <action application="hash" data="insert/${domain_name}- 
> last_dial/${called_party_callgroup}/${uuid}"/>
>         <action application="bridge" data="user/${dialed_extension}@$ 
> {domain_name}"/>
>         <!-- <action application="record_fsv" data="$${base_dir}/ 
> recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_$ 
> {destination_number}_${caller_id_number}.fsv"/>-->
>         <action application="answer"/>
>         <action application="sleep" data="1000"/>
>         <action application="bridge" data="loopback/ 
> app=voicemail:default ${domain_name} ${dialed_extension}"/>
>       </condition>
>     </extension>
>     <extension name="Dial to 38">
>       <condition field="destination_number" expression="^(10[01] 
> [0-9])$">
>         <action application="bridge" data="sofia/internal/$1 at 60.248.175.38 
> "/>
>       </condition>
>     </extension>
>
>  60.248.175.38
> <extension name="Local_Extension">
>       <condition field="destination_number" expression="^(10[01] 
> [0-9])$">
>         <action application="set" data="RECORD_COPYRIGHT=(c) 2010"/>
>         <action application="set" data="RECORD_SOFTWARE=FreeSwitch"/>
>         <action application="set" data="RECORD_ARTIST=FreeSwitch"/>
>         <action application="set" data="RECORD_COMMENT=FreeSwitch"/>
>         <action application="set" data="RECORD_DATE=${strftime(%Y-%m- 
> %d %H:%M)}"/>
>         <action application="set" data="RECORD_STEREO=true"/>
>        <!-- <action application="record_session" data="$${base_dir}/ 
> recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_$ 
> {destination_number}_${caller_id_number}.wav"/>-->
>         <action application="set" data="dialed_extension=$1"/>
>         <action application="export" data="dialed_extension=$1"/>
>         <!-- add for busy transfer -->
>         <action application="set_user" data="${dialed_extension}@$ 
> {domain_name}"/>
>         <action application="limit" data="$${domain_name} $ 
> {dialed_extension} ${max_calls} ${fail_over}"/>
>         <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o| 
> s] <app> -->
>         <action application="bind_meta_app" data="1 b s  
> execute_extension::dx XML features"/>
>         <action application="bind_meta_app" data="2 b s  
> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y- 
> %m-%d-%H-%M-%S)}.wav"/>
>         <action application="bind_meta_app" data="3 b s  
> execute_extension::cf XML features"/>
>         <action application="set" data="ringback=${us-ring}"/>
>         <action application="set" data="transfer_ringback=$$ 
> {hold_music}"/>
>         <action application="set" data="call_timeout=30"/>
>         <!-- <action application="set" data="sip_exclude_contact=$ 
> {network_addr}"/> -->
>         <action application="set" data="hangup_after_bridge=true"/>
>         <!--<action application="set"  
> data 
> = 
> "continue_on_fail 
> = 
> NORMAL_TEMPORARY_FAILURE 
> ,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
>         <action application="set" data="continue_on_fail=true"/>
>         <action application="hash" data="insert/${domain_name}- 
> call_return/${dialed_extension}/${caller_id_number}"/>
>         <action application="hash" data="insert/${domain_name}- 
> last_dial_ext/${dialed_extension}/${uuid}"/>
>         <action application="set" data="called_party_callgroup=$ 
> {user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
>         <!--<action application="export"  
> data="nolocal:sip_secure_media=${user_data(${dialed_extension}@$ 
> {domain_name} var sip_secure_media)}"/>-->
>         <action application="hash" data="insert/${domain_name}- 
> last_dial/${called_party_callgroup}/${uuid}"/>
>         <action application="bridge" data="user/${dialed_extension}@$ 
> {domain_name}"/>
>         <!-- <action application="record_fsv" data="$${base_dir}/ 
> recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_$ 
> {destination_number}_${caller_id_number}.fsv"/>-->
>         <action application="answer"/>
>         <action application="sleep" data="1000"/>
>         <action application="bridge" data="loopback/ 
> app=voicemail:default ${domain_name} ${dialed_extension}"/>
>       </condition>
>     </extension>
>     <extension name="Dial to 37">
>       <condition field="destination_number" expression="^(40[01] 
> [0-9])$">
>         <action application="bridge" data="sofia/internal/$1 at 60.248.175.37 
> "/>
>       </condition>
>     </extension>
>
> 3, dialplan/public.xml
> 60.248.175.37
> <extension name="Calls from 38">
>       <condition field="destination_number" expression="^(40[01] 
> [0-9])$">
>         <action application="transfer" data="$1 XML default"/>
>       </condition>
>     </extension>
>
> 60.248.175.38
> <extension name="Calls from 37">
>       <condition field="destination_number" expression="^(10[01] 
> [0-9])$">
>         <action application="transfer" data="$1 XML default"/>
>       </condition>
>     </extension>
>
> 4,DEBUG log
> I use 1013 at 64.248.175.38 calls to 4001 at 64.248.175.37
> log from 38 (that's all)
>
> freeswitch at ubuntu> 2010-10-18 15:56:15.221328 [DEBUG] sofia.c:5849  
> IP 60.248.175.43 Approved by acl "domains[]". Access Granted.
> 2010-10-18 15:56:15.221328 [NOTICE] switch_channel.c:675 New Channel sofia/internal/1013 at 60.248.175.38 
> [bfc190d6-9371-4bfb-9822-2cbfaf1d6ebc]
> 2010-10-18 15:56:15.224062 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_NEW
> 2010-10-18 15:56:15.224062 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1013 at 60.248.175.38 
> ) State NEW
> 2010-10-18 15:56:15.236344 [DEBUG] sofia.c:4200 Channel sofia/internal/1013 at 60.248.175.38 
>  entering state [received][100]
> 2010-10-18 15:56:15.236344 [DEBUG] sofia.c:4211 Remote SDP:
> v=0
> o=- 6 2 IN IP4 60.248.175.43
> s=CounterPath X-Lite 3.0
> c=IN IP4 60.248.175.43
> t=0 0
> m=audio 41530 RTP/AVP 107 0 8 101
> a=rtpmap:107 BV32/16000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=alt:1 3 : KOojNg3s EbMN6/84 60.248.175.43 41530
> a=alt:2 2 : FZp3J381 HADuy4Pa 192.168.182.1 41530
> a=alt:3 1 : Zq7Ue+wV Zj2BjStZ 192.168.60.1 41530
> 2010-10-18 15:56:15.236344 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[G7221:115:32000:20]
> 2010-10-18 15:56:15.236344 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[G7221:107:16000:20]
> 2010-10-18 15:56:15.236344 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[G722:9:8000:20]
> 2010-10-18 15:56:15.236344 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[PCMU:0:8000:20]
> 2010-10-18 15:56:15.236344 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[PCMA:8:8000:20]
> 2010-10-18 15:56:15.236344 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[GSM:3:8000:20]
> 2010-10-18 15:56:15.236344 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [PCMU:0:8000:20]/[G7221:115:32000:20]
> 2010-10-18 15:56:15.236344 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [PCMU:0:8000:20]/[G7221:107:16000:20]
> 2010-10-18 15:56:15.236344 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [PCMU:0:8000:20]/[G722:9:8000:20]
> 2010-10-18 15:56:15.236344 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [PCMU:0:8000:20]/[PCMU:0:8000:20]
> 2010-10-18 15:56:15.237437 [DEBUG] sofia_glue.c:2376 Set Codec sofia/internal/1013 at 60.248.175.38PCMU 
> /8000 20 ms 160 samples
> 2010-10-18 15:56:15.237437 [DEBUG] sofia_glue.c:3610 Set 2833 dtmf  
> send/recv payload to 101
> 2010-10-18 15:56:15.237437 [DEBUG] sofia.c:4357 (sofia/internal/1013 at 60.248.175.38 
> ) State Change CS_NEW -> CS_INIT
> 2010-10-18 15:56:15.237437 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_INIT
> 2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1013 at 60.248.175.38 
> ) State INIT
> 2010-10-18 15:56:15.237437 [DEBUG] mod_sofia.c:83 sofia/internal/1013 at 60.248.175.38 
>  SOFIA INIT
> 2010-10-18 15:56:15.237437 [DEBUG] mod_sofia.c:117 (sofia/internal/1013 at 60.248.175.38 
> ) State Change CS_INIT -> CS_ROUTING
> 2010-10-18 15:56:15.237437 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1013 at 60.248.175.38 
> ) State INIT going to sleep
> 2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_ROUTING
> 2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38 
> ) State ROUTING
> 2010-10-18 15:56:15.237437 [DEBUG] mod_sofia.c:140 sofia/internal/1013 at 60.248.175.38 
>  SOFIA ROUTING
> 2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1013 at 60.248.175.38Standard 
>  ROUTING
> 2010-10-18 15:56:15.237437 [INFO] mod_dialplan_xml.c:331 Processing  
> 1013->4001 in context public
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->unloop]  
> continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (PASS) [unloop] $ 
> {unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [unloop] $ 
> {sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public- 
> >outside_call] continue=true
> Dialplan: sofia/internal/1013 at 60.248.175.38 Absolute Condition  
> [outside_call]
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(outside_call=true)
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public- 
> >call_debug] continue=true
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL)  
> [call_debug] ${call_debug}(false) =~ /^true$/ break=never
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public- 
> >public_extensions] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL)  
> [public_extensions] destination_number(4001) =~ /^(10[01][0-9])$/  
> break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->Calls  
> from 37] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [Calls from  
> 37] destination_number(4001) =~ /^(10[01][0-9])$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public- 
> >public_did] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL)  
> [public_did] destination_number(4001) =~ /^(5551212)$/ break=on-false
> 2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1013 at 60.248.175.38 
> ) State Change CS_ROUTING -> CS_EXECUTE
> 2010-10-18 15:56:15.237437 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38 
> ) State ROUTING going to sleep
> 2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_EXECUTE
> 2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38 
> ) State EXECUTE
> 2010-10-18 15:56:15.237437 [DEBUG] mod_sofia.c:233 sofia/internal/1013 at 60.248.175.38 
>  SOFIA EXECUTE
> 2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1013 at 60.248.175.38Standard 
>  EXECUTE
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(outside_call=true)
> 2010-10-18 15:56:15.237437 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [outside_call]=[true]
> 2010-10-18 15:56:15.237437 [NOTICE] switch_core_state_machine.c:185 sofia/internal/1013 at 60.248.175.38has 
>  executed the last dialplan instruction, hanging up.
> 2010-10-18 15:56:15.237437 [NOTICE] switch_core_state_machine.c:187 Hangupsofia/internal/1013 at 60.248.175.38 
>  [CS_EXECUTE] [NORMAL_CLEARING]
> 2010-10-18 15:56:15.237437 [DEBUG] switch_channel.c:2145 Send signal sofia/internal/1013 at 60.248.175.38 
> [KILL]
> 2010-10-18 15:56:15.237437 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38 
> ) State EXECUTE going to sleep
> 2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_HANGUP
> 2010-10-18 15:56:15.237437 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/1013 at 60.248.175.38 
> ) State HANGUP
> 2010-10-18 15:56:15.237437 [DEBUG] mod_sofia.c:441 Channel sofia/internal/1013 at 60.248.175.38 
>  hanging up, cause: NORMAL_CLEARING
> 2010-10-18 15:56:15.263831 [DEBUG] mod_sofia.c:503 Responding to  
> INVITE with: 480
> 2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1013 at 60.248.175.38Standard 
>  HANGUP, cause: NORMAL_CLEARING
> 2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/1013 at 60.248.175.38 
> ) State HANGUP going to sleep
> 2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/1013 at 60.248.175.38 
> ) State Change CS_HANGUP -> CS_REPORTING
> 2010-10-18 15:56:15.263831 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_REPORTING
> 2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:589 (sofia/internal/1013 at 60.248.175.38 
> ) State REPORTING
> 2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1013 at 60.248.175.38Standard 
>  REPORTING, cause: NORMAL_CLEARING
> 2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:589 (sofia/internal/1013 at 60.248.175.38 
> ) State REPORTING going to sleep
> 2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/1013 at 60.248.175.38 
> ) State Change CS_REPORTING -> CS_DESTROY
> 2010-10-18 15:56:15.263831 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-18 15:56:15.263831 [DEBUG] switch_core_session.c:1170  
> Session 4 (sofia/internal/1013 at 60.248.175.38) Locked, Waiting on  
> external entities
> 2010-10-18 15:56:15.263831 [NOTICE] switch_core_session.c:1188  
> Session 4 (sofia/internal/1013 at 60.248.175.38) Ended
> 2010-10-18 15:56:15.263831 [NOTICE] switch_core_session.c:1190 Close Channelsofia/internal/1013 at 60.248.175.38 
>  [CS_DESTROY]
> 2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:428 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_DESTROY
> 2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:438 (sofia/internal/1013 at 60.248.175.38 
> ) State DESTROY
> 2010-10-18 15:56:15.263831 [DEBUG] mod_sofia.c:350 sofia/internal/1013 at 60.248.175.38 
>  SOFIA DESTROY
> 2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1013 at 60.248.175.38Standard 
>  DESTROY
> 2010-10-18 15:56:15.263831 [DEBUG] switch_core_state_machine.c:438 (sofia/internal/1013 at 60.248.175.38 
> ) State DESTROY going to sleep
>
> and I use 4001 at 64.248.175.37 calls to 1013 at 64.248.175.38
> log from 37 (that's all)
>
> 2010-10-18 15:59:04.730021 [DEBUG] sofia.c:5849 IP 60.248.175.42  
> Approved by acl "domains[]". Access Granted.
> 2010-10-18 15:59:04.730021 [NOTICE] switch_channel.c:675 New Channel sofia/internal/4001 at 60.248.175.37 
> [0a71af92-1564-49e1-9a8e-9f5432b1653b]
> 2010-10-18 15:59:04.735619 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/4001 at 60.248.175.37 
> ) Running State Change CS_NEW
> 2010-10-18 15:59:04.735619 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/4001 at 60.248.175.37 
> ) State NEW
> 2010-10-18 15:59:04.877644 [DEBUG] sofia.c:4200 Channel sofia/internal/4001 at 60.248.175.37 
>  entering state [received][100]
> 2010-10-18 15:59:04.877644 [DEBUG] sofia.c:4211 Remote SDP:
> v=0
> o=- 6 2 IN IP4 60.248.175.42
> s=CounterPath X-Lite 3.0
> c=IN IP4 60.248.175.42
> t=0 0
> m=audio 51406 RTP/AVP 107 0 8 101
> a=rtpmap:107 BV32/16000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=alt:1 3 : s7WKw7mB W35tlaoK 60.248.175.42 51406
> a=alt:2 2 : SfvGIlEF m2IrZtjW 192.168.24.1 51406
> a=alt:3 1 : fQ/a57gE kxcr2NP4 192.168.137.1 51406
> 2010-10-18 15:59:04.880340 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[G7221:115:32000:20]
> 2010-10-18 15:59:04.881429 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[G7221:107:16000:20]
> 2010-10-18 15:59:04.881429 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[G722:9:8000:20]
> 2010-10-18 15:59:04.881429 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[PCMU:0:8000:20]
> 2010-10-18 15:59:04.883611 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[PCMA:8:8000:20]
> 2010-10-18 15:59:04.883611 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[GSM:3:8000:20]
> 2010-10-18 15:59:04.884891 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [PCMU:0:8000:20]/[G7221:115:32000:20]
> 2010-10-18 15:59:04.886125 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [PCMU:0:8000:20]/[G7221:107:16000:20]
> 2010-10-18 15:59:04.886125 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [PCMU:0:8000:20]/[G722:9:8000:20]
> 2010-10-18 15:59:04.886125 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [PCMU:0:8000:20]/[PCMU:0:8000:20]
> 2010-10-18 15:59:04.887562 [DEBUG] sofia_glue.c:2376 Set Codec sofia/internal/4001 at 60.248.175.37PCMU 
> /8000 20 ms 160 samples
> 2010-10-18 15:59:04.888900 [DEBUG] sofia_glue.c:3610 Set 2833 dtmf  
> send/recv payload to 101
> 2010-10-18 15:59:04.888900 [DEBUG] sofia.c:4357 (sofia/internal/4001 at 60.248.175.37 
> ) State Change CS_NEW -> CS_INIT
> 2010-10-18 15:59:04.890190 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/4001 at 60.248.175.37 
>  [BREAK]
> 2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/4001 at 60.248.175.37 
> ) Running State Change CS_INIT
> 2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/4001 at 60.248.175.37 
> ) State INIT
> 2010-10-18 15:59:04.891645 [DEBUG] mod_sofia.c:83 sofia/internal/4001 at 60.248.175.37 
>  SOFIA INIT
> 2010-10-18 15:59:04.891645 [DEBUG] mod_sofia.c:117 (sofia/internal/4001 at 60.248.175.37 
> ) State Change CS_INIT -> CS_ROUTING
> 2010-10-18 15:59:04.891645 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/4001 at 60.248.175.37 
>  [BREAK]
> 2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/4001 at 60.248.175.37 
> ) State INIT going to sleep
> 2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/4001 at 60.248.175.37 
> ) Running State Change CS_ROUTING
> 2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/4001 at 60.248.175.37 
> ) State ROUTING
> 2010-10-18 15:59:04.891645 [DEBUG] mod_sofia.c:140 sofia/internal/4001 at 60.248.175.37 
>  SOFIA ROUTING
> 2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:77 sofia/internal/4001 at 60.248.175.37Standard 
>  ROUTING
> 2010-10-18 15:59:04.891645 [INFO] mod_dialplan_xml.c:331 Processing  
> 4001->1013 in context public
> Dialplan: sofia/internal/4001 at 60.248.175.37 parsing [public->unloop]  
> continue=false
> Dialplan: sofia/internal/4001 at 60.248.175.37 Regex (PASS) [unloop] $ 
> {unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/4001 at 60.248.175.37 Regex (FAIL) [unloop] $ 
> {sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/4001 at 60.248.175.37 parsing [public- 
> >outside_call] continue=true
> Dialplan: sofia/internal/4001 at 60.248.175.37 Absolute Condition  
> [outside_call]
> Dialplan: sofia/internal/4001 at 60.248.175.37 Action  
> set(outside_call=true)
> Dialplan: sofia/internal/4001 at 60.248.175.37 parsing [public- 
> >call_debug] continue=true
> Dialplan: sofia/internal/4001 at 60.248.175.37 Regex (FAIL)  
> [call_debug] ${call_debug}(false) =~ /^true$/ break=never
> Dialplan: sofia/internal/4001 at 60.248.175.37 parsing [public- 
> >public_extensions] continue=false
> Dialplan: sofia/internal/4001 at 60.248.175.37 Regex (FAIL)  
> [public_extensions] destination_number(1013) =~ /^(40[01][0-9])$/  
> break=on-false
> Dialplan: sofia/internal/4001 at 60.248.175.37 parsing [public->Calls  
> from 38] continue=false
> Dialplan: sofia/internal/4001 at 60.248.175.37 Regex (FAIL) [Calls from  
> 38] destination_number(1013) =~ /^(40[01][0-9])$/ break=on-false
> Dialplan: sofia/internal/4001 at 60.248.175.37 parsing [public- 
> >public_did] continue=false
> Dialplan: sofia/internal/4001 at 60.248.175.37 Regex (FAIL)  
> [public_did] destination_number(1013) =~ /^(5551212)$/ break=on-false
> 2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/4001 at 60.248.175.37 
> ) State Change CS_ROUTING -> CS_EXECUTE
> 2010-10-18 15:59:04.891645 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/4001 at 60.248.175.37 
>  [BREAK]
> 2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/4001 at 60.248.175.37 
> ) State ROUTING going to sleep
> 2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/4001 at 60.248.175.37 
> ) Running State Change CS_EXECUTE
> 2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/4001 at 60.248.175.37 
> ) State EXECUTE
> 2010-10-18 15:59:04.891645 [DEBUG] mod_sofia.c:233 sofia/internal/4001 at 60.248.175.37 
>  SOFIA EXECUTE
> 2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:157 sofia/internal/4001 at 60.248.175.37Standard 
>  EXECUTE
> EXECUTE sofia/internal/4001 at 60.248.175.37 set(outside_call=true)
> 2010-10-18 15:59:04.891645 [DEBUG] mod_dptools.c:817 sofia/internal/4001 at 60.248.175.37 
>  SET [outside_call]=[true]
> 2010-10-18 15:59:04.891645 [NOTICE] switch_core_state_machine.c:185 sofia/internal/4001 at 60.248.175.37has 
>  executed the last dialplan instruction, hanging up.
> 2010-10-18 15:59:04.891645 [NOTICE] switch_core_state_machine.c:187 Hangupsofia/internal/4001 at 60.248.175.37 
>  [CS_EXECUTE] [NORMAL_CLEARING]
> 2010-10-18 15:59:04.891645 [DEBUG] switch_channel.c:2145 Send signal sofia/internal/4001 at 60.248.175.37 
> [KILL]
> 2010-10-18 15:59:04.891645 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/4001 at 60.248.175.37 
>  [BREAK]
> 2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/4001 at 60.248.175.37 
> ) State EXECUTE going to sleep
> 2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/4001 at 60.248.175.37 
> ) Running State Change CS_HANGUP
> 2010-10-18 15:59:04.891645 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/4001 at 60.248.175.37 
> ) State HANGUP
> 2010-10-18 15:59:04.891645 [DEBUG] mod_sofia.c:441 Channel sofia/internal/4001 at 60.248.175.37 
>  hanging up, cause: NORMAL_CLEARING
> 2010-10-18 15:59:04.928639 [DEBUG] mod_sofia.c:503 Responding to  
> INVITE with: 480
> 2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:46 sofia/internal/4001 at 60.248.175.37Standard 
>  HANGUP, cause: NORMAL_CLEARING
> 2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/4001 at 60.248.175.37 
> ) State HANGUP going to sleep
> 2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/4001 at 60.248.175.37 
> ) State Change CS_HANGUP -> CS_REPORTING
> 2010-10-18 15:59:04.928639 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/4001 at 60.248.175.37 
>  [BREAK]
> 2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/4001 at 60.248.175.37 
> ) Running State Change CS_REPORTING
> 2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:589 (sofia/internal/4001 at 60.248.175.37 
> ) State REPORTING
> 2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:53 sofia/internal/4001 at 60.248.175.37Standard 
>  REPORTING, cause: NORMAL_CLEARING
> 2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:589 (sofia/internal/4001 at 60.248.175.37 
> ) State REPORTING going to sleep
> 2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/4001 at 60.248.175.37 
> ) State Change CS_REPORTING -> CS_DESTROY
> 2010-10-18 15:59:04.928639 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/4001 at 60.248.175.37 
>  [BREAK]
> 2010-10-18 15:59:04.928639 [DEBUG] switch_core_session.c:1170  
> Session 3 (sofia/internal/4001 at 60.248.175.37) Locked, Waiting on  
> external entities
> 2010-10-18 15:59:04.928639 [NOTICE] switch_core_session.c:1188  
> Session 3 (sofia/internal/4001 at 60.248.175.37) Ended
> 2010-10-18 15:59:04.928639 [NOTICE] switch_core_session.c:1190 Close Channelsofia/internal/4001 at 60.248.175.37 
>  [CS_DESTROY]
> 2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:428 (sofia/internal/4001 at 60.248.175.37 
> ) Running State Change CS_DESTROY
> 2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:438 (sofia/internal/4001 at 60.248.175.37 
> ) State DESTROY
> 2010-10-18 15:59:04.928639 [DEBUG] mod_sofia.c:350 sofia/internal/4001 at 60.248.175.37 
>  SOFIA DESTROY
> 2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:60 sofia/internal/4001 at 60.248.175.37Standard 
>  DESTROY
> 2010-10-18 15:59:04.928639 [DEBUG] switch_core_state_machine.c:438 (sofia/internal/4001 at 60.248.175.37 
> ) State DESTROY going to sleep
>
> that's all my information I really hope that I can solve this  
> problem but I have no idea about this function
>
> Thank you for your attsnsion
>
> Best Regards
>
> Gary
> From: david.ponzone at ipeva.fr
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 18 Oct 2010 09:25:25 +0200
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> You failed to tell us which server this log comes from.
>
> In order to get a result, I think you need to try to understand what  
> the logs mean and you need to understand that to help you, we need  
> detailed logs and relevant config files.
> From the beginning of this thread, you failed to provide both at the  
> same time.
>
> Also, that's not a basic config.
> Have you tried before to do a single server config ?
> I would start there, so that you can understand how FreeSWITCH  
> works, why the logs are meaning ful and how to read them, which  
> dialplans are parsed and when.
> If you don't do that, you're going to waste people's time because  
> you want to avoid some required (and important) learning steps.
>
> Thank you
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
> message, merci de le d?truire imm?diatement et d'avertir  
> l'exp?diteur.
>
>
>
>
> Le 18/10/2010 ? 02:37, ? Chiang ?? Chih-Chung Wybie a ?crit :
>
> I tried to set different expression but it still can't work
>
> debug log below
>
> 2010-10-18 08:34:19.462617 [DEBUG] sofia.c:5849 IP 60.248.175.43  
> Approved by acl "domains[]". Access Granted.
> 2010-10-18 08:34:19.462617 [NOTICE] switch_channel.c:675 New Channel sofia/internal/1013 at 60.248.175.38 
> [1f4277c1-9ded-49c5-b63f-a312bead9d5f]
> 2010-10-18 08:34:19.464814 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_NEW
> 2010-10-18 08:34:19.464814 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1013 at 60.248.175.38 
> ) State NEW
> 2010-10-18 08:34:19.475647 [DEBUG] sofia.c:4200 Channel sofia/internal/1013 at 60.248.175.38 
>  entering state [received][100]
> 2010-10-18 08:34:19.475647 [DEBUG] sofia.c:4211 Remote SDP:
> v=0
> o=- 0 2 IN IP4 60.248.175.43
> s=CounterPath X-Lite 3.0
> c=IN IP4 60.248.175.43
> t=0 0
> m=audio 32692 RTP/AVP 107 0 8 101
> a=rtpmap:107 BV32/16000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=alt:1 3 : d7A5MyJF tN44o7uR 60.248.175.43 32692
> a=alt:2 2 : cCI83KQt qU62M7Ia 192.168.182.1 32692
> a=alt:3 1 : qtl4Jt3z ug+kYltk 192.168.60.1 32692
> 2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[G7221:115:32000:20]
> 2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[G7221:107:16000:20]
> 2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[G722:9:8000:20]
> 2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[PCMU:0:8000:20]
> 2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[PCMA:8:8000:20]
> 2010-10-18 08:34:19.475647 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [BV32:107:16000:20]/[GSM:3:8000:20]
> 2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [PCMU:0:8000:20]/[G7221:115:32000:20]
> 2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [PCMU:0:8000:20]/[G7221:107:16000:20]
> 2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [PCMU:0:8000:20]/[G722:9:8000:20]
> 2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3671 Audio Codec  
> Compare [PCMU:0:8000:20]/[PCMU:0:8000:20]
> 2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:2376 Set Codec sofia/internal/1013 at 60.248.175.38PCMU 
> /8000 20 ms 160 samples
> 2010-10-18 08:34:19.476688 [DEBUG] sofia_glue.c:3610 Set 2833 dtmf  
> send/recv payload to 101
> 2010-10-18 08:34:19.476688 [DEBUG] sofia.c:4357 (sofia/internal/1013 at 60.248.175.38 
> ) State Change CS_NEW -> CS_INIT
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_INIT
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1013 at 60.248.175.38 
> ) State INIT
> 2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:83 sofia/internal/1013 at 60.248.175.38 
>  SOFIA INIT
> 2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:117 (sofia/internal/1013 at 60.248.175.38 
> ) State Change CS_INIT -> CS_ROUTING
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1013 at 60.248.175.38 
> ) State INIT going to sleep
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_ROUTING
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38 
> ) State ROUTING
> 2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:140 sofia/internal/1013 at 60.248.175.38 
>  SOFIA ROUTING
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1013 at 60.248.175.38Standard 
>  ROUTING
> 2010-10-18 08:34:19.476688 [INFO] mod_dialplan_xml.c:331 Processing  
> 1013->4001 in context public
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->unloop]  
> continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (PASS) [unloop] $ 
> {unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [unloop] $ 
> {sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public- 
> >outside_call] continue=true
> Dialplan: sofia/internal/1013 at 60.248.175.38 Absolute Condition  
> [outside_call]
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(outside_call=true)
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public- 
> >call_debug] continue=true
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL)  
> [call_debug] ${call_debug}(false) =~ /^true$/ break=never
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public- 
> >public_extensions] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL)  
> [public_extensions] destination_number(4001) =~ /^(10[01][0-9])$/  
> break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public->Calls  
> from 37] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [Calls from  
> 37] destination_number(4001) =~ /^(10[01][0-9])$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [public- 
> >public_did] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL)  
> [public_did] destination_number(4001) =~ /^(5551212)$/ break=on-false
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1013 at 60.248.175.38 
> ) State Change CS_ROUTING -> CS_EXECUTE
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38 
> ) State ROUTING going to sleep
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_EXECUTE
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38 
> ) State EXECUTE
> 2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:233 sofia/internal/1013 at 60.248.175.38 
>  SOFIA EXECUTE
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1013 at 60.248.175.38Standard 
>  EXECUTE
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(outside_call=true)
> 2010-10-18 08:34:19.476688 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [outside_call]=[true]
> 2010-10-18 08:34:19.476688 [NOTICE] switch_core_state_machine.c:185 sofia/internal/1013 at 60.248.175.38has 
>  executed the last dialplan instruction, hanging up.
> 2010-10-18 08:34:19.476688 [NOTICE] switch_core_state_machine.c:187 Hangupsofia/internal/1013 at 60.248.175.38 
>  [CS_EXECUTE] [NORMAL_CLEARING]
> 2010-10-18 08:34:19.476688 [DEBUG] switch_channel.c:2145 Send signal sofia/internal/1013 at 60.248.175.38 
> [KILL]
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38 
> ) State EXECUTE going to sleep
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_HANGUP
> 2010-10-18 08:34:19.476688 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/1013 at 60.248.175.38 
> ) State HANGUP
> 2010-10-18 08:34:19.476688 [DEBUG] mod_sofia.c:441 Channel sofia/internal/1013 at 60.248.175.38 
>  hanging up, cause: NORMAL_CLEARING
> 2010-10-18 08:34:19.494277 [DEBUG] mod_sofia.c:503 Responding to  
> INVITE with: 480
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1013 at 60.248.175.38Standard 
>  HANGUP, cause: NORMAL_CLEARING
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/1013 at 60.248.175.38 
> ) State HANGUP going to sleep
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/1013 at 60.248.175.38 
> ) State Change CS_HANGUP -> CS_REPORTING
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_REPORTING
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:589 (sofia/internal/1013 at 60.248.175.38 
> ) State REPORTING
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1013 at 60.248.175.38Standard 
>  REPORTING, cause: NORMAL_CLEARING
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:589 (sofia/internal/1013 at 60.248.175.38 
> ) State REPORTING going to sleep
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/1013 at 60.248.175.38 
> ) State Change CS_REPORTING -> CS_DESTROY
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_session.c:1170  
> Session 61 (sofia/internal/1013 at 60.248.175.38) Locked, Waiting on  
> external entities
> 2010-10-18 08:34:19.494277 [NOTICE] switch_core_session.c:1188  
> Session 61 (sofia/internal/1013 at 60.248.175.38) Ended
> 2010-10-18 08:34:19.494277 [NOTICE] switch_core_session.c:1190 Close Channelsofia/internal/1013 at 60.248.175.38 
>  [CS_DESTROY]
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:428 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_DESTROY
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:438 (sofia/internal/1013 at 60.248.175.38 
> ) State DESTROY
> 2010-10-18 08:34:19.494277 [DEBUG] mod_sofia.c:350 sofia/internal/1013 at 60.248.175.38 
>  SOFIA DESTROY
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1013 at 60.248.175.38Standard 
>  DESTROY
> 2010-10-18 08:34:19.494277 [DEBUG] switch_core_state_machine.c:438 (sofia/internal/1013 at 60.248.175.38 
> ) State DESTROY going to sleep
>
> Thank you for your attension
>
> Best Regards
>
> Gary
>
> From: david.ponzone at ipeva.fr
> To: freeswitch-users at lists.freeswitch.org
> Date: Fri, 15 Oct 2010 11:16:41 +0200
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> You need to understand that if you dial a number that you want to be  
> routed to your 60.248.175.37, you need NOT TO HAVE this number  
> intercepted by the default context on 60.248.175.38.
> In the log you sent, the call is staying local to 60.248.175.38.
> It never reaches 60.248.175.37.
>
> You need to have different numbering plan (you can't have a 4001 on  
> both machines and hope to reach both with the same number).
> Or you keep the same numbering plan, but you use a prefix in the  
> dialplan to reach the other host.
> For instance you would dial 994001, but your dialplan will match it  
> with a regexp like ^99(4\d{3})$ and bridge $1 (so only 4001) to the  
> other FS.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
> message, merci de le d?truire imm?diatement et d'avertir  
> l'exp?diteur.
>
>
>
>
> Le 15/10/2010 ? 10:52, ? Chiang ?? Chih-Chung Wybie a ?crit :
>
> Hello this is my debug log
>
> I use 1013 at 60.248.175.38 call to 4001 at 60.248.175.37
>
> EXECUTE sofia/internal/1013 at 60.248.175.38 transfer(4001 XML default)
> 2010-10-15 16:49:30.666644 [DEBUG] switch_ivr.c:1444 (sofia/internal/1013 at 60.248.175.38 
> ) State Change CS_EXECUTE -> CS_ROUTING
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_session.c:642 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-15 16:49:30.666644 [NOTICE] switch_ivr.c:1450 Transfer sofia/internal/1013 at 60.248.175.38 
>  to XML[4001 at default]
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38 
> ) State EXECUTE going to sleep
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_ROUTING
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38 
> ) State ROUTING
> 2010-10-15 16:49:30.666644 [DEBUG] mod_sofia.c:140 sofia/internal/1013 at 60.248.175.38 
>  SOFIA ROUTING
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1013 at 60.248.175.38Standard 
>  ROUTING
> 2010-10-15 16:49:30.666644 [INFO] mod_dialplan_xml.c:331 Processing  
> 1013->4001 in context default
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >unloop] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (PASS) [unloop] $ 
> {unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [unloop] $ 
> {sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >tod_example] continue=true
> Dialplan: sofia/internal/1013 at 60.248.175.38 Date/Time Match (PASS)  
> [tod_example] break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(open=true)
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >holiday_example] continue=true
> Dialplan: sofia/internal/1013 at 60.248.175.38 Date/Time Match (FAIL)  
> [holiday_example] break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->global- 
> intercept] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [global- 
> intercept] destination_number(4001) =~ /^886$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->group- 
> intercept] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [group- 
> intercept] destination_number(4001) =~ /^\*8$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >intercept-ext] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [intercept- 
> ext] destination_number(4001) =~ /^\*\*(\d+)$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >redial] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [redial]  
> destination_number(4001) =~ /^(redial|870)$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >global] continue=true
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [global] $ 
> {call_debug}(false) =~ /^true$/ break=never
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [global] $ 
> {sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32| 
> AES_CM_128_HMAC_SHA1_80)$/ break=never
> Dialplan: sofia/internal/1013 at 60.248.175.38 Absolute Condition  
> [global]
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/$ 
> {domain_name}-spymap/${caller_id_number}/${uuid})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/$ 
> {domain_name}-last_dial/${caller_id_number}/${destination_number})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/$ 
> {domain_name}-last_dial/global/${uuid})
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->snom- 
> demo-2] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [snom- 
> demo-2] destination_number(4001) =~ /^9001$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->snom- 
> demo-1] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [snom- 
> demo-1] destination_number(4001) =~ /^9000$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >eavesdrop] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [eavesdrop]  
> destination_number(4001) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >eavesdrop] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [eavesdrop]  
> destination_number(4001) =~ /^779$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >call_return] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL)  
> [call_return] destination_number(4001) =~ /^\*69$|^869$|^lcr$/  
> break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->del- 
> group] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [del-group]  
> destination_number(4001) =~ /^80(\d{2})$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->add- 
> group] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [add-group]  
> destination_number(4001) =~ /^81(\d{2})$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->call- 
> group-simo] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call-group- 
> simo] destination_number(4001) =~ /^82(\d{2})$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default->call- 
> group-order] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [call-group- 
> order] destination_number(4001) =~ /^83(\d{2})$/ break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >extension-intercom] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (FAIL) [extension- 
> intercom] destination_number(4001) =~ /^8(10[01][0-9])$/ break=on- 
> false
> Dialplan: sofia/internal/1013 at 60.248.175.38 parsing [default- 
> >Local_Extension] continue=false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Regex (PASS)  
> [Local_Extension] destination_number(4001) =~ /^([0-4]0[01][0-9])$/  
> break=on-false
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(RECORD_COPYRIGHT=(c) 2010)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(RECORD_SOFTWARE=FreeSwitch)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(RECORD_ARTIST=FreeSwitch)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(RECORD_COMMENT=FreeSwitch)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(RECORD_DATE=$ 
> {strftime(%Y-%m-%d %H:%M)})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(RECORD_STEREO=true)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(dialed_extension=4001)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> export(dialed_extension=4001)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action set_user($ 
> {dialed_extension}@${domain_name})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> limit(60.248.175.38 ${dialed_extension} ${max_calls} ${fail_over})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action bind_meta_app(1 b  
> s execute_extension::dx XML features)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action bind_meta_app(2 b  
> s record_session::/usr/local/freeswitch/recordings/$ 
> {caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action bind_meta_app(3 b  
> s execute_extension::cf XML features)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action set(ringback=${us- 
> ring})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(transfer_ringback=local_stream://moh)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(call_timeout=30)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(hangup_after_bridge=true)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(continue_on_fail=true)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/$ 
> {domain_name}-call_return/${dialed_extension}/${caller_id_number})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/$ 
> {domain_name}-last_dial_ext/${dialed_extension}/${uuid})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action  
> set(called_party_callgroup=${user_data(${dialed_extension}@$ 
> {domain_name} var callgroup)})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action hash(insert/$ 
> {domain_name}-last_dial/${called_party_callgroup}/${uuid})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action bridge(user/$ 
> {dialed_extension}@${domain_name})
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action answer()
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action sleep(1000)
> Dialplan: sofia/internal/1013 at 60.248.175.38 Action bridge(loopback/ 
> app=voicemail:default ${domain_name} ${dialed_extension})
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1013 at 60.248.175.38 
> ) State Change CS_ROUTING -> CS_EXECUTE
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_session.c:1022 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1013 at 60.248.175.38 
> ) State ROUTING going to sleep
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1013 at 60.248.175.38 
> ) Running State Change CS_EXECUTE
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1013 at 60.248.175.38 
> ) State EXECUTE
> 2010-10-15 16:49:30.666644 [DEBUG] mod_sofia.c:233 sofia/internal/1013 at 60.248.175.38 
>  SOFIA EXECUTE
> 2010-10-15 16:49:30.666644 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1013 at 60.248.175.38Standard 
>  EXECUTE
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(open=true)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [open]=[true]
> EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38- 
> spymap/1013/fe66f80a-992b-43f7-b130-47f0c87f4c15)
> EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38- 
> last_dial/1013/4001)
> EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38- 
> last_dial/global/fe66f80a-992b-43f7-b130-47f0c87f4c15)
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_COPYRIGHT=(c)  
> 2010)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [RECORD_COPYRIGHT]=[(c) 2010]
> EXECUTE sofia/internal/1013 at 60.248.175.38  
> set(RECORD_SOFTWARE=FreeSwitch)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [RECORD_SOFTWARE]=[FreeSwitch]
> EXECUTE sofia/internal/1013 at 60.248.175.38  
> set(RECORD_ARTIST=FreeSwitch)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [RECORD_ARTIST]=[FreeSwitch]
> EXECUTE sofia/internal/1013 at 60.248.175.38  
> set(RECORD_COMMENT=FreeSwitch)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [RECORD_COMMENT]=[FreeSwitch]
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_DATE=2010-10-15  
> 16:49)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [RECORD_DATE]=[2010-10-15 16:49]
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(RECORD_STEREO=true)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [RECORD_STEREO]=[true]
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(dialed_extension=4001)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [dialed_extension]=[4001]
> EXECUTE sofia/internal/1013 at 60.248.175.38  
> export(dialed_extension=4001)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:901 EXPORT  
> [dialed_extension]=[4001]
> EXECUTE sofia/internal/1013 at 60.248.175.38 set_user(4001 at 60.248.175.38)
> EXECUTE sofia/internal/1013 at 60.248.175.38 limit(60.248.175.38 4001  )
> 2010-10-15 16:49:30.666644 [WARNING] mod_limit.c:779 USAGE: limit  
> <realm> <id> [<max> [number  [dialplan [context]]]]
> EXECUTE sofia/internal/1013 at 60.248.175.38 bind_meta_app(1 b s  
> execute_extension::dx XML features)
> 2010-10-15 16:49:30.666644 [INFO] switch_ivr_async.c:2372 Bound B- 
> Leg: 1 execute_extension::dx XML features
> EXECUTE sofia/internal/1013 at 60.248.175.38 bind_meta_app(2 b s  
> record_session::/usr/local/freeswitch/recordings/ 
> 1013.2010-10-15-16-49-30.wav)
> 2010-10-15 16:49:30.666644 [INFO] switch_ivr_async.c:2372 Bound B- 
> Leg: 2 record_session::/usr/local/freeswitch/recordings/ 
> 1013.2010-10-15-16-49-30.wav
> EXECUTE sofia/internal/1013 at 60.248.175.38 bind_meta_app(3 b s  
> execute_extension::cf XML features)
> 2010-10-15 16:49:30.666644 [INFO] switch_ivr_async.c:2372 Bound B- 
> Leg: 3 execute_extension::cf XML features
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(ringback= 
> %(2000,4000,440.0,480.0))
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [ringback]=[%(2000,4000,440.0,480.0)]
> EXECUTE sofia/internal/1013 at 60.248.175.38  
> set(transfer_ringback=local_stream://moh)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [transfer_ringback]=[local_stream://moh]
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(call_timeout=30)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [call_timeout]=[30]
> EXECUTE sofia/internal/1013 at 60.248.175.38  
> set(hangup_after_bridge=true)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [hangup_after_bridge]=[true]
> EXECUTE sofia/internal/1013 at 60.248.175.38 set(continue_on_fail=true)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [continue_on_fail]=[true]
> EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38- 
> call_return/4001/1013)
> EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38- 
> last_dial_ext/4001/fe66f80a-992b-43f7-b130-47f0c87f4c15)
> EXECUTE sofia/internal/1013 at 60.248.175.38  
> set(called_party_callgroup=techsupport)
> 2010-10-15 16:49:30.666644 [DEBUG] mod_dptools.c:817 sofia/internal/1013 at 60.248.175.38 
>  SET [called_party_callgroup]=[techsupport]
> EXECUTE sofia/internal/1013 at 60.248.175.38 hash(insert/60.248.175.38- 
> last_dial/techsupport/fe66f80a-992b-43f7-b130-47f0c87f4c15)
> EXECUTE sofia/internal/1013 at 60.248.175.38 bridge(user/4001 at 60.248.175.38 
> )
> 2010-10-15 16:49:30.666644 [DEBUG] switch_ivr_originate.c:1946  
> variable string 0 = [presence_id=4001 at 60.248.175.38]
> 2010-10-15 16:49:30.666644 [ERR] switch_ivr_originate.c:2493 Cannot  
> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> 2010-10-15 16:49:30.666644 [DEBUG] switch_ivr_originate.c:3308  
> Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
> 2010-10-15 16:49:30.666644 [ERR] switch_ivr_originate.c:2493 Cannot  
> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> 2010-10-15 16:49:30.666644 [DEBUG] switch_ivr_originate.c:3308  
> Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
> 2010-10-15 16:49:30.666644 [INFO] mod_dptools.c:2356 Originate  
> Failed.  Cause: USER_NOT_REGISTERED
> EXECUTE sofia/internal/1013 at 60.248.175.38 answer()
> 2010-10-15 16:49:30.694130 [DEBUG] sofia_glue.c:2616 AUDIO RTP [sofia/internal/1013 at 60.248.175.38 
> ] 60.248.175.38 port 25606 -> 60.248.175.43 port 35070 codec: 0 ms: 20
> 2010-10-15 16:49:30.694130 [DEBUG] switch_rtp.c:1343 Starting timer  
> [soft] 160 bytes per 20ms
> 2010-10-15 16:49:30.695467 [DEBUG] sofia_glue.c:2822 Set 2833 dtmf  
> send payload to 101
> 2010-10-15 16:49:30.695467 [DEBUG] sofia_glue.c:2827 Set 2833 dtmf  
> receive payload to 101
> 2010-10-15 16:49:30.695467 [DEBUG] mod_sofia.c:663 Local SDP sofia/internal/1013 at 60.248.175.38 
> :
> v=0
> o=FreeSWITCH 1287106964 1287106965 IN IP4 60.248.175.38
> s=FreeSWITCH
> c=IN IP4 60.248.175.38
> t=0 0
> m=audio 25606 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> 2010-10-15 16:49:30.695467 [DEBUG] switch_core_session.c:642 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-15 16:49:30.695467 [NOTICE] mod_dptools.c:720 Channel [sofia/internal/1013 at 60.248.175.38 
> ] has been answered
> EXECUTE sofia/internal/1013 at 60.248.175.38 sleep(1000)
> 2010-10-15 16:49:30.698172 [DEBUG] sofia.c:4200 Channel sofia/internal/1013 at 60.248.175.38 
>  entering state [completed][200]
> 2010-10-15 16:49:30.745378 [DEBUG] switch_rtp.c:2443 Correct ip/port  
> confirmed.
> 2010-10-15 16:49:30.807388 [DEBUG] sofia.c:4200 Channel sofia/internal/1013 at 60.248.175.38 
>  entering state [ready][200]
> EXECUTE sofia/internal/1013 at 60.248.175.38 bridge(loopback/ 
> app=voicemail:default 60.248.175.38 4001)
> 2010-10-15 16:49:31.704837 [NOTICE] switch_channel.c:675 New Channel  
> loopback/app=voicemail:default 60.248.175.38 4001-a [23f3d5de- 
> f985-44bd-8b1a-118b98bb5eae]
> 2010-10-15 16:49:31.704837 [DEBUG] mod_loopback.c:127 loopback/ 
> app=voicemail:default 60.248.175.38 4001-a setup codec PCMU/8000/20
> 2010-10-15 16:49:31.704837 [NOTICE] switch_channel.c:673 Rename  
> Channel loopback/app=voicemail:default 60.248.175.38 4001-a- 
> >loopback/voicemail-a [23f3d5de-f985-44bd-8b1a-118b98bb5eae]
> 2010-10-15 16:49:31.704837 [DEBUG] mod_loopback.c:911 (loopback/ 
> voicemail-a) State Change CS_NEW -> CS_INIT
> 2010-10-15 16:49:31.704837 [DEBUG] switch_core_session.c:1022 Send  
> signal loopback/voicemail-a [BREAK]
> 2010-10-15 16:49:31.704837 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-a CHANNEL KILL
> 2010-10-15 16:49:31.706218 [DEBUG] switch_core_state_machine.c:314  
> (loopback/voicemail-a) Running State Change CS_INIT
> 2010-10-15 16:49:31.706218 [DEBUG] switch_core_state_machine.c:338  
> (loopback/voicemail-a) State INIT
> 2010-10-15 16:49:31.706218 [NOTICE] switch_channel.c:675 New Channel  
> loopback/voicemail-b [1abc53a2-df93-40db-96ab-547f9ddd1b1d]
> 2010-10-15 16:49:31.706218 [DEBUG] mod_loopback.c:127 loopback/ 
> voicemail-b setup codec PCMU/8000/20
> 2010-10-15 16:49:31.706218 [DEBUG] mod_loopback.c:240 (loopback/ 
> voicemail-b) State Change CS_NEW -> CS_INIT
> 2010-10-15 16:49:31.706218 [DEBUG] switch_core_session.c:1022 Send  
> signal loopback/voicemail-b [BREAK]
> 2010-10-15 16:49:31.706218 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-b CHANNEL KILL
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:314  
> (loopback/voicemail-b) Running State Change CS_INIT
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:338  
> (loopback/voicemail-b) State INIT
> 2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:281 (loopback/ 
> voicemail-b) State Change CS_INIT -> CS_ROUTING
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_session.c:1022 Send  
> signal loopback/voicemail-b [BREAK]
> 2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-b CHANNEL KILL
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:338  
> (loopback/voicemail-b) State INIT going to sleep
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:314  
> (loopback/voicemail-b) Running State Change CS_ROUTING
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:341  
> (loopback/voicemail-b) State ROUTING
> 2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:315 loopback/ 
> voicemail-b CHANNEL ROUTING
> 2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:326 (loopback/ 
> voicemail-b) State Change CS_ROUTING -> CS_EXECUTE
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_session.c:1022 Send  
> signal loopback/voicemail-b [BREAK]
> 2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-b CHANNEL KILL
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:341  
> (loopback/voicemail-b) State ROUTING going to sleep
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:314  
> (loopback/voicemail-b) Running State Change CS_EXECUTE
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:348  
> (loopback/voicemail-b) State EXECUTE
> 2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:346 loopback/ 
> voicemail-b CHANNEL EXECUTE
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_state_machine.c:157  
> loopback/voicemail-b Standard EXECUTE
> EXECUTE loopback/voicemail-b pre_answer()
> 2010-10-15 16:49:31.707132 [NOTICE] mod_loopback.c:716 Pre-Answer  
> loopback/voicemail-a!
> 2010-10-15 16:49:31.707132 [DEBUG] switch_channel.c:2254 Send signal sofia/internal/1013 at 60.248.175.38 
> [BREAK]
> 2010-10-15 16:49:31.707132 [DEBUG] switch_core_session.c:642 Send  
> signal loopback/voicemail-b [BREAK]
> 2010-10-15 16:49:31.707132 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-b CHANNEL KILL
> 2010-10-15 16:49:31.707132 [NOTICE] mod_dptools.c:746 Pre-Answer  
> loopback/voicemail-b!
> 2010-10-15 16:49:31.707132 [DEBUG] switch_channel.c:2254 Send signal sofia/internal/1013 at 60.248.175.38 
> [BREAK]
> EXECUTE loopback/voicemail-b voicemail(default 60.248.175.38 4001)
> 2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:281 (loopback/ 
> voicemail-a) State Change CS_INIT -> CS_ROUTING
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_session.c:1022 Send  
> signal loopback/voicemail-a [BREAK]
> 2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-a CHANNEL KILL
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:338  
> (loopback/voicemail-a) State INIT going to sleep
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:314  
> (loopback/voicemail-a) Running State Change CS_ROUTING
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:341  
> (loopback/voicemail-a) State ROUTING
> 2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:315 loopback/ 
> voicemail-a CHANNEL ROUTING
> 2010-10-15 16:49:31.708787 [DEBUG] switch_ivr_originate.c:66  
> (loopback/voicemail-a) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_session.c:1022 Send  
> signal loopback/voicemail-a [BREAK]
> 2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-a CHANNEL KILL
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:341  
> (loopback/voicemail-a) State ROUTING going to sleep
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:314  
> (loopback/voicemail-a) Running State Change CS_CONSUME_MEDIA
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:360  
> (loopback/voicemail-a) State CONSUME_MEDIA
> 2010-10-15 16:49:31.708787 [DEBUG] mod_loopback.c:512 CHANNEL  
> CONSUME_MEDIA
> 2010-10-15 16:49:31.708787 [DEBUG] switch_core_state_machine.c:360  
> (loopback/voicemail-a) State CONSUME_MEDIA going to sleep
> 2010-10-15 16:49:31.710698 [DEBUG] switch_ivr_originate.c:3228  
> Originate Resulted in Success: [loopback/voicemail-a]
> 2010-10-15 16:49:31.710698 [DEBUG] switch_core_session.c:642 Send  
> signal loopback/voicemail-a [BREAK]
> 2010-10-15 16:49:31.710698 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-a CHANNEL KILL
> 2010-10-15 16:49:31.710698 [DEBUG] switch_core_session.c:642 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-15 16:49:31.710698 [DEBUG] switch_ivr_bridge.c:1182  
> (loopback/voicemail-a) State Change CS_CONSUME_MEDIA ->  
> CS_EXCHANGE_MEDIA
> 2010-10-15 16:49:31.710698 [DEBUG] switch_core_session.c:1022 Send  
> signal loopback/voicemail-a [BREAK]
> 2010-10-15 16:49:31.710698 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-a CHANNEL KILL
> 2010-10-15 16:49:31.710698 [DEBUG] switch_core_state_machine.c:314  
> (loopback/voicemail-a) Running State Change CS_EXCHANGE_MEDIA
> 2010-10-15 16:49:31.710698 [DEBUG] switch_core_state_machine.c:351  
> (loopback/voicemail-a) State EXCHANGE_MEDIA
> 2010-10-15 16:49:31.710698 [DEBUG] mod_loopback.c:474 CHANNEL LOOPBACK
> 2010-10-15 16:49:31.712661 [DEBUG] switch_core_session.c:703 Send  
> signal loopback/voicemail-a [BREAK]
> 2010-10-15 16:49:31.712661 [DEBUG] mod_loopback.c:452 loopback/ 
> voicemail-a CHANNEL KILL
> 2010-10-15 16:49:31.712661 [DEBUG] switch_core_session.c:703 Send signalsofia/internal/1013 at 60.248.175.38 
>  [BREAK]
> 2010-10-15 16:49:31.824953 [DEBUG] switch_ivr_play_say.c:63 No  
> language specified - Using [en]
> 2010-10-15 16:49:31.824953 [DEBUG] switch_ivr_play_say.c:244 Handle  
> play-file:[voicemail/vm-person.wav] (en:en)
> 2010-10-15 16:49:31.824953 [DEBUG] switch_ivr_play_say.c:1152 Codec  
> Activated L16 at 8000hz 1 channels 20ms
> 2010-10-15 16:49:33.185370 [DEBUG] switch_ivr_play_say.c:1444 done  
> playing file
> 2010-10-15 16:49:33.305429 [DEBUG] switch_ivr_play_say.c:244 Handle  
> say:[4001] (en:en)
> 2010-10-15 16:49:33.305429 [DEBUG] switch_ivr_play_say.c:1152 Codec  
> Activated L16 at 8000hz 1 channels 20ms
> 2010-10-15 16:49:33.744961 [DEBUG] switch_ivr_play_say.c:1444 done  
> playing file
> 2010-10-15 16:49:33.744961 [DEBUG] switch_ivr_play_say.c:1152 Codec  
> Activated L16 at 8000hz 1 channels 20ms
>
>
> From: david.ponzone at ipeva.fr
> To: freeswitch-users at lists.freeswitch.org
> Date: Thu, 14 Oct 2010 12:20:56 +0200
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> Don't be shy on the logs, send the full ones!
>
> in fs_cli:
> fsctl loglevel debug
> /log 7
>
> and then make a call
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
> message, merci de le d?truire imm?diatement et d'avertir  
> l'exp?diteur.
>
>
>
>
> Le 14/10/2010 ? 11:53, ? Chiang ?? Chih-Chung Wybie a ?crit :
>
> I tried use 1013 at 38 calls to 1019 at 37
>
> show log below
>
> 2010-10-14 17:50:41.249876 [NOTICE] switch_channel.c:675 New Channel sofia/internal/1013 at 60.248.175.38 
> [875fdeef-b94e-41fb-a621-ea005bbaedbd]
> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing  
> 1013->1019 in context public
> 2010-10-14 17:50:41.263809 [NOTICE] switch_ivr.c:1450 Transfer sofia/internal/1013 at 60.248.175.38 
>  to XML[1019 at default]
> 2010-10-14 17:50:41.263809 [INFO] mod_dialplan_xml.c:331 Processing  
> 1013->1019 in context default
> 2010-10-14 17:50:41.283904 [INFO] switch_core_session.c:1764 Sending  
> early media
> 2010-10-14 17:50:41.285023 [NOTICE] mod_sofia.c:1994 Pre-Answer sofia/internal/1013 at 60.248.175.38 
> !
>
> seems not work
>
> Thank you for you attension
> Best Regards
>
> Gary
>
>
>
> From: david.ponzone at ipeva.fr
> To: freeswitch-users at lists.freeswitch.org
> Date: Thu, 14 Oct 2010 10:16:28 +0200
> Subject: Re: [Freeswitch-users] some questions of freeswitch & X-Lite
>
> I think we would need a complete log from the target FS, in order to  
> provide you with any efficient help.
>
> Thank
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
> message, merci de le d?truire imm?diatement et d'avertir  
> l'exp?diteur.
>
>
>
>
> Le 14/10/2010 ? 09:55, ? Chiang ?? Chih-Chung Wybie a ?crit :
>
> I tried Connect Two FreeSWITCH Boxes  http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes
> and my configuration :
>
> 60.248.175.37
>
> (autoload_configs/act.conf.xml)
> <list name="domains" default="deny">
>       <node type="allow" cidr="60.248.175.38/28"/>
>      <!-- <node type="allow" domain="$${domain}"/>-->
> </list>
>
> (dialplan/default.xml)
>
> <extension name="Dial to 38">
>       <condition field="destination_number" expression="^([0-4]0[01] 
> [0-9])$">
>         <action application="bridge" data="sofia/internal/$1 at 60.248.175.38 
> "/>
>       </condition>
>  </extension>
>
> (dialplan/public.xml)
>
> <extension name="Calls from 38">
>       <condition field="destination_number" expression="^([0-4]0[01] 
> [0-9])$">
>         <action application="transfer" data="$1 XML default"/>
>       </condition>
>  </extension>
>
>
>
> 60.248.175.38
>
> (autoload_configs/act.conf.xml)
> <list name="domains" default="deny">
>       <node type="allow" cidr="60.248.175.37/28"/>
>      <!-- <node type="allow" domain="$${domain}"/>-->
> </list>
>
> (dialplan/default.xml)
>
> <extension name="Dial to 37">
>       <condition field="destination_number" expression="^([0-4]0[01] 
> [0-9])$">
>         <action application="bridge" data="sofia/internal/$1 at 60.248.175.37 
> "/>
>       </condition>
>  </extension>
>
> (dialplan/public.xml)
>
> <extension name="Calls from 37">
>       <condition field="destination_number" expression="^([0-4]0[01] 
> [0-9])$">
>         <action application="transfer" data="$1 XML default"/>
>       </condition>
>  </extension>
>
> when I use 1007 at 38 calls to 1007 at 37
> it won't show errors but still not work
>
> how should I change something ?
>
> Thank you for your attension
>
> Best Regards
> Gary
>
>
>
>
>
> > Date: Wed, 13 Oct 2010 22:17:13 -0700
> > From: curriegrad2004 at gmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] some questions of freeswitch & X- 
> Lite
> >
> > You can create a SIP trunk between the two servers so calls can be
> > routed back and forth from the servers you've specified.
> >
> > 2010/10/13 ? Chiang ?? Chih-Chung Wybie <tear152 at hotmail.com>:
> > > I tried it
> > > but not working
> > > and show errors below
> > > 2010-10-14 11:03:05.336206 [ERR] switch_ivr_originate.c:2493  
> Cannot create
> > > outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> > > 2010-10-14 11:03:05.338283 [ERR] switch_ivr_originate.c:2493  
> Cannot create
> > > outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> > > 2010-10-14 11:03:05.338283 [INFO] mod_dptools.c:2356 Originate  
> Failed.
> > > Cause: USER_NOT_REGISTERED
> > >
> > > and for example
> > >  I use SIP server 60.248.175.38  user id=1007
> > > and I call to 1007 at 60.248.175.37    that will show these errors
> > >
> > > how should set more?
> > >
> > > thank you for your attension
> > >
> > > Best Regards
> > > Gary
> > >
> > > ________________________________
> > > Date: Thu, 14 Oct 2010 10:55:49 +0800
> > > From: xyangni at gmail.com
> > > To: freeswitch-users at lists.freeswitch.org
> > > Subject: Re: [Freeswitch-users] some questions of freeswitch & X- 
> Lite
> > >
> > > I think you can reg at server A and than call B as a normal SIP  
> call to
> > > external domain.
> > >
> > > 2010/10/14 ? Chiang ?? Chih-Chung Wybie  
> <tear152 at hotmail.com>
> > >
> > >
> > >
> > > Hello
> > >
> > > I set two SIP servers  60.248.175.37 & 60.248.175.38 and use X- 
> Lite to call
> > >
> > > my question is how can I call between these servers?
> > >
> > > if I can , how shoould I set in *.xml? or X-Lite?
> > >
> > > thank you for your attension
> > >
> > >
> > > Best Regards
> > > Gary
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> > >
> > > _______________________________________________ FreeSWITCH-users  
> mailing
> > > list FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________ FreeSWITCH-users  
> mailing listFreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
>  http:// 
> www.freeswitch.org_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________ FreeSWITCH-users  
> mailing listFreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
>  http:// 
> www.freeswitch.org_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________ FreeSWITCH-users  
> mailing listFreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
>  http:// 
> www.freeswitch.org_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________ FreeSWITCH-users  
> mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
>  http:// 
> www.freeswitch.org_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From b0ef at esben-stien.name  Mon Oct 18 03:44:27 2010
From: b0ef at esben-stien.name (Esben Stien)
Date: Mon, 18 Oct 2010 12:44:27 +0200
Subject: [Freeswitch-users] Freeswitch Video Capture and Playback
In-Reply-To: <AANLkTimtQsE7+=TQ4eedswLNy7my223z=rEtrjXiFqV+@mail.gmail.com>
	(Michael Collins's message of "Thu\,
	14 Oct 2010 17\:58\:47 -0700")
References: <87k4xlga1k.fsf@quasar.esben-stien.name>
	<87pr6yz5wa.fsf@quasar.esben-stien.name>
	<87bp6we05a.fsf@quasar.esben-stien.name>
	<AANLkTimtQsE7+=TQ4eedswLNy7my223z=rEtrjXiFqV+@mail.gmail.com>
Message-ID: <87aambaigk.fsf@quasar.esben-stien.name>

Michael Collins <msc at freeswitch.org> writes:

> Brian informs me that there may be a bug with this. I will confer with
> Tony tomorrow and find out.

Found out anything?.

-- 
Esben Stien is b0ef at e     s      a             
         http://www. s     t    n m
          irc://irc.  b  -  i  .   e/%23contact
           sip:b0ef@   e     e 
           jid:b0ef@    n     n


From dujinfang at gmail.com  Mon Oct 18 03:40:07 2010
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 18 Oct 2010 18:40:07 +0800
Subject: [Freeswitch-users] mod_erlang_event core dump on load with
 shortname=false on Mac
In-Reply-To: <20101017164257.GK15964@hijacked.us>
References: <AANLkTikToP3n7rGHO3DgaW3C1cK-XpN7WcmzNdzd8KHr@mail.gmail.com>
	<20101017164257.GK15964@hijacked.us>
Message-ID: <AANLkTinV_u076tHcGkhW7h1st_R2WwsN-ppJ8LZJbtTp@mail.gmail.com>

Fixed by Andrew.

On Mon, Oct 18, 2010 at 12:42 AM, Andrew Thompson <andrew at hijacked.us> wrote:
> Or just email me privately the information. I've been trying to rework
> that code to match how erlang infers long names (because the ones I
> inferred weren't matching erlang's).
>
> The email or the JIRA should include the backtrace, the output of
> 'hostname', 'domainname', and /etc/resolv.conf. Also, what erlang on the
> box guesses for short/long names.
>
> Andrew
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From covici at ccs.covici.com  Mon Oct 18 05:17:59 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 18 Oct 2010 08:17:59 -0400
Subject: [Freeswitch-users] problem when recording conferences
Message-ID: <26893.1287404279@ccs.covici.com>

Hi.  I am having a problem with conference recordings using
auto_record.  Now it does record, but with some users if the user is
silent for a while, then the first so many milliseconds of his first
word cuts off.  Is this a function of the energe setting, waste setting
or how else can I prevent this?

Thanks in advance for any suggestionns.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From bottleman at icf.org.ru  Mon Oct 18 05:51:44 2010
From: bottleman at icf.org.ru (Georgiewskiy Yuriy)
Date: Mon, 18 Oct 2010 16:51:44 +0400 (MSD)
Subject: [Freeswitch-users] Mod_h323 problem
In-Reply-To: <1286972082.1495.19.camel@Zaphod.gateway.2wire.net>
References: <1286972082.1495.19.camel@Zaphod.gateway.2wire.net>
Message-ID: <alpine.DEB.2.00.1010181651210.17271@icf.org.ru>

On 2010-10-13 13:14 +0100, John Carpenter wrote FreeSWITCH Users Help:

JC>Hi, I am trying to bridge an incoming h323 call to an external SIP
JC>provider, I am using latest git release. If I use G729 codec I get
JC>message
JC>
JC>"Unsupported ptime of 2 on  write Audio codec G.729{sw}  for connection
JC>[0xb4c0ee10]"
JC>
JC>and call fails to connect. If I use ulaw codec call fails to connect
JC>with
JC>
JC>"Write PDU fail: no control channel"
JC>
JC>If I make a straight SIP to SIP call though same provider all work ok. I
JC>have posted log of call in http://pastebin.freeswitch.org/14216 because
JC>it is rather large.
JC>
JC>my h323.conf.xml file looks like this
JC>
JC><configuration name="h323.conf" description="H323 Endpoints">
JC> <settings>
JC>   <param name="trace-level" value="10"/>
JC>   <param name="context" value="public"/>
JC>   <param name="dialplan" value="XML"/>
JC>   <param name="codec-prefs" value="PCMA,PCMU,GSM,G729"/>
JC>   <param name="use-rtp-timer" value="true"/> <!-- enable RTP timer -
JC>should always be enabled -->
JC>   <param name="rtp-timer-name" value="soft"/> <!-- Timer name, soft is
JC>default -->
JC>   <!-- <param name="ptime-override-value" value="20"/> --> <!--
JC>Override negotiated ptime value with this value -->
JC>   <param name="gk-address" value=""/>    <!-- empty to disable, "*" to
JC>search LAN -->
JC>   <param name="gk-identifer" value=""/>  <!-- optional name of gk -->
JC>   <param name="gk-interface" value=""/>  <!-- mandatory listener
JC>interface name -->
JC>   <param name="gk-retry" value="30"/>  <!-- optional GK register retry
JC>timer -->
JC>   <param name="faststart" value="true"/>  <!-- optional -->
JC>   <param name="h245tunneling" value="true"/>  <!-- optional -->
JC>   <param name="h245insetup" value="true"/>  <!-- optional -->
JC>   <param name="jitter-size" value="60"/>  <!-- optional -->
JC>   <param name="progress-indication" value="8"/> <!-- optional - PI
JC>value in progress message-->
JC>   <param name="alerting-indication" value="8"/> <!-- optional - PI
JC>value in alerting message-->
JC>   <param name="endpoint-name" value="fs"/>
JC>   <param name="fax-old-asn" value="true"/>
JC> </settings>
JC> <listeners>
JC>   <listener name="default">
JC>     <param name="h323-ip" value="$${local_ip_v4}"/>
JC>     <param name="h323-port" value="1720"/>
JC>   </listener>
JC> </listeners>
JC></configuration>
JC>
JC>And this is the log of the xml_curl dialplan that is executed
JC>
JC><?xml version="1.0" encoding="UTF-8" standalone="no"?>
JC><document type="freeswitch/xml">
JC><section name="dialplan" description="php dialplan">
JC><context name="public">
JC><extension name="normal">
JC><condition field="destination_number" expression="^(\d+)$">
JC><action application="set" data="bypass_media=false"/>
JC><action application="set" data="proxy_media=true"/>
JC><action application="bridge" data="sofia/gateway/arbinet-o/$1"/>
JC></condition>
JC></extension>
JC></context>
JC></section>
JC></document>
JC>
JC>Any help will be greatly appreciated 
JC>
JC>regards, John Carpenter
JC>

mod_h323 don't work with proxy_media.

C ?????????                       With Best Regards
???????????? ????.                Georgiewskiy Yuriy
+7 4872 711666                    +7 4872 711666
???? +7 4872 711143               fax +7 4872 711143
???????? ??? "?? ?? ??????"       IT Service Ltd
http://nkoort.ru                  http://nkoort.ru
JID: GHhost at jabber.tula-ix.net.ru JID: GHhost at jabber.tula-ix.net.ru
YG129-RIPE                        YG129-RIPE

From Holger.Esser at Convergys.com  Mon Oct 18 06:21:59 2010
From: Holger.Esser at Convergys.com (Esser, Holger)
Date: Mon, 18 Oct 2010 08:21:59 -0500
Subject: [Freeswitch-users] mod_cepstral 5.1 known issues?
In-Reply-To: <4CBB4126.9020208@solomo.de>
References: <AANLkTik8KZdxAnUUfv0nKF6EkTHe559PtC2bF74eng11@mail.gmail.com>
	<AANLkTi=MvWh14MQ9wPEboXZmMgAtURAY5zBoobn32Pkq@mail.gmail.com>
	<AANLkTimj=SwX7d1sot+NhOxMgD4F-PN=bM3hxrh3bF3O@mail.gmail.com>
	<4CB7F0C9.2040708@solomo.de>
	<0FF607C13B7F7A42B5B80DA9EB204C9109D44BDD8E@srv-ex01-dal.intervoice.int>
	<AANLkTi=cAW7Ru3KPy4LH7bwy0eGKce0q5z-yztF-Nhd2@mail.gmail.com>
	<4CBB4126.9020208@solomo.de>
Message-ID: <0FF607C13B7F7A42B5B80DA9EB204C9109D65B954F@srv-ex01-dal.intervoice.int>

Many thanks guys, this is what I was looking for.

Holger

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Daniel Neubert
Sent: Sunday, October 17, 2010 1:32 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_cepstral 5.1 known issues?

I'm sure it will ;-) This one is from me and the answers given / linked in this jira ticket helped me a lot.

Best regards / Mit freundlichen Gr??en,
Daniel Neubert

Am 16.10.2010 01:22, schrieb Marc Kellem:
Holger, this JIRA issue might answer some of your questions.

http://jira.freeswitch.org/browse/FS-10



On Fri, Oct 15, 2010 at 7:32 AM, Esser, Holger <Holger.Esser at convergys.com<mailto:Holger.Esser at convergys.com>> wrote:
Hi Daniel,

May I ask which guide you used to set it up?

Thanks,
Holger

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Daniel Neubert
Sent: Friday, October 15, 2010 1:12 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_cepstral 5.1 known issues?

I've exactly this setup which is in productive use for a few months now.

Using cepstral directly on our FreeSWITCH nodes did not work (since they are running on FreeBSD and Cepstral is only available for GNU Linux).

Current setup is using Voice Katrin (German) Version 5.1.0 on Gentoo Linux 64Bit via UniMRCP Server 1.0.0.
Best regards / Mit freundlichen Gr??en,
Daniel Neubert

On 14.10.2010 17:58, Marc Kellem wrote:
Is anyone using mod_unimrcp to access Cepstral 5.1 voices in a UniMRCP server?

On Thu, Oct 14, 2010 at 11:26 AM, Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>> wrote:
random deadlocks in the cepstral engine blocking all the speech
generation calls.


On Thu, Oct 14, 2010 at 9:56 AM, Marc Kellem <mkellem at vontoo.com<mailto:mkellem at vontoo.com>> wrote:
> The mod_cepstral wiki page says "Warning: the suggested version to use
> is 4.x since there are known issues with 5.1 (which is closed source)".
>  What exactly are the known issues?
> http://wiki.freeswitch.org/wiki/Mod_cepstral
> Thanks,
> Marc Kellem
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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<mailto:MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


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From nico at clickfono.com  Mon Oct 18 06:33:10 2010
From: nico at clickfono.com (Nicolas Brenner)
Date: Mon, 18 Oct 2010 10:33:10 -0300
Subject: [Freeswitch-users] Binding to virtual interface (ipsec vpn)
In-Reply-To: <AANLkTimqaPs0hPaeG_QtPoX1f06ETRM4Jj4X3mOaMSxk@mail.gmail.com>
References: <AANLkTi=u6VWM-ejjjFPPws9-ZMTR+KxdRHLTBJ+5O6Jh@mail.gmail.com>
	<20101011213354.C82E62C36FD@cuneorg-email.cune.pri>
	<AANLkTinFuqDHbU1yLhtafSv4P6XJGG=eRpiHuR6XLEDB@mail.gmail.com>
	<AANLkTinseQ7UEp2u+QRTxuGV6fGqwwaii1-ppT_pB__P@mail.gmail.com>
	<AANLkTimqaPs0hPaeG_QtPoX1f06ETRM4Jj4X3mOaMSxk@mail.gmail.com>
Message-ID: <AANLkTinTpkvMJcNQEXuBD9W+9YqBp8w_XoKJvSCieu4=@mail.gmail.com>

I haven't been able to solve this, but one thing I noticed though, is that
the internal profile is being aliased to the ip of the interface I'm trying
to use in the external profile, I'm guessing this causes trouble, the
problem is that if I disable aliasing, it doesn't make a difference, and
checking out the list archive, apparently is a bad idea to disable aliasing
anyway.

How do I prevent the internal profile from aliasing to a specific ip? and
what ip should the internal profile should alias to if the server has to
private ("internal") ip?

Thanks.



On Mon, Oct 11, 2010 at 7:21 PM, Nicolas Brenner <nico at clickfono.com> wrote:

> Yup, here's the output:
>
> http://pastebin.freeswitch.org/14203
>
>
> I'm trying to originate a call using the gateway on 200.13.15.220,
> originating it from 172.26.6.161.
>
> Seems like no packets are going out at all, at least tcpdump is not getting
> any. Oddly, if instead of setting sip-ip and rtp-ip to the ip of the
> interface I need, I set ext-sip-ip and ext-rtp-ip to that ip, tcpdump starts
> seeing traffic, but the traffic is originated from the wrong ip/interface
> (the public one, instead of the one with the ipsec vpn).
>
>
>
> On Mon, Oct 11, 2010 at 6:50 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> When you do "sofia status" do you see the correct IP address for the
>> profile in question?
>> Also, when you make an outbound call can you see where (if?) the packets
>> are going? Perhaps to another interface?
>>
>> -MC
>>
>>
>> On Mon, Oct 11, 2010 at 2:44 PM, Nicolas Brenner <nico at clickfono.com>wrote:
>>
>>> The system routes are well configured, I can ping and telnet the gateway
>>> from bash, all traffic to that gateway's ip is routed through the needed
>>> interface. Now, are there other routes I should be addressing in FS config
>>> files?
>>>
>>>
>>> On Mon, Oct 11, 2010 at 6:33 PM, <Russell.Mosemann at cune.org> wrote:
>>>
>>>> Another thing to check is that there is only one IP address assigned to
>>>> the interface. A problem with using the "right" IP address can occur if
>>>> there is more than one IP address assigned to the same interface.
>>>> Without
>>>> routing rules, the operating system chooses the IP address from the
>>>> interface that it believes is the best one for delivering the traffic,
>>>> and sometimes it is not the address you want to use.
>>>>
>>>> --
>>>> Russell Mosemann
>>>>
>>>>
>>>>
>>>> ________________________________________________________
>>>> Concordia University, Nebraska
>>>> See http://www.cune.edu/ for the latest news and events!
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From nico at clickfono.com  Mon Oct 18 07:20:52 2010
From: nico at clickfono.com (Nicolas Brenner)
Date: Mon, 18 Oct 2010 11:20:52 -0300
Subject: [Freeswitch-users] Binding to virtual interface (ipsec vpn)
In-Reply-To: <AANLkTinTpkvMJcNQEXuBD9W+9YqBp8w_XoKJvSCieu4=@mail.gmail.com>
References: <AANLkTi=u6VWM-ejjjFPPws9-ZMTR+KxdRHLTBJ+5O6Jh@mail.gmail.com>
	<20101011213354.C82E62C36FD@cuneorg-email.cune.pri>
	<AANLkTinFuqDHbU1yLhtafSv4P6XJGG=eRpiHuR6XLEDB@mail.gmail.com>
	<AANLkTinseQ7UEp2u+QRTxuGV6fGqwwaii1-ppT_pB__P@mail.gmail.com>
	<AANLkTimqaPs0hPaeG_QtPoX1f06ETRM4Jj4X3mOaMSxk@mail.gmail.com>
	<AANLkTinTpkvMJcNQEXuBD9W+9YqBp8w_XoKJvSCieu4=@mail.gmail.com>
Message-ID: <AANLkTinKaps=d7DCg3vR4MJQwvt1NBv11c4Zq9mjCUr8@mail.gmail.com>

I tried a few more things and discovered that apparently the sip profile
configuration was fine, but the problem has something to do with the
gateway, because when I try originating a call through another gateway, it
generates traffic fine, it doesn't work because is not on the right network
interface, but at least it tries.

This is the configuration of the gateway that is not working (although it
appears fine with sofia status on the console)

<gateway name="somegateway">
        <param name="username" value="someuser"/>
        <param name="password" value="somepass"/>
        <param name="from-domain" value="200.13.15.220:5061"/>
        <param name="realm" value="200.13.15.220:5061"/>
        <param name="proxy" value="200.13.15.220:5061"/>
        <param name="from-user" value="someuser"/>
        <param name="caller-id-in-from" value="false"/>
        <param name="expire-seconds" value="600"/>
        <param name="register" value="false"/>
        <param name="retry_seconds" value="5"/>
        <param name="extension" value="2011"/>
        <param name="context" value="public"/>
        <param name="codec-prefs" value="G729"/>
        <param name="rfc2833-pt" value="101"/>
      </gateway>


Could it be because of the port (5061)?




On Mon, Oct 18, 2010 at 10:33 AM, Nicolas Brenner <nico at clickfono.com>wrote:

> I haven't been able to solve this, but one thing I noticed though, is that
> the internal profile is being aliased to the ip of the interface I'm trying
> to use in the external profile, I'm guessing this causes trouble, the
> problem is that if I disable aliasing, it doesn't make a difference, and
> checking out the list archive, apparently is a bad idea to disable aliasing
> anyway.
>
> How do I prevent the internal profile from aliasing to a specific ip? and
> what ip should the internal profile should alias to if the server has to
> private ("internal") ip?
>
> Thanks.
>
>
>
> On Mon, Oct 11, 2010 at 7:21 PM, Nicolas Brenner <nico at clickfono.com>wrote:
>
>> Yup, here's the output:
>>
>> http://pastebin.freeswitch.org/14203
>>
>>
>> I'm trying to originate a call using the gateway on 200.13.15.220,
>> originating it from 172.26.6.161.
>>
>> Seems like no packets are going out at all, at least tcpdump is not
>> getting any. Oddly, if instead of setting sip-ip and rtp-ip to the ip of the
>> interface I need, I set ext-sip-ip and ext-rtp-ip to that ip, tcpdump starts
>> seeing traffic, but the traffic is originated from the wrong ip/interface
>> (the public one, instead of the one with the ipsec vpn).
>>
>>
>>
>> On Mon, Oct 11, 2010 at 6:50 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> When you do "sofia status" do you see the correct IP address for the
>>> profile in question?
>>> Also, when you make an outbound call can you see where (if?) the packets
>>> are going? Perhaps to another interface?
>>>
>>> -MC
>>>
>>>
>>> On Mon, Oct 11, 2010 at 2:44 PM, Nicolas Brenner <nico at clickfono.com>wrote:
>>>
>>>> The system routes are well configured, I can ping and telnet the gateway
>>>> from bash, all traffic to that gateway's ip is routed through the needed
>>>> interface. Now, are there other routes I should be addressing in FS config
>>>> files?
>>>>
>>>>
>>>> On Mon, Oct 11, 2010 at 6:33 PM, <Russell.Mosemann at cune.org> wrote:
>>>>
>>>>> Another thing to check is that there is only one IP address assigned to
>>>>> the interface. A problem with using the "right" IP address can occur if
>>>>> there is more than one IP address assigned to the same interface.
>>>>> Without
>>>>> routing rules, the operating system chooses the IP address from the
>>>>> interface that it believes is the best one for delivering the traffic,
>>>>> and sometimes it is not the address you want to use.
>>>>>
>>>>> --
>>>>> Russell Mosemann
>>>>>
>>>>>
>>>>>
>>>>> ________________________________________________________
>>>>> Concordia University, Nebraska
>>>>> See http://www.cune.edu/ for the latest news and events!
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
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From anthony.minessale at gmail.com  Mon Oct 18 09:50:17 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 18 Oct 2010 11:50:17 -0500
Subject: [Freeswitch-users] problem when recording conferences
In-Reply-To: <26893.1287404279@ccs.covici.com>
References: <26893.1287404279@ccs.covici.com>
Message-ID: <AANLkTimiYVbgm_UeLpfqPYUs5TVr2UBxnY2Xk03o1-hN@mail.gmail.com>

probably the energy, you could lower the default.

On Mon, Oct 18, 2010 at 7:17 AM,  <covici at ccs.covici.com> wrote:
> Hi. ?I am having a problem with conference recordings using
> auto_record. ?Now it does record, but with some users if the user is
> silent for a while, then the first so many milliseconds of his first
> word cuts off. ?Is this a function of the energe setting, waste setting
> or how else can I prevent this?
>
> Thanks in advance for any suggestionns.
>
> --
> Your life is like a penny. ?You're going to lose it. ?The question is:
> How do
> you spend it?
>
> ? ? ? ? John Covici
> ? ? ? ? covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From nico at clickfono.com  Mon Oct 18 09:50:20 2010
From: nico at clickfono.com (Nicolas Brenner)
Date: Mon, 18 Oct 2010 13:50:20 -0300
Subject: [Freeswitch-users] Gateway responding to random port (rport
	problem?)
Message-ID: <AANLkTi=ErWZP5SEr6p_=rej+xhioBQmxa7cCeg045c3i@mail.gmail.com>

Hello everyone,


I'm having an issue with a VoIP provider where the gateway will respond to
any random port instead of 5080. I sent the corresponding trace to the
provider and they told me to "disable rport and nat". I removed the auto-nat
parameters in the corresponding sip profile, but I keep having the same
issue (I don't think it has anything to do with nat anyway).

On the trace I noticed that INVITEs sent by FS have an empty rport parameter
on the Via header, but according to what I've read, this is standard and it
indicates that the response should be sent to the originating port (5080),
which is not being done by the gateway ("Avaya SIP Enablement Services").
How may I go about solving this? should I set the rport parameter to
something specific (eg. 5080)? or remove it altogether?


Thanks,



Nico
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From msc at freeswitch.org  Mon Oct 18 09:59:40 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Oct 2010 09:59:40 -0700
Subject: [Freeswitch-users] exec_after*
In-Reply-To: <0298143B64B347A996A1F1766DF3D7F8@mbnet.local>
References: <0298143B64B347A996A1F1766DF3D7F8@mbnet.local>
Message-ID: <AANLkTikQoNS+qqmVZWR76P7oR4CbNwn0VLm7NuB2sNAN@mail.gmail.com>

Well, the only difference is that one is executing an API and the other
executes a dialplan application. AFAIK they should operate in a similar
fashion. The only difference I see is that you have supplied an argument in
your API call ("${recfile}") but not in your exec_after_bridge call.

-MC

On Sun, Oct 17, 2010 at 3:33 PM, Jeroen C. van Gelderen
<slim at thegreek.com>wrote:

> Hi,
>
> My helpdesk extensions are recorded. When the recording is complete (after
> bridge) I want to run a script to create a ticket and attach said recording
> to it.
>
> Initially I tried to use
>
> [...]
> Dialplan: FreeTDM/3:1/ Action set(exec_after_bridge_app=system)
> Dialplan: FreeTDM/3:1/ Action
> set(exec_after_bridge_arg=/root/wav2ticket.php)
>
[...]
>
> but this doesn't result in an invocation of my script and seemingly no
> helpful diagnostic messages.
>
> I changed to using
>
> <action application="set"
>  data="api_after_bridge=system /root/wav2ticket.php ${recfile}"/>
>
> which DOES work.
>
> Can anyone clarify the difference between the two approaches?
>
> Cheers,
> -Slim
> --
> Jeroen C. "Slim" van Gelderen
> Olympic Sports Data Services
> Email: slim at thegreek.com
> Phone: +1 876 953 6182 x128
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sos at sokhapkin.dyndns.org  Mon Oct 18 10:10:58 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 18 Oct 2010 13:10:58 -0400
Subject: [Freeswitch-users] exec_after*
In-Reply-To: <AANLkTikQoNS+qqmVZWR76P7oR4CbNwn0VLm7NuB2sNAN@mail.gmail.com>
References: <0298143B64B347A996A1F1766DF3D7F8@mbnet.local>
	<AANLkTikQoNS+qqmVZWR76P7oR4CbNwn0VLm7NuB2sNAN@mail.gmail.com>
Message-ID: <201010181310.58288.sos@sokhapkin.dyndns.org>

Will the DP application specified with exec_after_bridge executed if the 
channel was hung up?

On Monday 18 October 2010, Michael Collins wrote:
> Well, the only difference is that one is executing an API and the other
> executes a dialplan application. AFAIK they should operate in a similar
> fashion. The only difference I see is that you have supplied an argument in
> your API call ("${recfile}") but not in your exec_after_bridge call.
> 
> -MC
> 
> On Sun, Oct 17, 2010 at 3:33 PM, Jeroen C. van Gelderen
> 
> <slim at thegreek.com>wrote:
> > Hi,
> >
> > My helpdesk extensions are recorded. When the recording is complete
> > (after bridge) I want to run a script to create a ticket and attach said
> > recording to it.
> >
> > Initially I tried to use
> >
> > [...]
> > Dialplan: FreeTDM/3:1/ Action set(exec_after_bridge_app=system)
> > Dialplan: FreeTDM/3:1/ Action
> > set(exec_after_bridge_arg=/root/wav2ticket.php)
> 
> [...]
> 
> > but this doesn't result in an invocation of my script and seemingly no
> > helpful diagnostic messages.
> >
> > I changed to using
> >
> > <action application="set"
> >  data="api_after_bridge=system /root/wav2ticket.php ${recfile}"/>
> >
> > which DOES work.
> >
> > Can anyone clarify the difference between the two approaches?
> >
> > Cheers,
> > -Slim
> > --
> > Jeroen C. "Slim" van Gelderen
> > Olympic Sports Data Services
> > Email: slim at thegreek.com
> > Phone: +1 876 953 6182 x128
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 



From slim at thegreek.com  Mon Oct 18 12:12:09 2010
From: slim at thegreek.com (Jeroen C. van Gelderen)
Date: Mon, 18 Oct 2010 14:12:09 -0500
Subject: [Freeswitch-users] exec_after*
In-Reply-To: <201010181310.58288.sos@sokhapkin.dyndns.org>
References: <0298143B64B347A996A1F1766DF3D7F8@mbnet.local><AANLkTikQoNS+qqmVZWR76P7oR4CbNwn0VLm7NuB2sNAN@mail.gmail.com>
	<201010181310.58288.sos@sokhapkin.dyndns.org>
Message-ID: <0BC944DEF9624F2CA6302921ACD06963@mbnet.local>


Bingo.

The exec_after_bridge_app call is guarded by:

  if (!inner_bridge && switch_channel_up(chan_a)) {
  }

whereas the exec_after_bridge_api is guarded by:

  if (!inner_bridge) {
  }

Anyone know the reason for this difference? It would seem common for the
channel to be hung up by the time one wants to execute the registered
application.

Cheers,
-Slim
--
Jeroen C. "Slim" van Gelderen

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey
Okhapkin
Sent: Monday, October 18, 2010 12:11
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] exec_after*

Will the DP application specified with exec_after_bridge executed if the 
channel was hung up?

On Monday 18 October 2010, Michael Collins wrote:
> Well, the only difference is that one is executing an API and the other
> executes a dialplan application. AFAIK they should operate in a similar
> fashion. The only difference I see is that you have supplied an argument
in
> your API call ("${recfile}") but not in your exec_after_bridge call.
> 
> -MC
> 
> On Sun, Oct 17, 2010 at 3:33 PM, Jeroen C. van Gelderen
> 
> <slim at thegreek.com>wrote:
> > Hi,
> >
> > My helpdesk extensions are recorded. When the recording is complete
> > (after bridge) I want to run a script to create a ticket and attach said
> > recording to it.
> >
> > Initially I tried to use
> >
> > [...]
> > Dialplan: FreeTDM/3:1/ Action set(exec_after_bridge_app=system)
> > Dialplan: FreeTDM/3:1/ Action
> > set(exec_after_bridge_arg=/root/wav2ticket.php)
> 
> [...]
> 
> > but this doesn't result in an invocation of my script and seemingly no
> > helpful diagnostic messages.
> >
> > I changed to using
> >
> > <action application="set"
> >  data="api_after_bridge=system /root/wav2ticket.php ${recfile}"/>
> >
> > which DOES work.
> >
> > Can anyone clarify the difference between the two approaches?
> >
> > Cheers,
> > -Slim
> > --
> > Jeroen C. "Slim" van Gelderen
> > Olympic Sports Data Services
> > Email: slim at thegreek.com
> > Phone: +1 876 953 6182 x128
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 


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



From msc at freeswitch.org  Mon Oct 18 12:29:03 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Oct 2010 12:29:03 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call This Wednesday Oct 20
Message-ID: <AANLkTi=ajK93uyCyYKg4X+MzujvxEW_gNgZxtkfTzdHJ@mail.gmail.com>

Hello all!

Here is the agenda page for this Wednesday's call:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_10_20

We didn't have a speaker scheduled for Wednesday so I will spend a few
minutes talking about the wiki. Please bring your wiki questions. In fact,
if you can please send me your wiki questions in advance so that I can
research them.

If you have any agenda items for the call this week please put them on the
list.

Thanks!
-Michael
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From sos at sokhapkin.dyndns.org  Mon Oct 18 12:40:30 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 18 Oct 2010 15:40:30 -0400
Subject: [Freeswitch-users] exec_after*
In-Reply-To: <0BC944DEF9624F2CA6302921ACD06963@mbnet.local>
References: <0298143B64B347A996A1F1766DF3D7F8@mbnet.local>
	<201010181310.58288.sos@sokhapkin.dyndns.org>
	<0BC944DEF9624F2CA6302921ACD06963@mbnet.local>
Message-ID: <201010181540.30153.sos@sokhapkin.dyndns.org>

The "h" extension in asterisk is a feature I used extensively. Lack of a 
hangup dialplan handler makes DP programming in FS harder and less clear. 
Fortunately, CDR handlers allow to do some post call cleanup work, but in-
dialplan implementation would be much better and clearer.

On Monday 18 October 2010, Jeroen C. van Gelderen wrote:
> Bingo.
> 
> The exec_after_bridge_app call is guarded by:
> 
>   if (!inner_bridge && switch_channel_up(chan_a)) {
>   }
> 
> whereas the exec_after_bridge_api is guarded by:
> 
>   if (!inner_bridge) {
>   }
> 
> Anyone know the reason for this difference? It would seem common for the
> channel to be hung up by the time one wants to execute the registered
> application.
> 
> Cheers,
> -Slim
> --
> Jeroen C. "Slim" van Gelderen
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey
> Okhapkin
> Sent: Monday, October 18, 2010 12:11
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] exec_after*
> 
> Will the DP application specified with exec_after_bridge executed if the
> channel was hung up?
> 
> On Monday 18 October 2010, Michael Collins wrote:
> > Well, the only difference is that one is executing an API and the other
> > executes a dialplan application. AFAIK they should operate in a similar
> > fashion. The only difference I see is that you have supplied an argument
> 
> in
> 
> > your API call ("${recfile}") but not in your exec_after_bridge call.
> >
> > -MC
> >
> > On Sun, Oct 17, 2010 at 3:33 PM, Jeroen C. van Gelderen
> >
> > <slim at thegreek.com>wrote:
> > > Hi,
> > >
> > > My helpdesk extensions are recorded. When the recording is complete
> > > (after bridge) I want to run a script to create a ticket and attach
> > > said recording to it.
> > >
> > > Initially I tried to use
> > >
> > > [...]
> > > Dialplan: FreeTDM/3:1/ Action set(exec_after_bridge_app=system)
> > > Dialplan: FreeTDM/3:1/ Action
> > > set(exec_after_bridge_arg=/root/wav2ticket.php)
> >
> > [...]
> >
> > > but this doesn't result in an invocation of my script and seemingly no
> > > helpful diagnostic messages.
> > >
> > > I changed to using
> > >
> > > <action application="set"
> > >  data="api_after_bridge=system /root/wav2ticket.php ${recfile}"/>
> > >
> > > which DOES work.
> > >
> > > Can anyone clarify the difference between the two approaches?
> > >
> > > Cheers,
> > > -Slim
> > > --
> > > Jeroen C. "Slim" van Gelderen
> > > Olympic Sports Data Services
> > > Email: slim at thegreek.com
> > > Phone: +1 876 953 6182 x128
> > >
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >s http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From msc at freeswitch.org  Mon Oct 18 14:20:05 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Oct 2010 14:20:05 -0700
Subject: [Freeswitch-users] exec_after*
In-Reply-To: <201010181540.30153.sos@sokhapkin.dyndns.org>
References: <0298143B64B347A996A1F1766DF3D7F8@mbnet.local>
	<201010181310.58288.sos@sokhapkin.dyndns.org>
	<0BC944DEF9624F2CA6302921ACD06963@mbnet.local>
	<201010181540.30153.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTi=9-OexbZWh0RQMtxqG_oDYXKDQ=J4ekJMEV4OJ@mail.gmail.com>

On Mon, Oct 18, 2010 at 12:40 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> The "h" extension in asterisk is a feature I used extensively. Lack of a
> hangup dialplan handler makes DP programming in FS harder and less clear.
>

That's what api_hangup_hook is for. Call ends, do an API that can do pretty
much anything you can imagine and you're done. You can even set
session_in_hangup_hook to true so that you have the channel's variables
available even though the channel has been disconnected.

> Fortunately, CDR handlers allow to do some post call cleanup work, but in-
> dialplan implementation would be much better and clearer.
>
Much better? No. Much clearer, possibly. The benefits of not doing cleanup
work in the dialplan AFTER the channel has gone away far outweigh the
drawbacks. Use api_hangup_hook and session_in_hangup_hook=true. Doing this
stuff outside the dialplan is more efficient and less error-prone.

-MC
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From sos at sokhapkin.dyndns.org  Mon Oct 18 14:32:30 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 18 Oct 2010 17:32:30 -0400
Subject: [Freeswitch-users] exec_after*
In-Reply-To: <AANLkTi=9-OexbZWh0RQMtxqG_oDYXKDQ=J4ekJMEV4OJ@mail.gmail.com>
References: <0298143B64B347A996A1F1766DF3D7F8@mbnet.local>
	<201010181540.30153.sos@sokhapkin.dyndns.org>
	<AANLkTi=9-OexbZWh0RQMtxqG_oDYXKDQ=J4ekJMEV4OJ@mail.gmail.com>
Message-ID: <201010181732.31085.sos@sokhapkin.dyndns.org>

Thank you for the session_in_hangup_hook hint...

On Monday 18 October 2010, Michael Collins wrote:
> On Mon, Oct 18, 2010 at 12:40 PM, Sergey Okhapkin
> 
> <sos at sokhapkin.dyndns.org>wrote:
> > The "h" extension in asterisk is a feature I used extensively. Lack of a
> > hangup dialplan handler makes DP programming in FS harder and less clear.
> 
> That's what api_hangup_hook is for. Call ends, do an API that can do pretty
> much anything you can imagine and you're done. You can even set
> session_in_hangup_hook to true so that you have the channel's variables
> available even though the channel has been disconnected.
> 
> > Fortunately, CDR handlers allow to do some post call cleanup work, but
> > in- dialplan implementation would be much better and clearer.
> 
> Much better? No. Much clearer, possibly. The benefits of not doing cleanup
> work in the dialplan AFTER the channel has gone away far outweigh the
> drawbacks. Use api_hangup_hook and session_in_hangup_hook=true. Doing this
> stuff outside the dialplan is more efficient and less error-prone.
> 
> -MC
> 



From pjintheusa at gmail.com  Mon Oct 18 14:58:05 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 18 Oct 2010 17:58:05 -0400
Subject: [Freeswitch-users] exec_after*
In-Reply-To: <201010181732.31085.sos@sokhapkin.dyndns.org>
References: <0298143B64B347A996A1F1766DF3D7F8@mbnet.local>
	<201010181540.30153.sos@sokhapkin.dyndns.org>
	<AANLkTi=9-OexbZWh0RQMtxqG_oDYXKDQ=J4ekJMEV4OJ@mail.gmail.com>
	<201010181732.31085.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTinv77uvuWJ-ZVxYY+b8guvvXNm2yQ8PzGcfyhKN@mail.gmail.com>

A cookbook entry perhaps.....

On Mon, Oct 18, 2010 at 5:32 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> Thank you for the session_in_hangup_hook hint...
>
> On Monday 18 October 2010, Michael Collins wrote:
> > On Mon, Oct 18, 2010 at 12:40 PM, Sergey Okhapkin
> >
> > <sos at sokhapkin.dyndns.org>wrote:
> > > The "h" extension in asterisk is a feature I used extensively. Lack of
> a
> > > hangup dialplan handler makes DP programming in FS harder and less
> clear.
> >
> > That's what api_hangup_hook is for. Call ends, do an API that can do
> pretty
> > much anything you can imagine and you're done. You can even set
> > session_in_hangup_hook to true so that you have the channel's variables
> > available even though the channel has been disconnected.
> >
> > > Fortunately, CDR handlers allow to do some post call cleanup work, but
> > > in- dialplan implementation would be much better and clearer.
> >
> > Much better? No. Much clearer, possibly. The benefits of not doing
> cleanup
> > work in the dialplan AFTER the channel has gone away far outweigh the
> > drawbacks. Use api_hangup_hook and session_in_hangup_hook=true. Doing
> this
> > stuff outside the dialplan is more efficient and less error-prone.
> >
> > -MC
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Oct 18 15:41:31 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Oct 2010 15:41:31 -0700
Subject: [Freeswitch-users] exec_after*
In-Reply-To: <AANLkTinv77uvuWJ-ZVxYY+b8guvvXNm2yQ8PzGcfyhKN@mail.gmail.com>
References: <0298143B64B347A996A1F1766DF3D7F8@mbnet.local>
	<201010181540.30153.sos@sokhapkin.dyndns.org>
	<AANLkTi=9-OexbZWh0RQMtxqG_oDYXKDQ=J4ekJMEV4OJ@mail.gmail.com>
	<201010181732.31085.sos@sokhapkin.dyndns.org>
	<AANLkTinv77uvuWJ-ZVxYY+b8guvvXNm2yQ8PzGcfyhKN@mail.gmail.com>
Message-ID: <AANLkTinPZGJsfBGojpUys6WfmdT3GU3p-doVN3h-ABMg@mail.gmail.com>

Definitely. File this one under "duh"
-MC

On Mon, Oct 18, 2010 at 2:58 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> A cookbook entry perhaps.....
>
> On Mon, Oct 18, 2010 at 5:32 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org
> > wrote:
>
>> Thank you for the session_in_hangup_hook hint...
>>
>> On Monday 18 October 2010, Michael Collins wrote:
>> > On Mon, Oct 18, 2010 at 12:40 PM, Sergey Okhapkin
>> >
>> > <sos at sokhapkin.dyndns.org>wrote:
>> > > The "h" extension in asterisk is a feature I used extensively. Lack of
>> a
>> > > hangup dialplan handler makes DP programming in FS harder and less
>> clear.
>> >
>> > That's what api_hangup_hook is for. Call ends, do an API that can do
>> pretty
>> > much anything you can imagine and you're done. You can even set
>> > session_in_hangup_hook to true so that you have the channel's variables
>> > available even though the channel has been disconnected.
>> >
>> > > Fortunately, CDR handlers allow to do some post call cleanup work, but
>> > > in- dialplan implementation would be much better and clearer.
>> >
>> > Much better? No. Much clearer, possibly. The benefits of not doing
>> cleanup
>> > work in the dialplan AFTER the channel has gone away far outweigh the
>> > drawbacks. Use api_hangup_hook and session_in_hangup_hook=true. Doing
>> this
>> > stuff outside the dialplan is more efficient and less error-prone.
>> >
>> > -MC
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From djbinter at gmail.com  Mon Oct 18 17:31:55 2010
From: djbinter at gmail.com (DJB International)
Date: Mon, 18 Oct 2010 17:31:55 -0700
Subject: [Freeswitch-users] Limit [ERROR] question
Message-ID: <AANLkTiktTtMkbBB4m+ErkHTvF_xxNtYV-hsTKoJ22wwJ@mail.gmail.com>

I have noticed once in a while in the log file with the following error:

[ERR] switch_limit.c:86 Unset limit backendlist!

What exactly would cause this error?

Thank you,
Dorn B.
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From dujinfang at gmail.com  Mon Oct 18 19:16:53 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 19 Oct 2010 10:16:53 +0800
Subject: [Freeswitch-users] FreeSWITCH HA question&talk
In-Reply-To: <AANLkTi=Z7q4n+tKMvBWiXWezNfLXy3-jQi--giKRuawB@mail.gmail.com>
References: <AANLkTimG_V=pBbSzf2hr=3KUex_Jx9GB=E5sDWSqCmcS@mail.gmail.com>
	<AANLkTi=Z7q4n+tKMvBWiXWezNfLXy3-jQi--giKRuawB@mail.gmail.com>
Message-ID: <AANLkTi=dB12ECNNdHydekjv_n4EQ_M45hXbj3NYBaCr_@mail.gmail.com>

Yes, just google freeswitch ha

http://wiki.freeswitch.org/wiki/Freeswitch_HA

On Sat, Oct 16, 2010 at 3:41 PM, Henry Huang <b_ball_henry at hotmail.com> wrote:
> For the FS HA, is there a wiki page you are referring to?
>
> Henry
>
> On Sat, Oct 16, 2010 at 11:12 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> Hi,
>>
>> I tested FS HA, it works like a charm. however, I'd like to talk about
>> some more:
>>
>> I tested on my local computer by enabling tack-calls. ?make a call and
>> kill FS, restart FS and execute sofia recover.
>>
>> 1) originate user/1000 &record(/tmp/blah.wav)
>>
>> Recover works, however, it overwrote the recording. Added
>> RECORD_APPEND=true made it work.
>>
>> But for other APPs like socket, it might make a new socket to a
>> outbound socket, can it send a var like channel_recovered so a
>> outbound socket can aware of a recover?
>>
>>
>> 2) call from 1000 to 1001
>>
>> Recover works. If I also do uuid_record, the recording cannot recover.
>> Seems there are no infos in sip_recovery table.
>>
>> Neither does record_session, though there's an entry in sip_recovery
>> table. I guess it might be possible to get this work.
>>
>> If it can send a channel_recovered event, then it might be possible to
>> continue the record in outside logic(say, event socket).
>>
>> 3) call from 1000 to a gateway outside (NAT-ed), leg-a can recover,
>> leg-b failed with:
>>
>> 2010-10-16 10:28:17.162241 [ERR] sofia_glue.c:3190 AUDIO RTP REPORTS
>> ERROR: [Bind Error!]
>>
>> I guess it is caused by NAT. (since I kill & restart FS manually, it
>> may take 10 seconds). Is it possible to change the router settings? Or
>> could FS send a re-Invite to re-establish media?
>>
>>
>> Below are some fantastic thoughts:
>>
>> 4) N + M redundancy
>>
>> Default FS HA is designed to 1+1 redundancy, it might be interesting
>> for N(active)+M(standby) in large farms. If one active FS fails, bind
>> the float IP to a standby server and go.
>>
>> 5) N (N>2) servers without float IP
>>
>> Let's say each client UA register to 2 (or more) servers, If one
>> server fails, a FS controller then re-call the client in an active
>> call through a pair-server and the client responsible to hangup the
>> originate call and automatically answer the new one. This need client
>> support of course. ?:)
>>
>>
>> --
>> Blog: http://www.dujinfang.com
>> Proj:? http://www.freeswitch.org.cn
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From andy at fabulous4.co.uk  Tue Oct 19 02:31:12 2010
From: andy at fabulous4.co.uk (Andy Ayers)
Date: Tue, 19 Oct 2010 10:31:12 +0100
Subject: [Freeswitch-users] record on demand for a conference call
In-Reply-To: <AANLkTimuurW-3SJX1+JB=rzbgmq1mGd12hQEvzmfD4A6@mail.gmail.com>
References: <6DFF80D26E14467299C2B7789D1D82D6@D810>
	<AANLkTimuurW-3SJX1+JB=rzbgmq1mGd12hQEvzmfD4A6@mail.gmail.com>
Message-ID: <AANLkTi=3mHhqtWbLiw+onv8MDAWGLF+0mhKt-M6kiWBe@mail.gmail.com>

thanks for your interest Michael, it would be to allow the moderator to
record an introduction to the call so when it's listened to by others it has
context. But actually more useful is to be able to start recording when the
moderator wants it to to cut off the usually messy start to a conference
call. Is this possible, could I configure a dtmf key press to initiate the
recording somehow?


On 13 October 2010 20:05, Michael Collins <msc at freeswitch.org> wrote:

> Just curious - what value is there to recording the call when only one
> person is there?
> -MC
>
>  On Wed, Oct 13, 2010 at 3:42 AM, Andy <andy at fabulous4.co.uk> wrote:
>
>>   Hi,
>>
>> I would like to set up my conferencing system so that it's possible to
>> initiate the recording of the conference when the moderator presses a key on
>> their keypad.
>>
>> Failing that I would like to know if there is any way of recording the
>> entire conference call from the moment the first caller arrives rather than
>> using the auto-record feature which only kicks in when 2 or more people are
>> connected.
>>
>> Can anyone give me some idea how this can be done or point me to the right
>> wiki pages?
>>
>> Many thanks for you help
>> Andy
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kond at nstel.ru  Tue Oct 19 02:59:07 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Tue, 19 Oct 2010 13:59:07 +0400
Subject: [Freeswitch-users] digium board
Message-ID: <20101019095907.845CE11525@mail.nstel.ru>

Hello all!
Can anybody please advise if Digium TE121 board is supported by FS?
In the wiki i have seen pages devoted to Sangoma. 
What will be right choice for E1 PRI interface? Sangoma or Digium? Or may be something else?
Thanks in advance,
Nikolay.
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From tculjaga at gmail.com  Tue Oct 19 03:07:54 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 19 Oct 2010 12:07:54 +0200
Subject: [Freeswitch-users] FreeSWITCH HA question&talk
In-Reply-To: <AANLkTi=dB12ECNNdHydekjv_n4EQ_M45hXbj3NYBaCr_@mail.gmail.com>
References: <AANLkTimG_V=pBbSzf2hr=3KUex_Jx9GB=E5sDWSqCmcS@mail.gmail.com>
	<AANLkTi=Z7q4n+tKMvBWiXWezNfLXy3-jQi--giKRuawB@mail.gmail.com>
	<AANLkTi=dB12ECNNdHydekjv_n4EQ_M45hXbj3NYBaCr_@mail.gmail.com>
Message-ID: <AANLkTi=tbZUO+Nnpr0=Sw0Gbcmfn6uJYMv3js7+QbwWU@mail.gmail.com>

On Tue, Oct 19, 2010 at 4:16 AM, Seven Du <dujinfang at gmail.com> wrote:

> Yes, just google freeswitch ha
>
> http://wiki.freeswitch.org/wiki/Freeswitch_HA
>
>
ok, why don't we document it properly?

the information on this page is inconsistent and insufficient.
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From gmaruzz at gmail.com  Tue Oct 19 03:18:03 2010
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 19 Oct 2010 12:18:03 +0200
Subject: [Freeswitch-users] digium board
In-Reply-To: <20101019095907.845CE11525@mail.nstel.ru>
References: <20101019095907.845CE11525@mail.nstel.ru>
Message-ID: <AANLkTika2=mAOpg2f58v1_rtH=3ur+ueU5+8=7znRO6W@mail.gmail.com>

On Tue, Oct 19, 2010 at 11:59 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:
> Hello all!
> Can anybody please advise if Digium TE121 board is supported by FS?
> In the wiki i have seen pages devoted to Sangoma.
> What will be right choice for E1 PRI interface? Sangoma or Digium? Or may be
> something else?

Both Digium and Sangoma cards are supported by FS.

Sangoma (as a company) is actively developing and supporting FS, and
devotes resources to FS support and development, while Digium (as a
company) do not.

-giovanni

> Thanks in advance,
> Nikolay.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From mnhassan at usa.net  Tue Oct 19 03:27:56 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Tue, 19 Oct 2010 16:27:56 +0600
Subject: [Freeswitch-users] digium board
In-Reply-To: <20101019095907.845CE11525@mail.nstel.ru>
References: <20101019095907.845CE11525@mail.nstel.ru>
Message-ID: <AANLkTi=4=Y+S6oLsCgeU1AB7N6z=+s1cnRJxLORoTsKQ@mail.gmail.com>

FreeSWITCH user FreeTDM to interface with these hardware. FreeTDM was
earlier known as OpenZAP and supports Digium.

Regards
HASSAN


On 2010-10-19, Nikolay Kondratyev <kond at nstel.ru> wrote:
> Hello all!
> Can anybody please advise if Digium TE121 board is supported by FS?
> In the wiki i have seen pages devoted to Sangoma.
> What will be right choice for E1 PRI interface? Sangoma or Digium? Or may be
> something else?
> Thanks in advance,
> Nikolay.
>

-- 
Sent from my mobile device


From dujinfang at gmail.com  Tue Oct 19 03:39:24 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 19 Oct 2010 18:39:24 +0800
Subject: [Freeswitch-users] FreeSWITCH HA question&talk
In-Reply-To: <AANLkTi=tbZUO+Nnpr0=Sw0Gbcmfn6uJYMv3js7+QbwWU@mail.gmail.com>
References: <AANLkTimG_V=pBbSzf2hr=3KUex_Jx9GB=E5sDWSqCmcS@mail.gmail.com>
	<AANLkTi=Z7q4n+tKMvBWiXWezNfLXy3-jQi--giKRuawB@mail.gmail.com>
	<AANLkTi=dB12ECNNdHydekjv_n4EQ_M45hXbj3NYBaCr_@mail.gmail.com>
	<AANLkTi=tbZUO+Nnpr0=Sw0Gbcmfn6uJYMv3js7+QbwWU@mail.gmail.com>
Message-ID: <AANLkTinSuCgTGKK6mp12rE5y0eS6Ebm49kP6L4xT4nVA@mail.gmail.com>

Anyone are welcome to help document it consistent and sufficient.

On Tue, Oct 19, 2010 at 6:07 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>
>
> On Tue, Oct 19, 2010 at 4:16 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> Yes, just google freeswitch ha
>>
>> http://wiki.freeswitch.org/wiki/Freeswitch_HA
>>
>
> ok, why don't we document it properly?
>
> the information on this page is inconsistent and insufficient.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From tculjaga at gmail.com  Tue Oct 19 04:07:31 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 19 Oct 2010 13:07:31 +0200
Subject: [Freeswitch-users] FreeSWITCH HA question&talk
In-Reply-To: <AANLkTinSuCgTGKK6mp12rE5y0eS6Ebm49kP6L4xT4nVA@mail.gmail.com>
References: <AANLkTimG_V=pBbSzf2hr=3KUex_Jx9GB=E5sDWSqCmcS@mail.gmail.com>
	<AANLkTi=Z7q4n+tKMvBWiXWezNfLXy3-jQi--giKRuawB@mail.gmail.com>
	<AANLkTi=dB12ECNNdHydekjv_n4EQ_M45hXbj3NYBaCr_@mail.gmail.com>
	<AANLkTi=tbZUO+Nnpr0=Sw0Gbcmfn6uJYMv3js7+QbwWU@mail.gmail.com>
	<AANLkTinSuCgTGKK6mp12rE5y0eS6Ebm49kP6L4xT4nVA@mail.gmail.com>
Message-ID: <AANLkTikrOhycJ9x3bqm4=khJioxUfRcUFZjcVPhXOPyj@mail.gmail.com>

On Tue, Oct 19, 2010 at 12:39 PM, Seven Du <dujinfang at gmail.com> wrote:

> Anyone are welcome to help document it consistent and sufficient.
>
> On Tue, Oct 19, 2010 at 6:07 PM, Tihomir Culjaga <tculjaga at gmail.com>
> wrote:
> >
> >
> > On Tue, Oct 19, 2010 at 4:16 AM, Seven Du <dujinfang at gmail.com> wrote:
> >>
> >> Yes, just google freeswitch ha
> >>
> >> http://wiki.freeswitch.org/wiki/Freeswitch_HA
> >>
> >
> > ok, why don't we document it properly?
> >
> > the information on this page is inconsistent and insufficient.
> >
>

Please don't get me wrong, i can do it ... the only thing is that i missing
the 2nd installed server and test it properly.

can you, please describe this:

sysctl to set net.ipv4.ip_nonlocal_bind = 1
or
/proc/sys/net/ipv4/ip_nonlocal_bind


does it mean i can configure both FS to bind to an IP address that i will
keep floating between 2 servers:

from vars.conf
<X-PRE-PROCESS cmd="set" data="bind_server_ip=10.1.1.100"/>

assuming my floating IP is 10.1.1.100.

is that correct ?




regarding your NAT issue, i guess we can overcome it by butting FS out of
media path


uuid_media off <uuid>
uuid_media <uuid>


maybe there is something general that can do the job for all ongoing
calls....




anyhow, im going to do it once i got the 2nd server installed ...
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From david.ponzone at ipeva.fr  Tue Oct 19 04:24:47 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 19 Oct 2010 13:24:47 +0200
Subject: [Freeswitch-users] FreeSWITCH HA question&talk
In-Reply-To: <AANLkTikrOhycJ9x3bqm4=khJioxUfRcUFZjcVPhXOPyj@mail.gmail.com>
References: <AANLkTimG_V=pBbSzf2hr=3KUex_Jx9GB=E5sDWSqCmcS@mail.gmail.com>
	<AANLkTi=Z7q4n+tKMvBWiXWezNfLXy3-jQi--giKRuawB@mail.gmail.com>
	<AANLkTi=dB12ECNNdHydekjv_n4EQ_M45hXbj3NYBaCr_@mail.gmail.com>
	<AANLkTi=tbZUO+Nnpr0=Sw0Gbcmfn6uJYMv3js7+QbwWU@mail.gmail.com>
	<AANLkTinSuCgTGKK6mp12rE5y0eS6Ebm49kP6L4xT4nVA@mail.gmail.com>
	<AANLkTikrOhycJ9x3bqm4=khJioxUfRcUFZjcVPhXOPyj@mail.gmail.com>
Message-ID: <CFEA1B4E-6CC9-412E-8B27-FC8AD48D0A85@ipeva.fr>

Tihomir,

Setting ipv4.ip_nonlocal_bind to 1 will allow any process to bind to  
an IP which is not configured on an interface.
So you can have your slave FS configured, running, on the right IP,  
but without the IP being actually configured on ethX.
When you want this slave to become active, you just have to configure  
the IP on ethX with ifconfig, and FS will start working.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 19/10/2010 ? 13:07, Tihomir Culjaga a ?crit :

>
>
> On Tue, Oct 19, 2010 at 12:39 PM, Seven Du <dujinfang at gmail.com>  
> wrote:
> Anyone are welcome to help document it consistent and sufficient.
>
> On Tue, Oct 19, 2010 at 6:07 PM, Tihomir Culjaga  
> <tculjaga at gmail.com> wrote:
> >
> >
> > On Tue, Oct 19, 2010 at 4:16 AM, Seven Du <dujinfang at gmail.com>  
> wrote:
> >>
> >> Yes, just google freeswitch ha
> >>
> >> http://wiki.freeswitch.org/wiki/Freeswitch_HA
> >>
> >
> > ok, why don't we document it properly?
> >
> > the information on this page is inconsistent and insufficient.
> >
>
> Please don't get me wrong, i can do it ... the only thing is that i  
> missing the 2nd installed server and test it properly.
>
> can you, please describe this:
>
> sysctl to set net.ipv4.ip_nonlocal_bind = 1
> or
> /proc/sys/net/ipv4/ip_nonlocal_bind
>
> does it mean i can configure both FS to bind to an IP address that i  
> will keep floating between 2 servers:
>
> from vars.conf
> <X-PRE-PROCESS cmd="set" data="bind_server_ip=10.1.1.100"/>
>
> assuming my floating IP is 10.1.1.100.
>
> is that correct ?
>
>
>
>
> regarding your NAT issue, i guess we can overcome it by butting FS  
> out of media path
>
>
> uuid_media off <uuid>
> uuid_media <uuid>
>
>
> maybe there is something general that can do the job for all ongoing  
> calls....
>
>
>
>
> anyhow, im going to do it once i got the 2nd server installed ...
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From steveayre at gmail.com  Tue Oct 19 04:35:14 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 19 Oct 2010 12:35:14 +0100
Subject: [Freeswitch-users] FreeSWITCH HA question&talk
In-Reply-To: <AANLkTikrOhycJ9x3bqm4=khJioxUfRcUFZjcVPhXOPyj@mail.gmail.com>
References: <AANLkTimG_V=pBbSzf2hr=3KUex_Jx9GB=E5sDWSqCmcS@mail.gmail.com>
	<AANLkTi=Z7q4n+tKMvBWiXWezNfLXy3-jQi--giKRuawB@mail.gmail.com>
	<AANLkTi=dB12ECNNdHydekjv_n4EQ_M45hXbj3NYBaCr_@mail.gmail.com>
	<AANLkTi=tbZUO+Nnpr0=Sw0Gbcmfn6uJYMv3js7+QbwWU@mail.gmail.com>
	<AANLkTinSuCgTGKK6mp12rE5y0eS6Ebm49kP6L4xT4nVA@mail.gmail.com>
	<AANLkTikrOhycJ9x3bqm4=khJioxUfRcUFZjcVPhXOPyj@mail.gmail.com>
Message-ID: <AANLkTi=eE0XOdk53c1V2SMpfycGSVbZzZi5PT5kQVU81@mail.gmail.com>

For the 'HA configuration' part, there is also this page (well, pages):
http://wiki.freeswitch.org/wiki/Enterprise_deployment

Which discusses a few ways to get high availability through things
like SRV DNS, Heartbeat etc.

The Freeswitch_HA page content was initially only for the sofia
recover features that allows calls to be restarted after a crash or
server migration.

-Steve


On 19 October 2010 12:07, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>
>
> On Tue, Oct 19, 2010 at 12:39 PM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> Anyone are welcome to help document it consistent and sufficient.
>>
>> On Tue, Oct 19, 2010 at 6:07 PM, Tihomir Culjaga <tculjaga at gmail.com>
>> wrote:
>> >
>> >
>> > On Tue, Oct 19, 2010 at 4:16 AM, Seven Du <dujinfang at gmail.com> wrote:
>> >>
>> >> Yes, just google freeswitch ha
>> >>
>> >> http://wiki.freeswitch.org/wiki/Freeswitch_HA
>> >>
>> >
>> > ok, why don't we document it properly?
>> >
>> > the information on this page is inconsistent and insufficient.
>> >
>
> Please don't get me wrong, i can do it ... the only thing is that i missing
> the 2nd installed server and test it properly.
>
> can you, please describe this:
>
> sysctl to set net.ipv4.ip_nonlocal_bind = 1
> or
> /proc/sys/net/ipv4/ip_nonlocal_bind
>
> does it mean i can configure both FS to bind to an IP address that i will
> keep floating between 2 servers:
>
> from vars.conf
> <X-PRE-PROCESS cmd="set" data="bind_server_ip=10.1.1.100"/>
>
> assuming my floating IP is 10.1.1.100.
>
> is that correct ?
>
>
>
>
> regarding your NAT issue, i guess we can overcome it by butting FS out of
> media path
>
>
> uuid_media off <uuid>
> uuid_media <uuid>
>
>
> maybe there is something general that can do the job for all ongoing
> calls....
>
>
>
>
> anyhow, im going to do it once i got the 2nd server installed ...
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From kond at nstel.ru  Tue Oct 19 04:46:25 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Tue, 19 Oct 2010 15:46:25 +0400
Subject: [Freeswitch-users] digium board
In-Reply-To: <AANLkTika2=mAOpg2f58v1_rtH=3ur+ueU5+8=7znRO6W@mail.gmail.com>
Message-ID: <20101019114626.001D211F53@mail.nstel.ru>

Giovanni, Hassan,
Thanks for the answers.
I'm going to go with Sangoma.
Nikolay. 

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On 
> Behalf Of Giovanni Maruzzelli
> Sent: Tuesday, October 19, 2010 2:18 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] digium board
> 
> On Tue, Oct 19, 2010 at 11:59 AM, Nikolay Kondratyev 
> <kond at nstel.ru> wrote:
> > Hello all!
> > Can anybody please advise if Digium TE121 board is supported by FS?
> > In the wiki i have seen pages devoted to Sangoma.
> > What will be right choice for E1 PRI interface? Sangoma or 
> Digium? Or 
> > may be something else?
> 
> Both Digium and Sangoma cards are supported by FS.
> 
> Sangoma (as a company) is actively developing and supporting 
> FS, and devotes resources to FS support and development, 
> while Digium (as a
> company) do not.
> 
> -giovanni
> 
> > Thanks in advance,
> > Nikolay.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> > rs
> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> --
> Sincerely,
> 
> Giovanni Maruzzelli
> Cell : +39-347-2665618
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
> itch-users
> http://www.freeswitch.org



From jmesquita at freeswitch.org  Tue Oct 19 05:10:20 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 19 Oct 2010 09:10:20 -0300
Subject: [Freeswitch-users] digium board
In-Reply-To: <20101019114626.001D211F53@mail.nstel.ru>
References: <AANLkTika2=mAOpg2f58v1_rtH=3ur+ueU5+8=7znRO6W@mail.gmail.com>
	<20101019114626.001D211F53@mail.nstel.ru>
Message-ID: <AANLkTi=T5m1c2z-cF8ODpOQagBzqi=05n1BXdNfKHap1@mail.gmail.com>

Keep in mind that there is a Brazilian company called Khomp too. They have
good quality hardware at decent price. Might be a good option for a try out.

mod_khomp is being actively developed and soon to be release under the git
head.

Regards,
Jo?o Mesquita


On Tue, Oct 19, 2010 at 8:46 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

> Giovanni, Hassan,
> Thanks for the answers.
> I'm going to go with Sangoma.
> Nikolay.
>
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On
> > Behalf Of Giovanni Maruzzelli
> > Sent: Tuesday, October 19, 2010 2:18 PM
> > To: FreeSWITCH Users Help
> > Subject: Re: [Freeswitch-users] digium board
> >
> > On Tue, Oct 19, 2010 at 11:59 AM, Nikolay Kondratyev
> > <kond at nstel.ru> wrote:
> > > Hello all!
> > > Can anybody please advise if Digium TE121 board is supported by FS?
> > > In the wiki i have seen pages devoted to Sangoma.
> > > What will be right choice for E1 PRI interface? Sangoma or
> > Digium? Or
> > > may be something else?
> >
> > Both Digium and Sangoma cards are supported by FS.
> >
> > Sangoma (as a company) is actively developing and supporting
> > FS, and devotes resources to FS support and development,
> > while Digium (as a
> > company) do not.
> >
> > -giovanni
> >
> > > Thanks in advance,
> > > Nikolay.
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> > > rs
> > > http://www.freeswitch.org
> > >
> > >
> >
> >
> >
> > --
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > Cell : +39-347-2665618
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
> > itch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From nico at clickfono.com  Tue Oct 19 05:51:14 2010
From: nico at clickfono.com (Nicolas Brenner)
Date: Tue, 19 Oct 2010 09:51:14 -0300
Subject: [Freeswitch-users] Remove rport from Via header
Message-ID: <AANLkTiky=9vA5uk-gqKHHjuabF0VQE1tGk0wODPc7Op2@mail.gmail.com>

Hello, I need to either remove the rport parameter from the Via header on an
initial invite or set its value to a specific port. I've looked through the
mailing archives and the wiki but I haven't found anything. I appreciate any
help or pointers I can get, thanks.


Nico
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From rupa at rupa.com  Tue Oct 19 05:56:36 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 19 Oct 2010 07:56:36 -0500
Subject: [Freeswitch-users] Limit [ERROR] question
In-Reply-To: <AANLkTiktTtMkbBB4m+ErkHTvF_xxNtYV-hsTKoJ22wwJ@mail.gmail.com>
References: <AANLkTiktTtMkbBB4m+ErkHTvF_xxNtYV-hsTKoJ22wwJ@mail.gmail.com>
Message-ID: <AANLkTinecv9E6YnfN+8XfKYoBNWaEu4UpxciPD0izAzj@mail.gmail.com>

This is a "shouldn't happen" sanity check you are tripping over -- I
probably should have worded the error message better.

Immediate cause is that the list of backends (hash, sql, etc) that have
registered interest in this call is empty.   This shouldn't happen because
we register the handler at the same time that we set the channel var of
interested backends.

Can you find a way to reproduce this reliably?

The effect of this would be to (potentially) not decrement a counter when we
should have.

On Mon, Oct 18, 2010 at 7:31 PM, DJB International <djbinter at gmail.com>wrote:

> I have noticed once in a while in the log file with the following error:
>
> [ERR] switch_limit.c:86 Unset limit backendlist!
>
> What exactly would cause this error?
>
> Thank you,
> Dorn B.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From david.ponzone at ipeva.fr  Tue Oct 19 06:04:33 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 19 Oct 2010 15:04:33 +0200
Subject: [Freeswitch-users] Remove rport from Via header
In-Reply-To: <AANLkTiky=9vA5uk-gqKHHjuabF0VQE1tGk0wODPc7Op2@mail.gmail.com>
References: <AANLkTiky=9vA5uk-gqKHHjuabF0VQE1tGk0wODPc7Op2@mail.gmail.com>
Message-ID: <3799D7D4-A663-4C55-A4A5-A63585E413CE@ipeva.fr>

Nicolas,

you may try to set sip_via_rport in the dialplan, but I don't know if  
this will work.
I suppose it depends on what you are trying to defeat.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 19/10/2010 ? 14:51, Nicolas Brenner a ?crit :

> Hello, I need to either remove the rport parameter from the Via  
> header on an initial invite or set its value to a specific port.  
> I've looked through the mailing archives and the wiki but I haven't  
> found anything. I appreciate any help or pointers I can get, thanks.
>
>
> Nico
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Tue Oct 19 06:19:45 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Oct 2010 08:19:45 -0500
Subject: [Freeswitch-users] digium board
In-Reply-To: <AANLkTi=T5m1c2z-cF8ODpOQagBzqi=05n1BXdNfKHap1@mail.gmail.com>
References: <AANLkTika2=mAOpg2f58v1_rtH=3ur+ueU5+8=7znRO6W@mail.gmail.com>
	<20101019114626.001D211F53@mail.nstel.ru>
	<AANLkTi=T5m1c2z-cF8ODpOQagBzqi=05n1BXdNfKHap1@mail.gmail.com>
Message-ID: <DE930715-8275-4B95-924A-0988FF87E03C@freeswitch.org>

Why wasn't their api put into FreeTDM?

/b

On Oct 19, 2010, at 7:10 AM, Jo?o Mesquita wrote:

> Keep in mind that there is a Brazilian company called Khomp too. They have good quality hardware at decent price. Might be a good option for a try out.
> 
> mod_khomp is being actively developed and soon to be release under the git head.
> 
> Regards,
> Jo?o Mesquita



From jason at jasonjgw.net  Sun Oct 17 00:08:59 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 17 Oct 2010 18:08:59 +1100
Subject: [Freeswitch-users] Binding to external IP address during system
	boot
References: <87wrpj0wdu.fsf@jdc.jasonjgw.net>
	<686AA79D-D8FE-4A5B-8209-DE8EBF272AF2@gmail.com>
Message-ID: <877hhhuwhg.fsf@jdc.jasonjgw.net>

Steven Ayre <steveayre at gmail.com> writes:

> One if these would do it:
> sofia reload
> sofia profile <name> restart

The latter doesn't solve it, as I noted in my original post, but thank
you for suggesting "sofia reload", which I'll try when next it happens.



From ash at archerdrive.com  Sun Oct 17 19:47:48 2010
From: ash at archerdrive.com (Ash)
Date: Mon, 18 Oct 2010 13:47:48 +1100
Subject: [Freeswitch-users] mod_nibblebill hangs up A-Leg and continues
	dialing B-Leg
Message-ID: <CB3C8C74-B8AF-4503-B449-22CF2E3E63CB@archerdrive.com>

Hi All,

I am trying to setup mod nibble bill to bill based on the B-Leg.  I have followed the wiki but have been unable to resolve my issue.  I first had this issue with 1.0.6 and tried the latest git version and can reproduce the fault on both versions.

When the customers credit is 0 or below the handset I am using to test gets transferred to the hangup destination and the A-Leg is hungup.  After a couple of seconds my mobile will ring and when I answer it there is silence so a one way call leg.  As anyone else seen this behaviour?

This is my dialplan that I am outputting to XML Curl.

<?xml version="1.0"?>
<document type="freeswitch/xml">
  <section name="dialplan" description="Outbound">
    <context name="sip_customerdomain_net_au">
      <extension>
        <condition field="destination_number" expression="^(04xxxxxxxx)$"/>
        <condition field="${toll_mobile}" expression="true">
          <action application="export" data="service_id=113"/>
          <action application="set" data="effective_caller_id_number=613xxxxxxx"/>
          <action application="set" data="privacy_hide_name=1"/>
          <action application="bridge" data="{enable_heartbeat_events=5,nibble_rate=1,nibble_account=113}sofia/gateway/primary/614xxxxxxxxx"/>
        </condition>
      </extension>
    </context>
  </section>
</document>

<?xml version="1.0"?>
<document type="freeswitch/xml">
  <section name="dialplan" description="hangup">
    <context name="default">
      <extension>
        <condition field="destination_number" expression="^(hangup)$">
          <action application="log" data="INFO Credit Exceeeded"/>
          <action application="hangup"/>
        </condition>
      </extension>
    </context>
  </section>
</document>

Here is my nibblebill.conf.xml

<configuration name="nibblebill.conf" description="Nibble Billing">
  <settings>
    <!-- See http://wiki.freeswitch.org/index.php?title=Mod_nibblebill for help with these options -->

    <!-- Information for connecting to your database -->
    <param name="db_username" value="switch"/>
    <param name="db_password" value="XXXXXXXX"/>
    <param name="db_dsn" value="fs_odbc"/>

    <!-- The database table where your CASH column is located -->
    <param name="db_table" value="services"/>

    <!-- The column name where we store the value of the account -->
    <param name="db_column_cash" value="credit"/>

    <!-- The column name for the unique ID identifying the account -->
    <param name="db_column_account" value="id"/>


    <!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e. bill only at end of call) -->
    <param name="global_heartbeat" value="60"/>

    <!-- By default, warn a caller when their balance is at $5.00. You can set this to a negative number. -->
    <param name="lowbal_amt" value="0"/>
    <param name="lowbal_action" value="play ding"/>

    <!-- By default, terminate a caller when their balance hits $0.00. You can set this to a negative number. -->
    <param name="nobal_amt" value="0"/>
    <param name="nobal_action" value="nibble_hangup XML default"/>

    <!-- If a call goes beyond a certain dollar amount, flag or terminate it -->
    <param name="percall_max_amt" value="100"/>
    <param name="percall_action" value="hangup"/>

  </settings>
</configuration>

Thanks in advance.

Ash.

From balabaev.m at gmail.com  Mon Oct 18 03:32:20 2010
From: balabaev.m at gmail.com (Maxim Balabaev)
Date: Mon, 18 Oct 2010 14:32:20 +0400
Subject: [Freeswitch-users] SPA921 Problem Bad Session Description
Message-ID: <AANLkTikfwv=yiorKvGegPmT408xgSOL_8f=gZ=0WnPKV@mail.gmail.com>

I can`t make calls from linksys spa921 because of "400 Bad Session
Description", incoming are ok. pap2t works perfectly. freeswitch is rev from
git trunk. Here comes logs:
   ------------------------------------------------------------------------
recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397:
   ------------------------------------------------------------------------
   INVITE sip:1001 at xxx SIP/2.0
   Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport
   From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
   To: <sip:1001 at xxx>
   Call-ID: 14d793ed-1abf9d0e at xxx
   CSeq: 101 INVITE
   Max-Forwards: 70
   Contact: <sip:022 at xxx:5060>
   Expires: 240
   User-Agent: Linksys/SPA921-5.1.8
   Content-Length: 386
   Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
   Supported: replaces
   Content-Type: application/sdp

   v=0
   o=- 603544 603544 IN IP4 xxx
   s=-
   c=IN IP4 xxx
   t=0 0
   m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101
   a=rtpmap:8 /8000
   a=rtpmap:0 /8000
   a=rtpmap:2 G726-32/8000
   a=rtpmap:4 G723/8000
   a=rtpmap:18 G729/8000
   a=rtpmap:96 G726-40/8000
   a=rtpmap:97 G726-24/8000
   a=rtpmap:98 G726-16/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ptime:30
   a=sendrecv
   ------------------------------------------------------------------------
send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607:
   ------------------------------------------------------------------------
   SIP/2.0 400 Bad Session Description
   Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060
   From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
   To: <sip:1001 at xxx>;tag=ytUD271ypvy6r
   Call-ID: 14d793ed-1abf9d0e at xxx
   CSeq: 101 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18 03-19-16
-0400
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Content-Length: 0
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From mwlucas at blackhelicopters.org  Mon Oct 18 07:33:40 2010
From: mwlucas at blackhelicopters.org (Michael W. Lucas)
Date: Mon, 18 Oct 2010 10:33:40 -0400
Subject: [Freeswitch-users] problem with freeswitch and Cisco 7960
In-Reply-To: <AANLkTi=qXdrG2rbPtv2xsSLMzBVE4diw9krCqoiWSTB2@mail.gmail.com>
References: <20101013134855.GA26190@bewilderbeast.blackhelicopters.org>
	<AANLkTin8o0d3Ki2zz0Syhb=cCs8wVK7FQxAmot=kyw2_@mail.gmail.com>
	<20101013171422.GA27259@bewilderbeast.blackhelicopters.org>
	<AANLkTi=qXdrG2rbPtv2xsSLMzBVE4diw9krCqoiWSTB2@mail.gmail.com>
Message-ID: <20101018143340.GA51516@bewilderbeast.blackhelicopters.org>


Hi,

Sorry for the delay, was called out of town.

I've uploaded a REGISTER sequence from the phone as 
http://pastebin.freeswitch.org/14254

Thanks again,
==ml

On Wed, Oct 13, 2010 at 07:57:24PM -0400, Kristian Kielhofner wrote:
> Michael,
> 
>   Do you have a sip trace that includes the REGISTER sequence from the Cisco?
> 
> On Wed, Oct 13, 2010 at 1:14 PM, Michael W. Lucas
> <mwlucas at blackhelicopters.org> wrote:
> > On Wed, Oct 13, 2010 at 11:59:50AM -0400, Kristian Kielhofner wrote:
> >> Michael,
> >>
> >> ? It looks like something strange is happening to the SIP traffic
> >> between the Cisco and FreeSWITCH. ?You shouldn't have to force
> >> FreeSWITCH to send traffic to a registered user on a specific port,
> >> the Contact: header and/or FreeSWITCH NAT handling take care of that
> >> for you.
> >
> > I thought FreeSWITCH handled all those things automatically, that's
> > one reason I wanted to use it.
> >
> > There's no NAT or firewall between the phone and FreeSWITCH, though.
> > They are on different networks, but only because the VM server farm is
> > across my T1.
> >
> >> ? ?Can you show us a siptrace and console output with "sofia profile
> >> internal siptrace on" and upload it to pastebin.freeswitch.org?
> >
> > Done, as user mwlucas.
> >
> > I appreciate any help you can offer.
> >
> > Thanks,
> > ==ml
> >
> >>
> >> On Wed, Oct 13, 2010 at 9:48 AM, Michael W. Lucas
> >> <mwlucas at blackhelicopters.org> wrote:
> >> > Hi,
> >> >
> >> > I'm using Monday's FreeSWITCH build, with a config originally loaded
> >> > 19 August 2010. ?I'm working my way through the FS 1.0.6 book.
> >> >
> >> > I have two phones, one Zoiper soft phone and one Cisco 7960. ?Both
> >> > register fine, and the Cisco can call Zoiper. ?Zoiper cannot call the
> >> > Cisco, however.
> >> >
> >> > The Cisco phone sends its register requests from UDP port 50790. ?When
> >> > FreeSWITCH sends an INVITE to the Cisco, it tries to contact
> >> > UDP/50790, and the phone returns an ICMP "destination unreachable."
> >> >
> >> > This same phone is also registered to an Asterisk box. ?When I call
> >> > that number, asterisk connects to UDP 5060 on this phone. ?The phone
> >> > replies from a high-numbered port to 5060.
> >> >
> >> > It seems I need some option to tell FreeSWITCH to send the INVITE to
> >> > 5060? ?Or am I missing something?
> >> >
> >> > Thanks,
> >> > ==ml
> >> >
> >> > --
> >> > Michael W. Lucas ? ? ? ?mwlucas at BlackHelicopters.org
> >> > http://www.MichaelWLucas.com/, http://blather.MichaelWLucas.com/
> >> > New book available: Network Flow Analysis
> >> > http://www.networkflowanalysis.com/
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >>
> >>
> >> --
> >> Kristian Kielhofner
> >> http://www.astlinux.org
> >> http://blog.krisk.org
> >> http://www.star2star.com
> >> http://www.submityoursip.com
> >> http://www.voalte.com
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > --
> > Michael W. Lucas ? ? ? ?mwlucas at BlackHelicopters.org
> > http://www.MichaelWLucas.com/, http://blather.MichaelWLucas.com/
> > New book available: Network Flow Analysis
> > http://www.networkflowanalysis.com/
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> 
> -- 
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Michael W. Lucas 	mwlucas at BlackHelicopters.org
http://www.MichaelWLucas.com/, http://blather.MichaelWLucas.com/
New book available: Network Flow Analysis
http://www.networkflowanalysis.com/


From christian.knoblauch at astylos.de  Mon Oct 18 08:15:16 2010
From: christian.knoblauch at astylos.de (Christian Knoblauch)
Date: Mon, 18 Oct 2010 17:15:16 +0200
Subject: [Freeswitch-users] FreeSwitch and Cisco 7941 IP Phone ( INVITE port
	number )
Message-ID: <057bc2928b30a59af5a20d08c3fb5141@astylos.de>


Hello,
?
?
This?is about FreeSwitch 1.0.6 (Windows Build)?and Cisco 7941 IP Phone, and the same issue was recently reported by another subscriber of this list (Michael W. Lucas).
?
Anyhow, I attach traces so that maybe someone can look into it.
?
?
CONTEXT:
?
The 7941 successfull REGISTER in FreeSwitch, and is able to call another softphone. 
The softphone is not able to call the 7941, and this is because the INVITE from FreeSwitch towards 7941?goes to the UDP port that the 7941 was using during REGISTER (high port number, instead of 5060)
?
I compared this with traces for 3CX where all works fine. 3CX sends the INVITE towards 7941 to UDP port 5060, and the 7941 seems to like this :-)
?
?
Please find attached the REGISTER / INVITE traces for booth 3CX and FreeSwitch, and also the FreeSwitch console-output from"sofia profile internal siptrace on"
?
?
Thanks for review !
?
Regards
Christian
?

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From cjbujold at accra.ca  Mon Oct 18 11:39:49 2010
From: cjbujold at accra.ca (Charles Bujold)
Date: Mon, 18 Oct 2010 15:39:49 -0300
Subject: [Freeswitch-users] Looking for configuration example for a Linksys
	PAP2t-NA
Message-ID: <007a01cb6ef3$d8459c20$88d0d460$@accra.ca>

Trying to connect a Linksys PAP2T-NA ATA to Freeswitch and trying to find a
configuration example and settings to make it work.  If anybody has a
suggestion please forward it.  Also do I need to upgrade the firmware of the
Pap2T to the latest version for it to work with Freeswitch (v5.16)?

 

Thanks

 

cjb

 

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From joedotslash at gmail.com  Mon Oct 18 12:20:04 2010
From: joedotslash at gmail.com (joedotslash)
Date: Mon, 18 Oct 2010 12:20:04 -0700
Subject: [Freeswitch-users] FreeSwitch as SBC, no registration?
Message-ID: <C1B9B3DB-066D-4853-AD47-C83139A76211@gmail.com>

Hello all,

I'm looking to implement the following scenario:

Step 1
SIP Server A sends INVITE to port 5060 over IPv6 to FreeSwitch:
2001::1 --udp/tcp--> 2001::2:5060

Step 2
FreeSwitch SIP NATs the INVITE and sends it out IPv4 to SIP Server B on port 6000
1.1.1.1 --udp/tcp--> 1.1.1.2:6000

Step 3
All SIP Server B responses within this SIP dialog are sent back to SIP Server A along the same path.

REGISTER messages WILL NOT be processed by FreeSwitch. They are handle elsewhere.

There is more to this but I want to accomplish this first call flow go from there.

Is this possible with FreeSwitch? Can someone point me toward a resource on how to get this created?

The following looks like a good start but it seems it's setup for users to register. I would prefer not to have this.

http://wiki.freeswitch.org/wiki/SBC_Setup

Thanks!
Joe
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From nbhatti at gmail.com  Tue Oct 19 01:17:17 2010
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Tue, 19 Oct 2010 11:17:17 +0300
Subject: [Freeswitch-users] nibble to update DB after hangup
Message-ID: <AANLkTi=7GU7RXPMod+F7eTOWKROeJp5VVNX4hk1fNHQg@mail.gmail.com>

Hello,
I am using nibble_bill to bill a few calls. Trying to implement a
scenario where I can update the database or CDR? after the call has
been billed. Could not find this info elsewhere. Is there a way we can
achieve this with nibble or do I have to run a script, to check
balance before and after the call, and right after the hangup update
the DB with new info?

-B


From scott at laughingraven.org  Mon Oct 18 17:10:19 2010
From: scott at laughingraven.org (Scott Brown)
Date: Mon, 18 Oct 2010 17:10:19 -0700
Subject: [Freeswitch-users] Conference leg recording
Message-ID: <000001cb6f22$05537d80$0ffa7880$@org>

I'm wondering if there currently is a way to record one leg of a conference
call. I have tried several different methods, but always seem to get the
entire conference.

 

Thanks!

Scott

 

 

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From perry at sipglobalphone.com  Thu Oct 14 16:54:07 2010
From: perry at sipglobalphone.com (Perry P)
Date: Thu, 14 Oct 2010 23:54:07 +0000 (UTC)
Subject: [Freeswitch-users]
	=?utf-8?q?icall=27s_carrier_service_returns_48?=
	=?utf-8?q?0=09Temporarily_Unavailable_for_outbound_INVITE?=
References: <DDAA9F55FC4B43DFB2AA309EA74BDE7E@HandsOff>
	<003501cb4acc$1f77cd80$5e676880$@com>
Message-ID: <loom.20101015T015113-352@post.gmane.org>

This thread might be closed already but I too am having this issue...  I then
started testing phone numbers and it appears that certain routes have an issue.
 I called an AT&T mobile number and got a 480 but I then immediately called a
downtown Atlanta phone number and it went through no problem...  I have an open
ticket with them and have updated with these findings.  And I thought this was
my problem for the longest time...  Hmmmm...
Thanks,
Perry




From timb0311 at hotmail.com  Mon Oct 18 14:29:17 2010
From: timb0311 at hotmail.com (Tim B)
Date: Mon, 18 Oct 2010 17:29:17 -0400
Subject: [Freeswitch-users] Compile on Windows
In-Reply-To: <BAY157-w6D3D334579856A7F1E818DE5A0@phx.gbl>
References: <mailman.11271.1287285783.19150.freeswitch-users@lists.freeswitch.org>,
	<BAY157-w6D3D334579856A7F1E818DE5A0@phx.gbl>
Message-ID: <BAY157-w56107C5E9B290FE432741DDE5A0@phx.gbl>






Thanks guys... I have tried building...
 

Nightly Snapshot of Git Source freeswitch-snapshot.tar.gz 
the official release http://files.freeswitch.org/freeswitch-1.0.6.tar.gz
and git clone with autocrlf=false  
still getting errors and failed builds....

 
========== Rebuild All: 58 succeeded, 76 failed, 18 skipped ==========
 
 
Now is there an specifics I need to set in VS2010 for my c++ options, etc?  I am a c# developer so I don't do much with c++, so I never messed with any of the options there.  I did download the windows 7 sdk so I would have all the windows header files.
 
 
Tim
 
 
 

 



  		 	   		  
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From brian at freeswitch.org  Tue Oct 19 06:21:26 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Oct 2010 08:21:26 -0500
Subject: [Freeswitch-users] Remove rport from Via header
In-Reply-To: <3799D7D4-A663-4C55-A4A5-A63585E413CE@ipeva.fr>
References: <AANLkTiky=9vA5uk-gqKHHjuabF0VQE1tGk0wODPc7Op2@mail.gmail.com>
	<3799D7D4-A663-4C55-A4A5-A63585E413CE@ipeva.fr>
Message-ID: <87E1B1EF-4401-43F5-844B-D629DA1CBB54@freeswitch.org>

Don't think you can remove it on everything.  If a device doesn't understand it why not beat that manufacture over the head to fix their older device thats broken.

/b

On Oct 19, 2010, at 8:04 AM, David Ponzone wrote:

> Nicolas,
> 
> you may try to set sip_via_rport in the dialplan, but I don't know if this will work.
> I suppose it depends on what you are trying to defeat.



From brian at freeswitch.org  Tue Oct 19 06:22:32 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Oct 2010 08:22:32 -0500
Subject: [Freeswitch-users] record on demand for a conference call
In-Reply-To: <AANLkTi=3mHhqtWbLiw+onv8MDAWGLF+0mhKt-M6kiWBe@mail.gmail.com>
References: <6DFF80D26E14467299C2B7789D1D82D6@D810>
	<AANLkTimuurW-3SJX1+JB=rzbgmq1mGd12hQEvzmfD4A6@mail.gmail.com>
	<AANLkTi=3mHhqtWbLiw+onv8MDAWGLF+0mhKt-M6kiWBe@mail.gmail.com>
Message-ID: <A30A05F4-5B20-44E6-971F-993E713112A3@freeswitch.org>

Again what value do you see in recording a conference call with only one person sitting in it?  Once the second person joins the recording will start automatically.

/b

On Oct 19, 2010, at 4:31 AM, Andy Ayers wrote:

> thanks for your interest Michael, it would be to allow the moderator to record an introduction to the call so when it's listened to by others it has context. But actually more useful is to be able to start recording when the moderator wants it to to cut off the usually messy start to a conference call. Is this possible, could I configure a dtmf key press to initiate the recording somehow?
> 
>  
> On 13 October 2010 20:05, Michael Collins <msc at freeswitch.org> wrote:
> Just curious - what value is there to recording the call when only one person is there?
> -MC



From abid_freeswitch at live.com  Tue Oct 19 06:26:27 2010
From: abid_freeswitch at live.com (Abid Saleem)
Date: Tue, 19 Oct 2010 19:26:27 +0600
Subject: [Freeswitch-users] Radius AAA
In-Reply-To: <AANLkTi=P1QH2L5B-HckXvqRgHsQEpEia2X7mA_DDT=DN@mail.gmail.com>
References: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>,
	<AANLkTi=GiQaO-48BP=T8c-A+09sJMKzQH7BX=ewBdB8R@mail.gmail.com>,
	<AANLkTikw8LGDqQki16D30b8JL0eJid2UOTs-bEaZSc2k@mail.gmail.com>,
	<4CADC2E4.90608@gmail.com>,
	<COL109-W528C7DF8FD0E4D203FD866F4570@phx.gbl>,
	<AANLkTi=oXjajMBSn8tJCT8chm6mM8E4XVzXVynDda_+O@mail.gmail.com>,
	<AANLkTi=P1QH2L5B-HckXvqRgHsQEpEia2X7mA_DDT=DN@mail.gmail.com>
Message-ID: <COL109-W43A18E93B3B89A7D57AE61F45B0@phx.gbl>


Dear Tihomir,
Really appreciate your help but I still get the same error. I did the following steps as below and tried to run freeswitch but same issue. Please help. Thanks.

[root at terminus mod_rad_auth]#[root at terminus mod_rad_auth]# updatedb && locate libfreeradius-client.so | grep lib/billing/installers/freeradius-client-1.1.6/lib/.libs/libfreeradius-client.so/billing/installers/freeradius-client-1.1.6/lib/.libs/libfreeradius-client.so.2/billing/installers/freeradius-client-1.1.6/lib/.libs/libfreeradius-client.so.2.0.0/usr/local/lib/libfreeradius-client.so/usr/local/lib/libfreeradius-client.so.2/usr/local/lib/libfreeradius-client.so.2.0.0[root at terminus mod_rad_auth]# cd /usr/local/freeswitch/lib/[root at terminus lib]# pwd/usr/local/freeswitch/lib[root at terminus lib]# lslibfreeswitch.a   libfreeswitch.so.1      libjs.la    libjs.so.1.0.6  libplc4.a   libplds4.solibfreeswitch.la  libfreeswitch.so.1.0.0  libjs.so    libnspr4.a      libplc4.so  pkgconfiglibfreeswitch.so  libjs.a                 libjs.so.1  libnspr4.so     libplds4.a[root at terminus lib]# ln -sf /usr/local/lib/libfreeradius-client.so.2 libfreeradius-client.so[root at terminus lib]# lslibfreeradius-client.so  libfreeswitch.so        libjs.a   libjs.so.1      libnspr4.so  libplds4.alibfreeswitch.a          libfreeswitch.so.1      libjs.la  libjs.so.1.0.6  libplc4.a    libplds4.solibfreeswitch.la         libfreeswitch.so.1.0.0  libjs.so  libnspr4.a      libplc4.so   pkgconfig[root at terminus lib]# ls -l /usr/local/freeswitch/lib/ | grep radiuslrwxrwxrwx 1 root root      40 Oct 19 18:00 libfreeradius-client.so -> /usr/local/lib/libfreeradius-client.so.2
Date: Fri, 15 Oct 2010 19:56:10 +0200
From: tculjaga at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Radius AAA



On Fri, Oct 15, 2010 at 7:32 PM, Michael Collins <msc at freeswitch.org> wrote:

search to make sure that the libfreeradius-client.so.2 file exists somewhere on your system. If it does not then the freeradius-client install was not successful.

-MC

I was short on time .. sorry


so:

1. find your library ... 

updatedb && locate libfreeradius-client.s | grep lib

by default it goes into /usr/local/lib/ but you never know.



2. create a link to the library

cd /usr/local/freeswitch/lib/

ln -sf /usr/local/lib/libfreeradius-client.so.2 libfreeradius-client.so

please make sure you input the correct path to the library in question.



when you list your freeswitch/lib/ directory, you should have something like this:

$ ls -l /usr/local/freeswitch/lib/ | grep radius
lrwxrwxrwx 1 root root      40 Mar  7  2010 libfreeradius-client.so.2 -> /usr/local/lib/libfreeradius-client.so.2

 

the link should be there.

Now you can load mod_rad_auth without any issues.








_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From david.ponzone at ipeva.fr  Tue Oct 19 06:38:03 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 19 Oct 2010 15:38:03 +0200
Subject: [Freeswitch-users] FreeSwitch as SBC, no registration?
In-Reply-To: <C1B9B3DB-066D-4853-AD47-C83139A76211@gmail.com>
References: <C1B9B3DB-066D-4853-AD47-C83139A76211@gmail.com>
Message-ID: <7AB39D2F-ECB7-466C-8643-21D354750AE2@ipeva.fr>

So you just want to receive calls from A (which is a well-known IPv6  
address) and send them to B.

I would then modify the internal profile to be the exact same than  
external, except that I would bind internal to 2001::2:5060 and  
external to 1.1.1.1:5060.
Then you just need a basic dialplan (public.xml) to send to B calls  
received from A.
Of course, some security would be also a plus.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 18/10/2010 ? 21:20, joedotslash a ?crit :

> Hello all,
>
> I'm looking to implement the following scenario:
>
> Step 1
> SIP Server A sends INVITE to port 5060 over IPv6 to FreeSwitch:
> 2001::1 --udp/tcp--> 2001::2:5060
>
> Step 2
> FreeSwitch SIP NATs the INVITE and sends it out IPv4 to SIP Server B  
> on port 6000
> 1.1.1.1 --udp/tcp--> 1.1.1.2:6000
>
> Step 3
> All SIP Server B responses within this SIP dialog are sent back to  
> SIP Server A along the same path.
>
> REGISTER messages WILL NOT be processed by FreeSwitch. They are  
> handle elsewhere.
>
> There is more to this but I want to accomplish this first call flow  
> go from there.
>
> Is this possible with FreeSwitch? Can someone point me toward a  
> resource on how to get this created?
>
> The following looks like a good start but it seems it's setup for  
> users to register. I would prefer not to have this.
>
> http://wiki.freeswitch.org/wiki/SBC_Setup
>
> Thanks!
> Joe
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From tculjaga at gmail.com  Tue Oct 19 06:46:19 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 19 Oct 2010 15:46:19 +0200
Subject: [Freeswitch-users] Radius AAA
In-Reply-To: <COL109-W43A18E93B3B89A7D57AE61F45B0@phx.gbl>
References: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>
	<AANLkTi=GiQaO-48BP=T8c-A+09sJMKzQH7BX=ewBdB8R@mail.gmail.com>
	<AANLkTikw8LGDqQki16D30b8JL0eJid2UOTs-bEaZSc2k@mail.gmail.com>
	<4CADC2E4.90608@gmail.com>
	<COL109-W528C7DF8FD0E4D203FD866F4570@phx.gbl>
	<AANLkTi=oXjajMBSn8tJCT8chm6mM8E4XVzXVynDda_+O@mail.gmail.com>
	<AANLkTi=P1QH2L5B-HckXvqRgHsQEpEia2X7mA_DDT=DN@mail.gmail.com>
	<COL109-W43A18E93B3B89A7D57AE61F45B0@phx.gbl>
Message-ID: <AANLkTi=Cgi4LjDTfZAzvJOSU0u1+OWMsewa=-KOU1JeZ@mail.gmail.com>

>
>
> [root at terminus lib]# ln -sf /usr/local/lib/libfreeradius-client.so.2
> libfreeradius-client.so
>
>
try running this this:
ln -sf /usr/local/lib/libfreeradius-client.so.2 libfreeradius-client.so.2
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From mnhassan at usa.net  Tue Oct 19 06:47:50 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Tue, 19 Oct 2010 19:47:50 +0600
Subject: [Freeswitch-users] nibble to update DB after hangup
In-Reply-To: <AANLkTi=7GU7RXPMod+F7eTOWKROeJp5VVNX4hk1fNHQg@mail.gmail.com>
References: <AANLkTi=7GU7RXPMod+F7eTOWKROeJp5VVNX4hk1fNHQg@mail.gmail.com>
Message-ID: <AANLkTi=hV_7eDnHSP0_Fezz-DDC5OXYbZfO3NBN9MAko@mail.gmail.com>

Nibble bill takes care of deducting the charge from the database as
the call continues, right up to the hang up.

Regards
HASSAN


On 2010-10-19, Muhammad Naseer Bhatti <nbhatti at gmail.com> wrote:
> Hello,
> I am using nibble_bill to bill a few calls. Trying to implement a
> scenario where I can update the database or CDR? after the call has
> been billed. Could not find this info elsewhere. Is there a way we can
> achieve this with nibble or do I have to run a script, to check
> balance before and after the call, and right after the hangup update
> the DB with new info?
>
> -B
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device


From mnhassan at usa.net  Tue Oct 19 06:48:59 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Tue, 19 Oct 2010 19:48:59 +0600
Subject: [Freeswitch-users] digium board
In-Reply-To: <DE930715-8275-4B95-924A-0988FF87E03C@freeswitch.org>
References: <AANLkTika2=mAOpg2f58v1_rtH=3ur+ueU5+8=7znRO6W@mail.gmail.com>
	<20101019114626.001D211F53@mail.nstel.ru>
	<AANLkTi=T5m1c2z-cF8ODpOQagBzqi=05n1BXdNfKHap1@mail.gmail.com>
	<DE930715-8275-4B95-924A-0988FF87E03C@freeswitch.org>
Message-ID: <AANLkTikKnZ+8NSVXTdrKosRcO5Lr5+7W0BHjB5qxL2_Y@mail.gmail.com>

Good question!



On 2010-10-19, Brian West <brian at freeswitch.org> wrote:
> Why wasn't their api put into FreeTDM?
>
> /b
>
> On Oct 19, 2010, at 7:10 AM, Jo?o Mesquita wrote:
>
>> Keep in mind that there is a Brazilian company called Khomp too. They have
>> good quality hardware at decent price. Might be a good option for a try
>> out.
>>
>> mod_khomp is being actively developed and soon to be release under the git
>> head.
>>
>> Regards,
>> Jo?o Mesquita
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device


From dujinfang at gmail.com  Tue Oct 19 06:50:48 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 19 Oct 2010 21:50:48 +0800
Subject: [Freeswitch-users] Conference leg recording
In-Reply-To: <000001cb6f22$05537d80$0ffa7880$@org>
References: <000001cb6f22$05537d80$0ffa7880$@org>
Message-ID: <AANLkTinqGS8kkhdN6XTd862PCoheZ4sobRNzB90AchnP@mail.gmail.com>

It might be possible if you want to only record one-way.

make a new leg, the other end of the new leg can be the record app,
e.g.  conference dial loopback/record and set a diaplan to run the
record app.

then use "conference relate" to make sure the new leg can only hear
the leg you want to record.


On Tue, Oct 19, 2010 at 8:10 AM, Scott Brown <scott at laughingraven.org> wrote:
> I?m wondering if there currently is a way to record one leg of a conference
> call. I have tried several different methods, but always seem to get the
> entire conference.
>
>
>
> Thanks!
>
> Scott
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From brian at freeswitch.org  Tue Oct 19 06:40:57 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Oct 2010 08:40:57 -0500
Subject: [Freeswitch-users] Binding to external IP address during system
	boot
In-Reply-To: <87wrpj0wdu.fsf@jdc.jasonjgw.net>
References: <87wrpj0wdu.fsf@jdc.jasonjgw.net>
Message-ID: <7B07DBCC-65FA-4046-AC7A-E00D5F03C72E@freeswitch.org>

Are you on the latest FS GIT?

/b

On Oct 15, 2010, at 8:17 PM, Jason White wrote:

> 
> Clearly, it has bound to the loopback interface. Restarting the sip
> profile doesn't fix it, but restarting FreeSWITCH does.



From bcxml at hotmail.com  Tue Oct 19 06:57:17 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Tue, 19 Oct 2010 09:57:17 -0400
Subject: [Freeswitch-users] Compile on Windows
In-Reply-To: <BAY157-w56107C5E9B290FE432741DDE5A0@phx.gbl>
References: <mailman.11271.1287285783.19150.freeswitch-users@lists.freeswitch.org>, ,
	<BAY157-w6D3D334579856A7F1E818DE5A0@phx.gbl>,
	<BAY157-w56107C5E9B290FE432741DDE5A0@phx.gbl>
Message-ID: <SNT135-w44528AFDB4D0979738878EB75B0@phx.gbl>


Just went through a similar exercise...finally got it to work with the 1.0.6 release
 
The only way I could get a clean compile was VS2008 running on a Windows Server 2003 SP2 box
 
Hope that helps
 
 
Brian
 


From: timb0311 at hotmail.com
To: freeswitch-users at lists.freeswitch.org
Date: Mon, 18 Oct 2010 17:29:17 -0400
Subject: Re: [Freeswitch-users] Compile on Windows







Thanks guys... I have tried building...
 

Nightly Snapshot of Git Source freeswitch-snapshot.tar.gz 
the official release http://files.freeswitch.org/freeswitch-1.0.6.tar.gz
and git clone with autocrlf=false  
still getting errors and failed builds....

 
========== Rebuild All: 58 succeeded, 76 failed, 18 skipped ==========
 
 
Now is there an specifics I need to set in VS2010 for my c++ options, etc?  I am a c# developer so I don't do much with c++, so I never messed with any of the options there.  I did download the windows 7 sdk so I would have all the windows header files.
 
 
Tim
 
 
 

 



 
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 		 	   		  
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From kond at nstel.ru  Tue Oct 19 07:02:55 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Tue, 19 Oct 2010 18:02:55 +0400
Subject: [Freeswitch-users] SPA921 Problem Bad Session Description
In-Reply-To: <AANLkTikfwv=yiorKvGegPmT408xgSOL_8f=gZ=0WnPKV@mail.gmail.com>
Message-ID: <20101019140255.A69141143E@mail.nstel.ru>

I'm not sure, but 5.1.8 firmware on linksys is quite outdated, isn't it?
Nikolay.

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Maxim
Balabaev
Sent: Monday, October 18, 2010 2:32 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] SPA921 Problem Bad Session Description


I can`t make calls from linksys spa921 because of "400 Bad Session
Description", incoming are ok. pap2t works perfectly. freeswitch is rev from
git trunk. Here comes logs: 
   ------------------------------------------------------------------------
recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397:
   ------------------------------------------------------------------------
   INVITE sip:1001 at xxx SIP/2.0
   Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport
   From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
   To: <sip:1001 at xxx>
   Call-ID: 14d793ed-1abf9d0e at xxx
   CSeq: 101 INVITE
   Max-Forwards: 70
   Contact: <sip:022 at xxx:5060>
   Expires: 240
   User-Agent: Linksys/SPA921-5.1.8
   Content-Length: 386
   Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
   Supported: replaces
   Content-Type: application/sdp
   
   v=0
   o=- 603544 603544 IN IP4 xxx
   s=-
   c=IN IP4 xxx
   t=0 0
   m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101
   a=rtpmap:8 /8000
   a=rtpmap:0 /8000
   a=rtpmap:2 G726-32/8000
   a=rtpmap:4 G723/8000
   a=rtpmap:18 G729/8000
   a=rtpmap:96 G726-40/8000
   a=rtpmap:97 G726-24/8000
   a=rtpmap:98 G726-16/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ptime:30
   a=sendrecv
   ------------------------------------------------------------------------
send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607:
   ------------------------------------------------------------------------
   SIP/2.0 400 Bad Session Description
   Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060
   From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
   To: <sip:1001 at xxx>;tag=ytUD271ypvy6r
   Call-ID: 14d793ed-1abf9d0e at xxx
   CSeq: 101 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18 03-19-16
-0400
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Content-Length: 0
  

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From brian at freeswitch.org  Tue Oct 19 07:14:08 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Oct 2010 09:14:08 -0500
Subject: [Freeswitch-users] SPA921 Problem Bad Session Description
In-Reply-To: <AANLkTikfwv=yiorKvGegPmT408xgSOL_8f=gZ=0WnPKV@mail.gmail.com>
References: <AANLkTikfwv=yiorKvGegPmT408xgSOL_8f=gZ=0WnPKV@mail.gmail.com>
Message-ID: <935E8F10-90C8-481A-AA61-F3A2B5A2135B@freeswitch.org>

The SDP is invalid... I suspect your PAP2 has a properly formatted SDP... Can you not see on the two lines for "a=rtpmap:8 /8000" and "a=rtpmap:0 /8000" are the invalid lines they should be "a=rtpmap:8 PCMA/8000" and "a=rtpmap:0 PCMU/8000" .  We WILL NOT make this work because its invalid.  Its on you to contact the manufacture of your device and have them fix it.

/b

On Oct 18, 2010, at 5:32 AM, Maxim Balabaev wrote:

> I can`t make calls from linksys spa921 because of "400 Bad Session Description", incoming are ok. pap2t works perfectly. freeswitch is rev from git trunk. Here comes logs:
>    ------------------------------------------------------------------------
> recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397:
>    ------------------------------------------------------------------------
>    INVITE sip:1001 at xxx SIP/2.0
>    Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport
>    From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
>    To: <sip:1001 at xxx>
>    Call-ID: 14d793ed-1abf9d0e at xxx
>    CSeq: 101 INVITE
>    Max-Forwards: 70
>    Contact: <sip:022 at xxx:5060>
>    Expires: 240
>    User-Agent: Linksys/SPA921-5.1.8
>    Content-Length: 386
>    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
>    Supported: replaces
>    Content-Type: application/sdp
>    
>    v=0
>    o=- 603544 603544 IN IP4 xxx
>    s=-
>    c=IN IP4 xxx
>    t=0 0
>    m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101
>    a=rtpmap:8 /8000
>    a=rtpmap:0 /8000
>    a=rtpmap:2 G726-32/8000
>    a=rtpmap:4 G723/8000
>    a=rtpmap:18 G729/8000
>    a=rtpmap:96 G726-40/8000
>    a=rtpmap:97 G726-24/8000
>    a=rtpmap:98 G726-16/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-15
>    a=ptime:30
>    a=sendrecv
>    ------------------------------------------------------------------------
> send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607:
>    ------------------------------------------------------------------------
>    SIP/2.0 400 Bad Session Description
>    Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060
>    From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
>    To: <sip:1001 at xxx>;tag=ytUD271ypvy6r
>    Call-ID: 14d793ed-1abf9d0e at xxx
>    CSeq: 101 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18 03-19-16 -0400
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, refer
>    Content-Length: 0
>   
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Tue Oct 19 07:15:40 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Oct 2010 09:15:40 -0500
Subject: [Freeswitch-users] Compile on Windows
In-Reply-To: <SNT135-w44528AFDB4D0979738878EB75B0@phx.gbl>
References: <mailman.11271.1287285783.19150.freeswitch-users@lists.freeswitch.org>,
	, <BAY157-w6D3D334579856A7F1E818DE5A0@phx.gbl>,
	<BAY157-w56107C5E9B290FE432741DDE5A0@phx.gbl>
	<SNT135-w44528AFDB4D0979738878EB75B0@phx.gbl>
Message-ID: <632E5979-68C0-4059-9743-13F2597D5DE8@freeswitch.org>

I'm going to repeat this...  DO NOT USE 1.0.6 please you should be on GIT HEAD as its more stable and will become 1.0.7 at any moment... 

/b

On Oct 19, 2010, at 8:57 AM, Brian Campbell wrote:

> Just went through a similar exercise...finally got it to work with the 1.0.6 release
>  
> The only way I could get a clean compile was VS2008 running on a Windows Server 2003 SP2 box
>  
> Hope that helps
>  
>  
> Brian
>  
> From: timb0311 at hotmail.com
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 18 Oct 2010 17:29:17 -0400
> Subject: Re: [Freeswitch-users] Compile on Windows
> 
> 
> Thanks guys... I have tried building...
>  
> ? Nightly Snapshot of Git Source freeswitch-snapshot.tar.gz
> ? the official release http://files.freeswitch.org/freeswitch-1.0.6.tar.gz
> ? and git clone with autocrlf=false
>  
> still getting errors and failed builds....
> 
>  
> ========== Rebuild All: 58 succeeded, 76 failed, 18 skipped ==========
>  
>  
> Now is there an specifics I need to set in VS2010 for my c++ options, etc?  I am a c# developer so I don't do much with c++, so I never messed with any of the options there.  I did download the windows 7 sdk so I would have all the windows header files.
>  
>  
> Tim
>  
>  
>  
> 
>  
> 
>  
> 
> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Tue Oct 19 07:17:05 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Oct 2010 09:17:05 -0500
Subject: [Freeswitch-users] FreeSwitch as SBC, no registration?
In-Reply-To: <7AB39D2F-ECB7-466C-8643-21D354750AE2@ipeva.fr>
References: <C1B9B3DB-066D-4853-AD47-C83139A76211@gmail.com>
	<7AB39D2F-ECB7-466C-8643-21D354750AE2@ipeva.fr>
Message-ID: <1564036B-793A-491C-95B7-A1A72A8B29DF@freeswitch.org>

FreeSWITCH will always process the register request... Currently their is NO way to pass it thru.  See jira their is a bounty on adding this functionality.

/b

On Oct 19, 2010, at 8:38 AM, David Ponzone wrote:

> So you just want to receive calls from A (which is a well-known IPv6 address) and send them to B.
> 
> I would then modify the internal profile to be the exact same than external, except that I would bind internal to 2001::2:5060 and external to 1.1.1.1:5060.
> Then you just need a basic dialplan (public.xml) to send to B calls received from A.
> Of course, some security would be also a plus.
> 



From jmesquita at freeswitch.org  Tue Oct 19 07:18:00 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 19 Oct 2010 11:18:00 -0300
Subject: [Freeswitch-users] digium board
In-Reply-To: <AANLkTikKnZ+8NSVXTdrKosRcO5Lr5+7W0BHjB5qxL2_Y@mail.gmail.com>
References: <AANLkTika2=mAOpg2f58v1_rtH=3ur+ueU5+8=7znRO6W@mail.gmail.com>
	<20101019114626.001D211F53@mail.nstel.ru>
	<AANLkTi=T5m1c2z-cF8ODpOQagBzqi=05n1BXdNfKHap1@mail.gmail.com>
	<DE930715-8275-4B95-924A-0988FF87E03C@freeswitch.org>
	<AANLkTikKnZ+8NSVXTdrKosRcO5Lr5+7W0BHjB5qxL2_Y@mail.gmail.com>
Message-ID: <AANLkTimPuBhQ8suFJgx9FXLeODesB3Etgx2nC5Oe7FQG@mail.gmail.com>

Brian, fortunately or unfortunately, business decisions don't go hand in
hand with technical decisions. Khomp is a small (60 employee) brazilian
company starting to learn how to deal with opensource products. They have
made great improvement, mostly because I have been putting pressure, onto
moving forward with FreeSWITCH?. I am and will ever be a big sponsor of the
project and I now have a small participation on the company, therefore,
pushing Khomp even more towards sponsoring FreeSWITCH?. I can bet they are
going to be sponsoring ClueCon next year.

You have been following this process that has been taking over 2 years and a
lot of effort from my own personal time in favor of having more commercial
initiatives join the cause on other parts of the world besides the US and
Europe. We are finally getting somewhere and Khomp has been helping place
FreeSWITCH? on the map in Brazil and soon on the rest of Latin America. They
have put pressure on local branch of Linux Magazine for them to publish a
translated version of MCs article (which in fact was published), I have been
invited several times to several speeches to promote FreeSWITCH? and the
buzz has started to spread in the market.

Khomp comes from the Dialogic world where API is the big secret to be kept
and it is indeed a big cultural change to support opensource products
specially when this market started with Digium, which posed as a big threat
for all hard card manufacturers.

Anyway, this was an attempt to expose a bit of the "whys" on the decision
the company has made and try to improve the way ppl have been looking at
this small company that has great technical expertise and aims at having
high quality products instead of cheap cards with bad design. I congratulate
Sangoma for doing the same despite of us being competition (good competition
for all I care). :-)

I am sorry for the big email but I deeply involved with the FreeSWITCH?
success inside of Khomp and there's a lot at stake with my own career. Safe
to say that we have been having nothing but great experiences with the
development of mod_khomp and the community. The developers are very glad
with Tony and core devs assistance. For that, I thank you.

Regards,
Jo?o Mesquita


On Tue, Oct 19, 2010 at 10:48 AM, Nyamul Hassan <mnhassan at usa.net> wrote:

> Good question!
>
>
>
> On 2010-10-19, Brian West <brian at freeswitch.org> wrote:
> > Why wasn't their api put into FreeTDM?
> >
> > /b
> >
> > On Oct 19, 2010, at 7:10 AM, Jo?o Mesquita wrote:
> >
> >> Keep in mind that there is a Brazilian company called Khomp too. They
> have
> >> good quality hardware at decent price. Might be a good option for a try
> >> out.
> >>
> >> mod_khomp is being actively developed and soon to be release under the
> git
> >> head.
> >>
> >> Regards,
> >> Jo?o Mesquita
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> --
> Sent from my mobile device
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kris at kriskinc.com  Tue Oct 19 07:23:26 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 19 Oct 2010 10:23:26 -0400
Subject: [Freeswitch-users] SPA921 Problem Bad Session Description
In-Reply-To: <AANLkTikfwv=yiorKvGegPmT408xgSOL_8f=gZ=0WnPKV@mail.gmail.com>
References: <AANLkTikfwv=yiorKvGegPmT408xgSOL_8f=gZ=0WnPKV@mail.gmail.com>
Message-ID: <AANLkTikM_FJi=vV1zVf+SBJSL-H7Orf9X8t2e7O581ZM@mail.gmail.com>

Maxim,

  That SDP looks nasty:

   a=rtpmap:8 /8000
   a=rtpmap:0 /8000

  These are static payload types and don't require an rtpmap line (the
"offer" is in the m= line).  However, when you use an rtpmap you must
use the IANA payload type names: PCMU and PCMA in this case.  Either
of these two SDPs would be valid:

   v=0
   o=- 603544 603544 IN IP4 xxx
   s=-
   c=IN IP4 xxx
   t=0 0
   m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101
   a=rtpmap:2 G726-32/8000
   a=rtpmap:4 G723/8000
   a=rtpmap:18 G729/8000
   a=rtpmap:96 G726-40/8000
   a=rtpmap:97 G726-24/8000
   a=rtpmap:98 G726-16/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ptime:30
   a=sendrecv

-or-

   v=0
   o=- 603544 603544 IN IP4 xxx
   s=-
   c=IN IP4 xxx
   t=0 0
   m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:2 G726-32/8000
   a=rtpmap:4 G723/8000
   a=rtpmap:18 G729/8000
   a=rtpmap:96 G726-40/8000
   a=rtpmap:97 G726-24/8000
   a=rtpmap:98 G726-16/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=ptime:30
   a=sendrecv

  Yours isn't :).

http://www.iana.org/assignments/rtp-parameters

On Mon, Oct 18, 2010 at 6:32 AM, Maxim Balabaev <balabaev.m at gmail.com> wrote:
> I can`t make calls from linksys spa921 because of "400 Bad Session
> Description", incoming are ok. pap2t works perfectly. freeswitch is rev from
> git trunk. Here comes logs:
> ?? ------------------------------------------------------------------------
> recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397:
> ?? ------------------------------------------------------------------------
> ?? INVITE sip:1001 at xxx SIP/2.0
> ?? Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport
> ?? From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
> ?? To: <sip:1001 at xxx>
> ?? Call-ID: 14d793ed-1abf9d0e at xxx
> ?? CSeq: 101 INVITE
> ?? Max-Forwards: 70
> ?? Contact: <sip:022 at xxx:5060>
> ?? Expires: 240
> ?? User-Agent: Linksys/SPA921-5.1.8
> ?? Content-Length: 386
> ?? Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> ?? Supported: replaces
> ?? Content-Type: application/sdp
>
> ?? v=0
> ?? o=- 603544 603544 IN IP4 xxx
> ?? s=-
> ?? c=IN IP4 xxx
> ?? t=0 0
> ?? m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101
> ?? a=rtpmap:8 /8000
> ?? a=rtpmap:0 /8000
> ?? a=rtpmap:2 G726-32/8000
> ?? a=rtpmap:4 G723/8000
> ?? a=rtpmap:18 G729/8000
> ?? a=rtpmap:96 G726-40/8000
> ?? a=rtpmap:97 G726-24/8000
> ?? a=rtpmap:98 G726-16/8000
> ?? a=rtpmap:101 telephone-event/8000
> ?? a=fmtp:101 0-15
> ?? a=ptime:30
> ?? a=sendrecv
> ?? ------------------------------------------------------------------------
> send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607:
> ?? ------------------------------------------------------------------------
> ?? SIP/2.0 400 Bad Session Description
> ?? Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060
> ?? From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
> ?? To: <sip:1001 at xxx>;tag=ytUD271ypvy6r
> ?? Call-ID: 14d793ed-1abf9d0e at xxx
> ?? CSeq: 101 INVITE
> ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18 03-19-16
> -0400
> ?? Accept: application/sdp
> ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> ?? Supported: timer, precondition, path, replaces
> ?? Allow-Events: talk, hold, refer
> ?? Content-Length: 0
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From bcxml at hotmail.com  Tue Oct 19 07:23:31 2010
From: bcxml at hotmail.com (Brian Campbell)
Date: Tue, 19 Oct 2010 10:23:31 -0400
Subject: [Freeswitch-users] Compile on Windows
In-Reply-To: <632E5979-68C0-4059-9743-13F2597D5DE8@freeswitch.org>
References: <mailman.11271.1287285783.19150.freeswitch-users@lists.freeswitch.org>, ,
	, <BAY157-w6D3D334579856A7F1E818DE5A0@phx.gbl>, ,
	<BAY157-w56107C5E9B290FE432741DDE5A0@phx.gbl>,
	<SNT135-w44528AFDB4D0979738878EB75B0@phx.gbl>,
	<632E5979-68C0-4059-9743-13F2597D5DE8@freeswitch.org>
Message-ID: <SNT135-w34F309338D85576D15C9E1B75B0@phx.gbl>


Thanks...I will give it a try
 
 
Brian
 
> From: brian at freeswitch.org
> Date: Tue, 19 Oct 2010 09:15:40 -0500
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Compile on Windows
> 
> I'm going to repeat this... DO NOT USE 1.0.6 please you should be on GIT HEAD as its more stable and will become 1.0.7 at any moment... 
> 
> /b
> 
> On Oct 19, 2010, at 8:57 AM, Brian Campbell wrote:
> 
> > Just went through a similar exercise...finally got it to work with the 1.0.6 release
> > 
> > The only way I could get a clean compile was VS2008 running on a Windows Server 2003 SP2 box
> > 
> > Hope that helps
> > 
> > 
> > Brian
> > 
> > From: timb0311 at hotmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Mon, 18 Oct 2010 17:29:17 -0400
> > Subject: Re: [Freeswitch-users] Compile on Windows
> > 
> > 
> > Thanks guys... I have tried building...
> > 
> > ? Nightly Snapshot of Git Source freeswitch-snapshot.tar.gz
> > ? the official release http://files.freeswitch.org/freeswitch-1.0.6.tar.gz
> > ? and git clone with autocrlf=false
> > 
> > still getting errors and failed builds....
> > 
> > 
> > ========== Rebuild All: 58 succeeded, 76 failed, 18 skipped ==========
> > 
> > 
> > Now is there an specifics I need to set in VS2010 for my c++ options, etc? I am a c# developer so I don't do much with c++, so I never messed with any of the options there. I did download the windows 7 sdk so I would have all the windows header files.
> > 
> > 
> > Tim
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
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From nico at clickfono.com  Tue Oct 19 07:24:53 2010
From: nico at clickfono.com (Nicolas Brenner)
Date: Tue, 19 Oct 2010 11:24:53 -0300
Subject: [Freeswitch-users] Remove rport from Via header
In-Reply-To: <3799D7D4-A663-4C55-A4A5-A63585E413CE@ipeva.fr>
References: <AANLkTiky=9vA5uk-gqKHHjuabF0VQE1tGk0wODPc7Op2@mail.gmail.com>
	<3799D7D4-A663-4C55-A4A5-A63585E413CE@ipeva.fr>
Message-ID: <AANLkTimiwbf4yRKERy7uQ_gqhW3+N6Zg_Xpmd+J4giGM@mail.gmail.com>

Didn't work. My problem is the following: I have a VoIP provider that when
it gets an INVITE from FreeSWITCH, instead of replying to the originating
port (5080), it replies to a random port, and of course FreeSWITCH doesn't
get the reply, so it keeps sending the same INVITE until it times out, even
though the call is actually made and the call status is reported by the
provider, although to some random port where nothing is listening. I talked
to the provider and they are not going to change any config, but told me it
works fine with Asterisk and told me to "disable rport and nat". Only
there's no explicit option to disable nat (like nat=no on Asterisk's
sip.conf) nor to remove the rport parameter from the Via header (which I'm
trying to figure out if it would solve the problem and make the provider's
SIP server reply to the originating port).



2010/10/19 David Ponzone <david.ponzone at ipeva.fr>

> Nicolas,
>
> you may try to set sip_via_rport in the dialplan, but I don't know if this
> will work.
> I suppose it depends on what you are trying to defeat.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 19/10/2010 ? 14:51, Nicolas Brenner a ?crit :
>
> Hello, I need to either remove the rport parameter from the Via header on
> an initial invite or set its value to a specific port. I've looked through
> the mailing archives and the wiki but I haven't found anything. I appreciate
> any help or pointers I can get, thanks.
>
>
> Nico
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From balabaev.m at gmail.com  Tue Oct 19 07:26:02 2010
From: balabaev.m at gmail.com (Maxim Balabaev)
Date: Tue, 19 Oct 2010 18:26:02 +0400
Subject: [Freeswitch-users] SPA921 Problem Bad Session Description
In-Reply-To: <935E8F10-90C8-481A-AA61-F3A2B5A2135B@freeswitch.org>
References: <AANLkTikfwv=yiorKvGegPmT408xgSOL_8f=gZ=0WnPKV@mail.gmail.com>
	<935E8F10-90C8-481A-AA61-F3A2B5A2135B@freeswitch.org>
Message-ID: <AANLkTi=5xFK-HEdYHK8QuCFththWs4FvdJObLZcUE=LF@mail.gmail.com>

Problem was that fields with some codecs names were blank. Thats why there
were "a=rtpmap:0 /8000". Also FS don`t have aliase betwen PCMA and G711.a
for ex.
>> I'm not sure, but 5.1.8 firmware on linksys is quite outdated, isn't it?
It`s really old but this is last one on their website.

2010/10/19 Brian West <brian at freeswitch.org>

> The SDP is invalid... I suspect your PAP2 has a properly formatted SDP...
> Can you not see on the two lines for "a=rtpmap:8 /8000" and "a=rtpmap:0
> /8000" are the invalid lines they should be "a=rtpmap:8 PCMA/8000" and
> "a=rtpmap:0 PCMU/8000" .  We WILL NOT make this work because its invalid.
>  Its on you to contact the manufacture of your device and have them fix it.
>
> /b
>
> On Oct 18, 2010, at 5:32 AM, Maxim Balabaev wrote:
>
> > I can`t make calls from linksys spa921 because of "400 Bad Session
> Description", incoming are ok. pap2t works perfectly. freeswitch is rev from
> git trunk. Here comes logs:
> >
>  ------------------------------------------------------------------------
> > recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397:
> >
>  ------------------------------------------------------------------------
> >    INVITE sip:1001 at xxx SIP/2.0
> >    Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport
> >    From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
> >    To: <sip:1001 at xxx>
> >    Call-ID: 14d793ed-1abf9d0e at xxx
> >    CSeq: 101 INVITE
> >    Max-Forwards: 70
> >    Contact: <sip:022 at xxx:5060>
> >    Expires: 240
> >    User-Agent: Linksys/SPA921-5.1.8
> >    Content-Length: 386
> >    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> >    Supported: replaces
> >    Content-Type: application/sdp
> >
> >    v=0
> >    o=- 603544 603544 IN IP4 xxx
> >    s=-
> >    c=IN IP4 xxx
> >    t=0 0
> >    m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101
> >    a=rtpmap:8 /8000
> >    a=rtpmap:0 /8000
> >    a=rtpmap:2 G726-32/8000
> >    a=rtpmap:4 G723/8000
> >    a=rtpmap:18 G729/8000
> >    a=rtpmap:96 G726-40/8000
> >    a=rtpmap:97 G726-24/8000
> >    a=rtpmap:98 G726-16/8000
> >    a=rtpmap:101 telephone-event/8000
> >    a=fmtp:101 0-15
> >    a=ptime:30
> >    a=sendrecv
> >
>  ------------------------------------------------------------------------
> > send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607:
> >
>  ------------------------------------------------------------------------
> >    SIP/2.0 400 Bad Session Description
> >    Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060
> >    From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
> >    To: <sip:1001 at xxx>;tag=ytUD271ypvy6r
> >    Call-ID: 14d793ed-1abf9d0e at xxx
> >    CSeq: 101 INVITE
> >    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18
> 03-19-16 -0400
> >    Accept: application/sdp
> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> >    Supported: timer, precondition, path, replaces
> >    Allow-Events: talk, hold, refer
> >    Content-Length: 0
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From testeador01 at gmail.com  Tue Oct 19 07:35:11 2010
From: testeador01 at gmail.com (Milena)
Date: Tue, 19 Oct 2010 09:35:11 -0500
Subject: [Freeswitch-users] Remove rport from Via header
In-Reply-To: <AANLkTimiwbf4yRKERy7uQ_gqhW3+N6Zg_Xpmd+J4giGM@mail.gmail.com>
References: <AANLkTiky=9vA5uk-gqKHHjuabF0VQE1tGk0wODPc7Op2@mail.gmail.com>
	<3799D7D4-A663-4C55-A4A5-A63585E413CE@ipeva.fr>
	<AANLkTimiwbf4yRKERy7uQ_gqhW3+N6Zg_Xpmd+J4giGM@mail.gmail.com>
Message-ID: <AANLkTim0zFN4AcGxObrad9YfYw190GYgah+j1SUiO5T0@mail.gmail.com>

start freeswitch with -nonat

On Tue, Oct 19, 2010 at 9:24 AM, Nicolas Brenner <nico at clickfono.com> wrote:


> Only there's no explicit option to disable nat
>
>
>
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From kris at kriskinc.com  Tue Oct 19 07:38:16 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 19 Oct 2010 10:38:16 -0400
Subject: [Freeswitch-users] Remove rport from Via header
In-Reply-To: <AANLkTimiwbf4yRKERy7uQ_gqhW3+N6Zg_Xpmd+J4giGM@mail.gmail.com>
References: <AANLkTiky=9vA5uk-gqKHHjuabF0VQE1tGk0wODPc7Op2@mail.gmail.com>
	<3799D7D4-A663-4C55-A4A5-A63585E413CE@ipeva.fr>
	<AANLkTimiwbf4yRKERy7uQ_gqhW3+N6Zg_Xpmd+J4giGM@mail.gmail.com>
Message-ID: <AANLkTi=O0DcO4NFV8Z=MBUs5PZwf6e44PM+dBHPoW9ot@mail.gmail.com>

I hate to ask but what sort of provider uses Avaya SES?  For what it's
worth I've done interop with Avaya SES and FS before.  No problems
with rport there...

On Tue, Oct 19, 2010 at 10:24 AM, Nicolas Brenner <nico at clickfono.com> wrote:
> Didn't work. My problem is the following: I have a VoIP provider that when
> it gets an INVITE from FreeSWITCH, instead of replying to the originating
> port (5080), it replies to a random port, and of course FreeSWITCH doesn't
> get the reply, so it keeps sending the same INVITE until it times out, even
> though the call is actually made and the call status is reported by the
> provider, although to some random port where nothing is listening. I talked
> to the provider and they are not going to change any config, but told me it
> works fine with Asterisk and told me to "disable rport and nat". Only
> there's no explicit option to disable nat (like nat=no on Asterisk's
> sip.conf) nor to remove the rport parameter from the Via header (which I'm
> trying to figure out if it would solve the problem and make the provider's
> SIP server reply to the originating port).
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From anthony.minessale at gmail.com  Tue Oct 19 07:37:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 19 Oct 2010 09:37:51 -0500
Subject: [Freeswitch-users] SPA921 Problem Bad Session Description
In-Reply-To: <AANLkTikM_FJi=vV1zVf+SBJSL-H7Orf9X8t2e7O581ZM@mail.gmail.com>
References: <AANLkTikfwv=yiorKvGegPmT408xgSOL_8f=gZ=0WnPKV@mail.gmail.com>
	<AANLkTikM_FJi=vV1zVf+SBJSL-H7Orf9X8t2e7O581ZM@mail.gmail.com>
Message-ID: <AANLkTi=mh=b6NPDBmdHztYtRY4x95OWN5VrXYQE28be9@mail.gmail.com>

We already told him this when he filed a JIRA on it.
It's clearly broken and probably fixed with a firmware update to the device.


On Tue, Oct 19, 2010 at 9:23 AM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> Maxim,
>
> ?That SDP looks nasty:
>
> ? a=rtpmap:8 /8000
> ? a=rtpmap:0 /8000
>
> ?These are static payload types and don't require an rtpmap line (the
> "offer" is in the m= line). ?However, when you use an rtpmap you must
> use the IANA payload type names: PCMU and PCMA in this case. ?Either
> of these two SDPs would be valid:
>
> ? v=0
> ? o=- 603544 603544 IN IP4 xxx
> ? s=-
> ? c=IN IP4 xxx
> ? t=0 0
> ? m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101
> ? a=rtpmap:2 G726-32/8000
> ? a=rtpmap:4 G723/8000
> ? a=rtpmap:18 G729/8000
> ? a=rtpmap:96 G726-40/8000
> ? a=rtpmap:97 G726-24/8000
> ? a=rtpmap:98 G726-16/8000
> ? a=rtpmap:101 telephone-event/8000
> ? a=fmtp:101 0-15
> ? a=ptime:30
> ? a=sendrecv
>
> -or-
>
> ? v=0
> ? o=- 603544 603544 IN IP4 xxx
> ? s=-
> ? c=IN IP4 xxx
> ? t=0 0
> ? m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101
> ? a=rtpmap:8 PCMA/8000
> ? a=rtpmap:0 PCMU/8000
> ? a=rtpmap:2 G726-32/8000
> ? a=rtpmap:4 G723/8000
> ? a=rtpmap:18 G729/8000
> ? a=rtpmap:96 G726-40/8000
> ? a=rtpmap:97 G726-24/8000
> ? a=rtpmap:98 G726-16/8000
> ? a=rtpmap:101 telephone-event/8000
> ? a=fmtp:101 0-15
> ? a=ptime:30
> ? a=sendrecv
>
> ?Yours isn't :).
>
> http://www.iana.org/assignments/rtp-parameters
>
> On Mon, Oct 18, 2010 at 6:32 AM, Maxim Balabaev <balabaev.m at gmail.com> wrote:
>> I can`t make calls from linksys spa921 because of "400 Bad Session
>> Description", incoming are ok. pap2t works perfectly. freeswitch is rev from
>> git trunk. Here comes logs:
>> ?? ------------------------------------------------------------------------
>> recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397:
>> ?? ------------------------------------------------------------------------
>> ?? INVITE sip:1001 at xxx SIP/2.0
>> ?? Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport
>> ?? From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
>> ?? To: <sip:1001 at xxx>
>> ?? Call-ID: 14d793ed-1abf9d0e at xxx
>> ?? CSeq: 101 INVITE
>> ?? Max-Forwards: 70
>> ?? Contact: <sip:022 at xxx:5060>
>> ?? Expires: 240
>> ?? User-Agent: Linksys/SPA921-5.1.8
>> ?? Content-Length: 386
>> ?? Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
>> ?? Supported: replaces
>> ?? Content-Type: application/sdp
>>
>> ?? v=0
>> ?? o=- 603544 603544 IN IP4 xxx
>> ?? s=-
>> ?? c=IN IP4 xxx
>> ?? t=0 0
>> ?? m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101
>> ?? a=rtpmap:8 /8000
>> ?? a=rtpmap:0 /8000
>> ?? a=rtpmap:2 G726-32/8000
>> ?? a=rtpmap:4 G723/8000
>> ?? a=rtpmap:18 G729/8000
>> ?? a=rtpmap:96 G726-40/8000
>> ?? a=rtpmap:97 G726-24/8000
>> ?? a=rtpmap:98 G726-16/8000
>> ?? a=rtpmap:101 telephone-event/8000
>> ?? a=fmtp:101 0-15
>> ?? a=ptime:30
>> ?? a=sendrecv
>> ?? ------------------------------------------------------------------------
>> send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607:
>> ?? ------------------------------------------------------------------------
>> ?? SIP/2.0 400 Bad Session Description
>> ?? Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060
>> ?? From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
>> ?? To: <sip:1001 at xxx>;tag=ytUD271ypvy6r
>> ?? Call-ID: 14d793ed-1abf9d0e at xxx
>> ?? CSeq: 101 INVITE
>> ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18 03-19-16
>> -0400
>> ?? Accept: application/sdp
>> ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>> ?? Supported: timer, precondition, path, replaces
>> ?? Allow-Events: talk, hold, refer
>> ?? Content-Length: 0
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From anthony.minessale at gmail.com  Tue Oct 19 07:45:10 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 19 Oct 2010 09:45:10 -0500
Subject: [Freeswitch-users] Remove rport from Via header
In-Reply-To: <AANLkTimiwbf4yRKERy7uQ_gqhW3+N6Zg_Xpmd+J4giGM@mail.gmail.com>
References: <AANLkTiky=9vA5uk-gqKHHjuabF0VQE1tGk0wODPc7Op2@mail.gmail.com>
	<3799D7D4-A663-4C55-A4A5-A63585E413CE@ipeva.fr>
	<AANLkTimiwbf4yRKERy7uQ_gqhW3+N6Zg_Xpmd+J4giGM@mail.gmail.com>
Message-ID: <AANLkTimKgR_VMYM_ijxihVC97qKHVsD+fLu59i-0FB+5@mail.gmail.com>

If the top of their standards are to make it work with asterisk, they
are clearly not very wise.

Are you using some kind of asymmetric nat? Why is it getting a random port?
Even with rport enabled it should be the true port the packet is seen
coming from and a return path should be expected the same way.

Check your router for asymmetric nat or SIP ALG features that you can
disable in it's config.



On Tue, Oct 19, 2010 at 9:24 AM, Nicolas Brenner <nico at clickfono.com> wrote:
> Didn't work. My problem is the following: I have a VoIP provider that when
> it gets an INVITE from FreeSWITCH, instead of replying to the originating
> port (5080), it replies to a random port, and of course FreeSWITCH doesn't
> get the reply, so it keeps sending the same INVITE until it times out, even
> though the call is actually made and the call status is reported by the
> provider, although to some random port where nothing is listening. I talked
> to the provider and they are not going to change any config, but told me it
> works fine with Asterisk and told me to "disable rport and nat". Only
> there's no explicit option to disable nat (like nat=no on Asterisk's
> sip.conf) nor to remove the rport parameter from the Via header (which I'm
> trying to figure out if it would solve the problem and make the provider's
> SIP server reply to the originating port).
>
>
> 2010/10/19 David Ponzone <david.ponzone at ipeva.fr>
>>
>> Nicolas,
>> you may try to set sip_via_rport in the dialplan, but I don't know if this
>> will work.
>> I suppose it depends on what you are trying to defeat.
>> David Ponzone ?Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel: ? ? ?01 74 03 18 97
>> gsm: ? 06 66 98 76 34
>> Service Client?IPeva
>> tel: ? ? ?0811 46 26 26
>> www.ipeva.fr? -? ?www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>> Le 19/10/2010 ? 14:51, Nicolas Brenner a ?crit :
>>
>> Hello, I need to either remove the rport parameter from the Via header on
>> an initial invite or set its value to a specific port. I've looked through
>> the mailing archives and the wiki but I haven't found anything. I appreciate
>> any help or pointers I can get, thanks.
>>
>> Nico
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From anthony.minessale at gmail.com  Tue Oct 19 07:49:08 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 19 Oct 2010 09:49:08 -0500
Subject: [Freeswitch-users] SPA921 Problem Bad Session Description
In-Reply-To: <AANLkTi=5xFK-HEdYHK8QuCFththWs4FvdJObLZcUE=LF@mail.gmail.com>
References: <AANLkTikfwv=yiorKvGegPmT408xgSOL_8f=gZ=0WnPKV@mail.gmail.com>
	<935E8F10-90C8-481A-AA61-F3A2B5A2135B@freeswitch.org>
	<AANLkTi=5xFK-HEdYHK8QuCFththWs4FvdJObLZcUE=LF@mail.gmail.com>
Message-ID: <AANLkTin11RQJfqZnu82SOQ6w-cMZOF3cY_MKK=h-G=ST@mail.gmail.com>

set this param in your sofia profile
NDLB-allow-bad-iananame=true

Then you can use any name you want for the name.
but you can't use blank or the low level sip stack will reject it.

There is no good in supporting aliases. SIP is fluid enough and the
standards that actually work should be preserved.

On Tue, Oct 19, 2010 at 9:26 AM, Maxim Balabaev <balabaev.m at gmail.com> wrote:
> Problem was that fields with some codecs names were blank. Thats why there
> were?"a=rtpmap:0 /8000". Also FS don`t have aliase betwen PCMA and G711.a
> for ex.
>>> I'm not sure, but 5.1.8 firmware on linksys is quite?outdated, isn't it?
> It`s really old but this is last one on their website.
> 2010/10/19 Brian West <brian at freeswitch.org>
>>
>> The SDP is invalid... I suspect your PAP2 has a properly formatted SDP...
>> Can you not see on the two lines for "a=rtpmap:8 /8000" and "a=rtpmap:0
>> /8000" are the invalid lines they should be "a=rtpmap:8 PCMA/8000" and
>> "a=rtpmap:0 PCMU/8000" . ?We WILL NOT make this work because its invalid.
>> ?Its on you to contact the manufacture of your device and have them fix it.
>>
>> /b
>>
>> On Oct 18, 2010, at 5:32 AM, Maxim Balabaev wrote:
>>
>> > I can`t make calls from linksys spa921 because of "400 Bad Session
>> > Description", incoming are ok. pap2t works perfectly. freeswitch is rev from
>> > git trunk. Here comes logs:
>> >
>> > ?------------------------------------------------------------------------
>> > recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397:
>> >
>> > ?------------------------------------------------------------------------
>> > ? ?INVITE sip:1001 at xxx SIP/2.0
>> > ? ?Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport
>> > ? ?From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
>> > ? ?To: <sip:1001 at xxx>
>> > ? ?Call-ID: 14d793ed-1abf9d0e at xxx
>> > ? ?CSeq: 101 INVITE
>> > ? ?Max-Forwards: 70
>> > ? ?Contact: <sip:022 at xxx:5060>
>> > ? ?Expires: 240
>> > ? ?User-Agent: Linksys/SPA921-5.1.8
>> > ? ?Content-Length: 386
>> > ? ?Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
>> > ? ?Supported: replaces
>> > ? ?Content-Type: application/sdp
>> >
>> > ? ?v=0
>> > ? ?o=- 603544 603544 IN IP4 xxx
>> > ? ?s=-
>> > ? ?c=IN IP4 xxx
>> > ? ?t=0 0
>> > ? ?m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101
>> > ? ?a=rtpmap:8 /8000
>> > ? ?a=rtpmap:0 /8000
>> > ? ?a=rtpmap:2 G726-32/8000
>> > ? ?a=rtpmap:4 G723/8000
>> > ? ?a=rtpmap:18 G729/8000
>> > ? ?a=rtpmap:96 G726-40/8000
>> > ? ?a=rtpmap:97 G726-24/8000
>> > ? ?a=rtpmap:98 G726-16/8000
>> > ? ?a=rtpmap:101 telephone-event/8000
>> > ? ?a=fmtp:101 0-15
>> > ? ?a=ptime:30
>> > ? ?a=sendrecv
>> >
>> > ?------------------------------------------------------------------------
>> > send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607:
>> >
>> > ?------------------------------------------------------------------------
>> > ? ?SIP/2.0 400 Bad Session Description
>> > ? ?Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060
>> > ? ?From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
>> > ? ?To: <sip:1001 at xxx>;tag=ytUD271ypvy6r
>> > ? ?Call-ID: 14d793ed-1abf9d0e at xxx
>> > ? ?CSeq: 101 INVITE
>> > ? ?User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18
>> > 03-19-16 -0400
>> > ? ?Accept: application/sdp
>> > ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> > REGISTER, REFER, NOTIFY
>> > ? ?Supported: timer, precondition, path, replaces
>> > ? ?Allow-Events: talk, hold, refer
>> > ? ?Content-Length: 0
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From anthony.minessale at gmail.com  Tue Oct 19 07:54:15 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 19 Oct 2010 09:54:15 -0500
Subject: [Freeswitch-users] digium board
In-Reply-To: <AANLkTimPuBhQ8suFJgx9FXLeODesB3Etgx2nC5Oe7FQG@mail.gmail.com>
References: <AANLkTika2=mAOpg2f58v1_rtH=3ur+ueU5+8=7znRO6W@mail.gmail.com>
	<20101019114626.001D211F53@mail.nstel.ru>
	<AANLkTi=T5m1c2z-cF8ODpOQagBzqi=05n1BXdNfKHap1@mail.gmail.com>
	<DE930715-8275-4B95-924A-0988FF87E03C@freeswitch.org>
	<AANLkTikKnZ+8NSVXTdrKosRcO5Lr5+7W0BHjB5qxL2_Y@mail.gmail.com>
	<AANLkTimPuBhQ8suFJgx9FXLeODesB3Etgx2nC5Oe7FQG@mail.gmail.com>
Message-ID: <AANLkTi=Kp_D4CDZ6h6kjd9M0CDRNY4+NjVFs7yZiwnbw@mail.gmail.com>

That actually has nothing to do with the question.

Adding a module to FreeTDM is the same amount of exposure as a FS mod,
just at a different scope.

FreeTDM itself also loads it's own modules which can be distributed as
binaries in the same way.  In fact Sangoma has a binary PRI stack
module that they distribute with the driver.  I think the question was
more about the design and why they did not choose to enter the card's
support at a lower level in the FS architecture where it was actually
designed to accommodate TDM cards rather than do it at a higher level
with less abstraction.

I guess I don't really care but I can't resist to correct any
misconception about how modules work in FreeSWITCH or FreeTDM


2010/10/19 Jo?o Mesquita <jmesquita at freeswitch.org>:
> Brian, fortunately or unfortunately, business decisions don't go hand in
> hand with technical decisions. Khomp is a small (60 employee) brazilian
> company starting to learn how to deal with opensource products. They have
> made great improvement, mostly because I have been putting pressure, onto
> moving forward with FreeSWITCH?. I am and will ever be a big sponsor of the
> project and I now have a small participation on the company, therefore,
> pushing Khomp even more towards sponsoring FreeSWITCH?. I can bet they are
> going to be sponsoring ClueCon next year.
> You have been following this process that has been taking over 2 years and a
> lot of effort from my own personal time in favor of having more commercial
> initiatives join the cause on other parts of the world besides the US and
> Europe. We are finally getting somewhere and Khomp has been helping place
> FreeSWITCH? on the map in Brazil and soon on the rest of Latin America. They
> have put pressure on local branch of Linux Magazine for them to publish a
> translated version of MCs article (which in fact was published), I have been
> invited several times to several speeches to promote FreeSWITCH? and the
> buzz has started to spread in the market.
> Khomp comes from the Dialogic world where API is the big secret to be kept
> and it is indeed a big cultural change to support opensource products
> specially when this market started with Digium, which posed as a big threat
> for all hard card manufacturers.
> Anyway, this was an attempt to expose a bit of the "whys" on the decision
> the company has made and try to improve the way ppl have been looking at
> this small company that has great technical expertise and aims at having
> high quality products instead of cheap cards with bad design. I congratulate
> Sangoma for doing the same despite of us being competition (good competition
> for all I care). :-)
> I am sorry for the big email but I deeply involved with the FreeSWITCH?
> success inside of Khomp and there's a lot at stake with my own career. Safe
> to say that we have been having nothing but great experiences with the
> development of mod_khomp and the community. The developers are very glad
> with Tony and core devs assistance. For that, I thank you.
> Regards,
> Jo?o Mesquita
>
>
> On Tue, Oct 19, 2010 at 10:48 AM, Nyamul Hassan <mnhassan at usa.net> wrote:
>>
>> Good question!
>>
>>
>>
>> On 2010-10-19, Brian West <brian at freeswitch.org> wrote:
>> > Why wasn't their api put into FreeTDM?
>> >
>> > /b
>> >
>> > On Oct 19, 2010, at 7:10 AM, Jo?o Mesquita wrote:
>> >
>> >> Keep in mind that there is a Brazilian company called Khomp too. They
>> >> have
>> >> good quality hardware at decent price. Might be a good option for a try
>> >> out.
>> >>
>> >> mod_khomp is being actively developed and soon to be release under the
>> >> git
>> >> head.
>> >>
>> >> Regards,
>> >> Jo?o Mesquita
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> --
>> Sent from my mobile device
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From anthony.minessale at gmail.com  Tue Oct 19 07:58:44 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 19 Oct 2010 09:58:44 -0500
Subject: [Freeswitch-users] Gateway responding to random port (rport
	problem?)
In-Reply-To: <AANLkTi=ErWZP5SEr6p_=rej+xhioBQmxa7cCeg045c3i@mail.gmail.com>
References: <AANLkTi=ErWZP5SEr6p_=rej+xhioBQmxa7cCeg045c3i@mail.gmail.com>
Message-ID: <AANLkTimFmhZKw_boLcvArWBpDcJw-Yz0dN+=bGsG4TcK@mail.gmail.com>

Do you see a common theme that you have several threads open regarding
network topology behaviour?

I think you need to audit your network.


On Mon, Oct 18, 2010 at 11:50 AM, Nicolas Brenner <nico at clickfono.com> wrote:
> Hello everyone,
>
> I'm having an issue with a VoIP provider where the gateway will respond to
> any random port instead of 5080. I sent the corresponding trace to the
> provider and they told me to "disable rport and nat". I removed the auto-nat
> parameters in the corresponding sip profile, but I keep having the same
> issue (I don't think it has anything to do with nat anyway).
> On the trace I noticed that INVITEs sent by FS have an empty rport parameter
> on the Via header, but according to what I've read, this is standard and it
> indicates that the response should be sent to the originating port (5080),
> which is not being done by the gateway ("Avaya SIP Enablement Services").
> How may I go about solving this? should I set the rport parameter to
> something specific (eg. 5080)? or remove it altogether?
>
> Thanks,
>
>
> Nico
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From djbinter at gmail.com  Tue Oct 19 08:23:00 2010
From: djbinter at gmail.com (DJB International)
Date: Tue, 19 Oct 2010 08:23:00 -0700
Subject: [Freeswitch-users] Limit [ERROR] question
In-Reply-To: <AANLkTinecv9E6YnfN+8XfKYoBNWaEu4UpxciPD0izAzj@mail.gmail.com>
References: <AANLkTiktTtMkbBB4m+ErkHTvF_xxNtYV-hsTKoJ22wwJ@mail.gmail.com>
	<AANLkTinecv9E6YnfN+8XfKYoBNWaEu4UpxciPD0izAzj@mail.gmail.com>
Message-ID: <AANLkTin6=rQD3R7LpPG4vR6jJxC=dvjjEmSN5AwJFvQa@mail.gmail.com>

Rupa,

Thank you for your clear response.

This issue happened to be on my production server.  I can reproduce since it
happened almost every day, but only happened to 1-2 calls max out of many
concurrent calls.  Thus, I am not that concern right now until I started to
see more of this error.

I use limit just to keep track of concurrent calls per each gateway similar
to below:

For inbound:
<action application="limit" data="db inbound 5.6.7.8 10000"/>

For outbound:
<action application="limit_execute" data="db outbound 1.2.3.4 10000 bridge
sofia/gateway/1.2.3.4/${destination_number}" />

My next question would be whether it would generate more of this errors if I
routed more calls to this server, and if there is any recommendation on how
to prevent it.

Thank you,
Dorn.



On Tue, Oct 19, 2010 at 5:56 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> This is a "shouldn't happen" sanity check you are tripping over -- I
> probably should have worded the error message better.
>
> Immediate cause is that the list of backends (hash, sql, etc) that have
> registered interest in this call is empty.   This shouldn't happen because
> we register the handler at the same time that we set the channel var of
> interested backends.
>
> Can you find a way to reproduce this reliably?
>
> The effect of this would be to (potentially) not decrement a counter when
> we should have.
>
> On Mon, Oct 18, 2010 at 7:31 PM, DJB International <djbinter at gmail.com>wrote:
>
>> I have noticed once in a while in the log file with the following error:
>>
>> [ERR] switch_limit.c:86 Unset limit backendlist!
>>
>> What exactly would cause this error?
>>
>> Thank you,
>> Dorn B.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From thomas at chaschperli.ch  Tue Oct 19 08:22:59 2010
From: thomas at chaschperli.ch (Thomas Mueller)
Date: Tue, 19 Oct 2010 17:22:59 +0200
Subject: [Freeswitch-users] Compile on Windows
In-Reply-To: <SNT135-w44528AFDB4D0979738878EB75B0@phx.gbl>
References: <mailman.11271.1287285783.19150.freeswitch-users@lists.freeswitch.org>,
	, <BAY157-w6D3D334579856A7F1E818DE5A0@phx.gbl>,
	<BAY157-w56107C5E9B290FE432741DDE5A0@phx.gbl>
	<SNT135-w44528AFDB4D0979738878EB75B0@phx.gbl>
Message-ID: <4CBDB7D3.1090907@chaschperli.ch>


>
> The only way I could get a clean compile was VS2008 running on a 
> Windows Server 2003 SP2 box
>
i've got it compiling without major problems on WinXP with VisualStudio 
Express 2008 yesterday.

only thing to obey was the "autocrlf" thing that screwed up compilation 
of sofia. I checked before the first checkout, the check was not set in 
the TortoiseGui but after the checkout it was set. Deleted the source, 
removed the "autocrlf" check , "git clone" and then all was ok.

- Thomas


From balabaev.m at gmail.com  Tue Oct 19 07:42:46 2010
From: balabaev.m at gmail.com (Maxim Balabaev)
Date: Tue, 19 Oct 2010 18:42:46 +0400
Subject: [Freeswitch-users] Sofia gateway incoming problem
Message-ID: <AANLkTim9Zh6rsopD+Tc0krs7+TT2C08cM+mxg-8zA_Yd@mail.gmail.com>

I`m having really strange problem. I connect new provider, changed all
gateway settings but i can`t take calls. While debuggins external and so as
global no invite packet is showing up. I even tryed tcpdump - absolutely
nothing. Just pings. Outcoming are ok, register is ok. If I switch to
asterisk - no problem occures. Support says that server refuses connection,
while another gateway works perfectly. I don`t know what to do...
May be this can be usefull to. Prov have multiple realms on server and
accepts only @myprov.com format. While to asterisk invite is thrown with
sipid at myip. I tryed changing server hostname to ip, but still nothing.
Here is my conf:
http://pastebin.com/qKqAv38a <http://pastebin.com/mMrj33pN>
Really appritiate any help.
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From balabaev.m at gmail.com  Tue Oct 19 07:53:18 2010
From: balabaev.m at gmail.com (Maxim Balabaev)
Date: Tue, 19 Oct 2010 18:53:18 +0400
Subject: [Freeswitch-users] SPA921 Problem Bad Session Description
In-Reply-To: <AANLkTi=mh=b6NPDBmdHztYtRY4x95OWN5VrXYQE28be9@mail.gmail.com>
References: <AANLkTikfwv=yiorKvGegPmT408xgSOL_8f=gZ=0WnPKV@mail.gmail.com>
	<AANLkTikM_FJi=vV1zVf+SBJSL-H7Orf9X8t2e7O581ZM@mail.gmail.com>
	<AANLkTi=mh=b6NPDBmdHztYtRY4x95OWN5VrXYQE28be9@mail.gmail.com>
Message-ID: <AANLkTikFDoXpegvXqtgwhUgfNsbZez1Ha6tq0cV+8271@mail.gmail.com>

Yes. thank you so much. My prev server didn`t require such accurate headers.
It`s really cool that fs have such detalied debug. it`s not easy to find
that g711x is not acceptible format.

2010/10/19 Anthony Minessale <anthony.minessale at gmail.com>

> We already told him this when he filed a JIRA on it.
> It's clearly broken and probably fixed with a firmware update to the
> device.
>
>
> On Tue, Oct 19, 2010 at 9:23 AM, Kristian Kielhofner <kris at kriskinc.com>
> wrote:
> > Maxim,
> >
> >  That SDP looks nasty:
> >
> >   a=rtpmap:8 /8000
> >   a=rtpmap:0 /8000
> >
> >  These are static payload types and don't require an rtpmap line (the
> > "offer" is in the m= line).  However, when you use an rtpmap you must
> > use the IANA payload type names: PCMU and PCMA in this case.  Either
> > of these two SDPs would be valid:
> >
> >   v=0
> >   o=- 603544 603544 IN IP4 xxx
> >   s=-
> >   c=IN IP4 xxx
> >   t=0 0
> >   m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101
> >   a=rtpmap:2 G726-32/8000
> >   a=rtpmap:4 G723/8000
> >   a=rtpmap:18 G729/8000
> >   a=rtpmap:96 G726-40/8000
> >   a=rtpmap:97 G726-24/8000
> >   a=rtpmap:98 G726-16/8000
> >   a=rtpmap:101 telephone-event/8000
> >   a=fmtp:101 0-15
> >   a=ptime:30
> >   a=sendrecv
> >
> > -or-
> >
> >   v=0
> >   o=- 603544 603544 IN IP4 xxx
> >   s=-
> >   c=IN IP4 xxx
> >   t=0 0
> >   m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101
> >   a=rtpmap:8 PCMA/8000
> >   a=rtpmap:0 PCMU/8000
> >   a=rtpmap:2 G726-32/8000
> >   a=rtpmap:4 G723/8000
> >   a=rtpmap:18 G729/8000
> >   a=rtpmap:96 G726-40/8000
> >   a=rtpmap:97 G726-24/8000
> >   a=rtpmap:98 G726-16/8000
> >   a=rtpmap:101 telephone-event/8000
> >   a=fmtp:101 0-15
> >   a=ptime:30
> >   a=sendrecv
> >
> >  Yours isn't :).
> >
> > http://www.iana.org/assignments/rtp-parameters
> >
> > On Mon, Oct 18, 2010 at 6:32 AM, Maxim Balabaev <balabaev.m at gmail.com>
> wrote:
> >> I can`t make calls from linksys spa921 because of "400 Bad Session
> >> Description", incoming are ok. pap2t works perfectly. freeswitch is rev
> from
> >> git trunk. Here comes logs:
> >>
> ------------------------------------------------------------------------
> >> recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397:
> >>
> ------------------------------------------------------------------------
> >>    INVITE sip:1001 at xxx SIP/2.0
> >>    Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport
> >>    From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
> >>    To: <sip:1001 at xxx>
> >>    Call-ID: 14d793ed-1abf9d0e at xxx
> >>    CSeq: 101 INVITE
> >>    Max-Forwards: 70
> >>    Contact: <sip:022 at xxx:5060>
> >>    Expires: 240
> >>    User-Agent: Linksys/SPA921-5.1.8
> >>    Content-Length: 386
> >>    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> >>    Supported: replaces
> >>    Content-Type: application/sdp
> >>
> >>    v=0
> >>    o=- 603544 603544 IN IP4 xxx
> >>    s=-
> >>    c=IN IP4 xxx
> >>    t=0 0
> >>    m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101
> >>    a=rtpmap:8 /8000
> >>    a=rtpmap:0 /8000
> >>    a=rtpmap:2 G726-32/8000
> >>    a=rtpmap:4 G723/8000
> >>    a=rtpmap:18 G729/8000
> >>    a=rtpmap:96 G726-40/8000
> >>    a=rtpmap:97 G726-24/8000
> >>    a=rtpmap:98 G726-16/8000
> >>    a=rtpmap:101 telephone-event/8000
> >>    a=fmtp:101 0-15
> >>    a=ptime:30
> >>    a=sendrecv
> >>
> ------------------------------------------------------------------------
> >> send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607:
> >>
> ------------------------------------------------------------------------
> >>    SIP/2.0 400 Bad Session Description
> >>    Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060
> >>    From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
> >>    To: <sip:1001 at xxx>;tag=ytUD271ypvy6r
> >>    Call-ID: 14d793ed-1abf9d0e at xxx
> >>    CSeq: 101 INVITE
> >>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18
> 03-19-16
> >> -0400
> >>    Accept: application/sdp
> >>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> REGISTER, REFER, NOTIFY
> >>    Supported: timer, precondition, path, replaces
> >>    Allow-Events: talk, hold, refer
> >>    Content-Length: 0
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
> >
> > --
> > Kristian Kielhofner
> > http://www.astlinux.org
> > http://blog.krisk.org
> > http://www.star2star.com
> > http://www.submityoursip.com
> > http://www.voalte.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From scott at laughingraven.org  Tue Oct 19 08:28:38 2010
From: scott at laughingraven.org (Scott Brown)
Date: Tue, 19 Oct 2010 08:28:38 -0700
Subject: [Freeswitch-users] Conference leg recording
In-Reply-To: <AANLkTinqGS8kkhdN6XTd862PCoheZ4sobRNzB90AchnP@mail.gmail.com>
References: <000001cb6f22$05537d80$0ffa7880$@org>
	<AANLkTinqGS8kkhdN6XTd862PCoheZ4sobRNzB90AchnP@mail.gmail.com>
Message-ID: <000601cb6fa2$4da27a70$e8e76f50$@org>

Thanks! I will give that a go, although for what I'm trying to accomplish
this might be a bit cumbersome. I'm trying to find a simple, automatic way
of recording every conference member individually during a conference. Using
the Relate method, I think I would have to create a "Dummy" loopback user
for ever Conference member, right? Which may not be that big of a deal with
the right dialplan. I'll hammer on it for a while and see what I can come up
with. Thanks again! 


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Seven Du
Sent: Tuesday, October 19, 2010 6:51 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Conference leg recording

It might be possible if you want to only record one-way.

make a new leg, the other end of the new leg can be the record app,
e.g.  conference dial loopback/record and set a diaplan to run the
record app.

then use "conference relate" to make sure the new leg can only hear
the leg you want to record.


On Tue, Oct 19, 2010 at 8:10 AM, Scott Brown <scott at laughingraven.org>
wrote:
> I?m wondering if there currently is a way to record one leg of a
conference
> call. I have tried several different methods, but always seem to get the
> entire conference.
>
>
>
> Thanks!
>
> Scott
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn

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

-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.




From brian at freeswitch.org  Tue Oct 19 08:48:12 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Oct 2010 10:48:12 -0500
Subject: [Freeswitch-users] SPA921 Problem Bad Session Description
In-Reply-To: <AANLkTikFDoXpegvXqtgwhUgfNsbZez1Ha6tq0cV+8271@mail.gmail.com>
References: <AANLkTikfwv=yiorKvGegPmT408xgSOL_8f=gZ=0WnPKV@mail.gmail.com>
	<AANLkTikM_FJi=vV1zVf+SBJSL-H7Orf9X8t2e7O581ZM@mail.gmail.com>
	<AANLkTi=mh=b6NPDBmdHztYtRY4x95OWN5VrXYQE28be9@mail.gmail.com>
	<AANLkTikFDoXpegvXqtgwhUgfNsbZez1Ha6tq0cV+8271@mail.gmail.com>
Message-ID: <5608D696-D7CD-4085-9C2E-15563A42DB17@freeswitch.org>

Its not about it not being acceptable.  Its about your device putting invalid formatting in the SDP.  Has ZERO to do with anything else... your device is busted.

/b

On Oct 19, 2010, at 9:53 AM, Maxim Balabaev wrote:

> Yes. thank you so much. My prev server didn`t require such accurate headers. It`s really cool that fs have such detalied debug. it`s not easy to find that g711x is not acceptible format.



From brian at freeswitch.org  Tue Oct 19 08:57:38 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Oct 2010 10:57:38 -0500
Subject: [Freeswitch-users] Sofia gateway incoming problem
In-Reply-To: <AANLkTim9Zh6rsopD+Tc0krs7+TT2C08cM+mxg-8zA_Yd@mail.gmail.com>
References: <AANLkTim9Zh6rsopD+Tc0krs7+TT2C08cM+mxg-8zA_Yd@mail.gmail.com>
Message-ID: <BF035383-6CC8-4FEB-AA1A-22F5FD82B767@freeswitch.org>

Without sip traces of any other valid information we can only guess...

/b

On Oct 19, 2010, at 9:42 AM, Maxim Balabaev wrote:

> I`m having really strange problem. I connect new provider, changed all gateway settings but i can`t take calls. While debuggins external and so as global no invite packet is showing up. I even tryed tcpdump - absolutely nothing. Just pings. Outcoming are ok, register is ok. If I switch to asterisk - no problem occures. Support says that server refuses connection, while another gateway works perfectly. I don`t know what to do...
> May be this can be usefull to. Prov have multiple realms on server and accepts only @myprov.com format. While to asterisk invite is thrown with sipid at myip. I tryed changing server hostname to ip, but still nothing.
> Here is my conf:
> http://pastebin.com/qKqAv38a
> Really appritiate any help.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From rupa at rupa.com  Tue Oct 19 09:00:16 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 19 Oct 2010 11:00:16 -0500
Subject: [Freeswitch-users] Limit [ERROR] question
In-Reply-To: <AANLkTin6=rQD3R7LpPG4vR6jJxC=dvjjEmSN5AwJFvQa@mail.gmail.com>
References: <AANLkTiktTtMkbBB4m+ErkHTvF_xxNtYV-hsTKoJ22wwJ@mail.gmail.com>
	<AANLkTinecv9E6YnfN+8XfKYoBNWaEu4UpxciPD0izAzj@mail.gmail.com>
	<AANLkTin6=rQD3R7LpPG4vR6jJxC=dvjjEmSN5AwJFvQa@mail.gmail.com>
Message-ID: <AANLkTi=bpNixuziYRCBk-Ap+A8qTx+JsiManGYeKGkfp@mail.gmail.com>

Thanks for the response.  From re-reviewing the code, I can't see how this
error can occur in actual usage.  I don't see a code path where the callback
handler would be called without a corresponding backend set.

Without knowing how this condition is occurring I can't say whether adding
traffic would increase the rate at which you see the error nor how to go
about preventing the error.

One question: you are using the db backend.  I assume this is because you
have a cluster of machines?  If not, maybe try the hash backend and see if
you get different behavior?

The code in question should not depend on the backend but maybe try "just in
case".

On Tue, Oct 19, 2010 at 10:23 AM, DJB International <djbinter at gmail.com>wrote:

> Rupa,
>
> Thank you for your clear response.
>
> This issue happened to be on my production server.  I can reproduce since
> it happened almost every day, but only happened to 1-2 calls max out of many
> concurrent calls.  Thus, I am not that concern right now until I started to
> see more of this error.
>
> I use limit just to keep track of concurrent calls per each gateway similar
> to below:
>
> For inbound:
> <action application="limit" data="db inbound 5.6.7.8 10000"/>
>
> For outbound:
> <action application="limit_execute" data="db outbound 1.2.3.4 10000 bridge
> sofia/gateway/1.2.3.4/${destination_number}<http://1.2.3.4/$%7Bdestination_number%7D>"
> />
>
> My next question would be whether it would generate more of this errors if
> I routed more calls to this server, and if there is any recommendation on
> how to prevent it.
>
> Thank you,
> Dorn.
>
>
>
> On Tue, Oct 19, 2010 at 5:56 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> This is a "shouldn't happen" sanity check you are tripping over -- I
>> probably should have worded the error message better.
>>
>> Immediate cause is that the list of backends (hash, sql, etc) that have
>> registered interest in this call is empty.   This shouldn't happen because
>> we register the handler at the same time that we set the channel var of
>> interested backends.
>>
>> Can you find a way to reproduce this reliably?
>>
>> The effect of this would be to (potentially) not decrement a counter when
>> we should have.
>>
>> On Mon, Oct 18, 2010 at 7:31 PM, DJB International <djbinter at gmail.com>wrote:
>>
>>> I have noticed once in a while in the log file with the following error:
>>>
>>> [ERR] switch_limit.c:86 Unset limit backendlist!
>>>
>>> What exactly would cause this error?
>>>
>>> Thank you,
>>> Dorn B.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From djbinter at gmail.com  Tue Oct 19 09:19:32 2010
From: djbinter at gmail.com (DJB International)
Date: Tue, 19 Oct 2010 09:19:32 -0700
Subject: [Freeswitch-users] Limit [ERROR] question
In-Reply-To: <AANLkTi=bpNixuziYRCBk-Ap+A8qTx+JsiManGYeKGkfp@mail.gmail.com>
References: <AANLkTiktTtMkbBB4m+ErkHTvF_xxNtYV-hsTKoJ22wwJ@mail.gmail.com>
	<AANLkTinecv9E6YnfN+8XfKYoBNWaEu4UpxciPD0izAzj@mail.gmail.com>
	<AANLkTin6=rQD3R7LpPG4vR6jJxC=dvjjEmSN5AwJFvQa@mail.gmail.com>
	<AANLkTi=bpNixuziYRCBk-Ap+A8qTx+JsiManGYeKGkfp@mail.gmail.com>
Message-ID: <AANLkTinBexiCrEw0U18tKdQJwinfsRiuA_q_9D-GVKxE@mail.gmail.com>

Thank you, Rupa.  I will try to change it to hash and see what happen.

Just curiosity, is there any advantage/disadvantage using hash vs. db.  I am
currently not doing any cluster.  I picked mod_db because it was mentioned
first in the Wiki/Limit :)

Dorn.

On Tue, Oct 19, 2010 at 9:00 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> Thanks for the response.  From re-reviewing the code, I can't see how this
> error can occur in actual usage.  I don't see a code path where the callback
> handler would be called without a corresponding backend set.
>
> Without knowing how this condition is occurring I can't say whether adding
> traffic would increase the rate at which you see the error nor how to go
> about preventing the error.
>
> One question: you are using the db backend.  I assume this is because you
> have a cluster of machines?  If not, maybe try the hash backend and see if
> you get different behavior?
>
> The code in question should not depend on the backend but maybe try "just
> in case".
>
>
> On Tue, Oct 19, 2010 at 10:23 AM, DJB International <djbinter at gmail.com>wrote:
>
>> Rupa,
>>
>> Thank you for your clear response.
>>
>> This issue happened to be on my production server.  I can reproduce since
>> it happened almost every day, but only happened to 1-2 calls max out of many
>> concurrent calls.  Thus, I am not that concern right now until I started to
>> see more of this error.
>>
>> I use limit just to keep track of concurrent calls per each gateway
>> similar to below:
>>
>> For inbound:
>> <action application="limit" data="db inbound 5.6.7.8 10000"/>
>>
>> For outbound:
>> <action application="limit_execute" data="db outbound 1.2.3.4 10000 bridge
>> sofia/gateway/1.2.3.4/${destination_number}<http://1.2.3.4/$%7Bdestination_number%7D>"
>> />
>>
>> My next question would be whether it would generate more of this errors if
>> I routed more calls to this server, and if there is any recommendation on
>> how to prevent it.
>>
>> Thank you,
>> Dorn.
>>
>>
>>
>> On Tue, Oct 19, 2010 at 5:56 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>>> This is a "shouldn't happen" sanity check you are tripping over -- I
>>> probably should have worded the error message better.
>>>
>>> Immediate cause is that the list of backends (hash, sql, etc) that have
>>> registered interest in this call is empty.   This shouldn't happen because
>>> we register the handler at the same time that we set the channel var of
>>> interested backends.
>>>
>>> Can you find a way to reproduce this reliably?
>>>
>>> The effect of this would be to (potentially) not decrement a counter when
>>> we should have.
>>>
>>> On Mon, Oct 18, 2010 at 7:31 PM, DJB International <djbinter at gmail.com>wrote:
>>>
>>>> I have noticed once in a while in the log file with the following error:
>>>>
>>>> [ERR] switch_limit.c:86 Unset limit backendlist!
>>>>
>>>> What exactly would cause this error?
>>>>
>>>> Thank you,
>>>> Dorn B.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From curriegrad2004 at gmail.com  Tue Oct 19 09:38:35 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Tue, 19 Oct 2010 09:38:35 -0700
Subject: [Freeswitch-users] Compile on Windows
In-Reply-To: <4CBDB7D3.1090907@chaschperli.ch>
References: <mailman.11271.1287285783.19150.freeswitch-users@lists.freeswitch.org>
	<BAY157-w6D3D334579856A7F1E818DE5A0@phx.gbl>
	<BAY157-w56107C5E9B290FE432741DDE5A0@phx.gbl>
	<SNT135-w44528AFDB4D0979738878EB75B0@phx.gbl>
	<4CBDB7D3.1090907@chaschperli.ch>
Message-ID: <AANLkTimH6W320ODWNr8LRnx20T+cH5-rp+OsweP28JeO@mail.gmail.com>

IIRC, the latest Windows SDK even compiled properly without issues on
my end. Was using Visual Studio 2008 SP1 on a Windows 7 x64 machine
without issues after I obeyed the autocrlf option as off.

On Tue, Oct 19, 2010 at 8:22 AM, Thomas Mueller <thomas at chaschperli.ch> wrote:
>
>>
>> The only way I could get a clean compile was VS2008 running on a
>> Windows Server 2003 SP2 box
>>
> i've got it compiling without major problems on WinXP with VisualStudio
> Express 2008 yesterday.
>
> only thing to obey was the "autocrlf" thing that screwed up compilation
> of sofia. I checked before the first checkout, the check was not set in
> the TortoiseGui but after the checkout it was set. Deleted the source,
> removed the "autocrlf" check , "git clone" and then all was ok.
>
> - Thomas
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mario_fs at mgtech.com  Tue Oct 19 09:40:43 2010
From: mario_fs at mgtech.com (Mario)
Date: Tue, 19 Oct 2010 09:40:43 -0700
Subject: [Freeswitch-users] Early media beginning clipped for certain callers
Message-ID: <4CBDCA0B.5070900@mgtech.com>

I have replaced what the incoming caller hears (ringing) with a
greeting, instructions, ringing. Works great but for some callers, about
1 second is clipped, it seems to do with who is calling:

Caller A cell phone & B land line works every time and is never clipped.
Caller C cell phone & D land line clips every time.

I added up to a 5 second "sleep" before the media to test but it had no
effect telling me the other end is clipping the audio when it starts.
Any ideas on what I might try? I can try to lengthen the audio but I
didn't want to make the working ones listen to extra silence. Thanks!
Mario


From covici at ccs.covici.com  Tue Oct 19 09:49:34 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 19 Oct 2010 12:49:34 -0400
Subject: [Freeswitch-users] transfer problem
Message-ID: <13464.1287506974@ccs.covici.com>

Hi.  I have an ivr which connects to conferences which I have in their
own context rather than the default.  The problem is that if I call
someone else and have a 3-way between the other party and the
conference, its fine till I hang up and then fs tries to transfer to the
correct conference name, but uses default for the context instead.  Is
this expected behavior or should I file a bug?

Thanks.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From covici at ccs.covici.com  Tue Oct 19 09:52:11 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 19 Oct 2010 12:52:11 -0400
Subject: [Freeswitch-users] energy level and conference recording
Message-ID: <13505.1287507131@ccs.covici.com>

Hi.  I have found that if the energy level is not 0 and I try to record
the conference, the recording loses several milliseconds -- maybe up to
20-50 after silence.  I don't think this should happen, but let me know
and I can file a bug or is there another work around?

Thanks in advance for any suggestions.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From balabaev.m at gmail.com  Tue Oct 19 09:48:01 2010
From: balabaev.m at gmail.com (Maxim Balabaev)
Date: Tue, 19 Oct 2010 20:48:01 +0400
Subject: [Freeswitch-users] Sofia gateway incoming problem
In-Reply-To: <BF035383-6CC8-4FEB-AA1A-22F5FD82B767@freeswitch.org>
References: <AANLkTim9Zh6rsopD+Tc0krs7+TT2C08cM+mxg-8zA_Yd@mail.gmail.com>
	<BF035383-6CC8-4FEB-AA1A-22F5FD82B767@freeswitch.org>
Message-ID: <AANLkTi=3n64VgYJbkP2KFDtF0Z_PULES-Ei8Moi+28c6@mail.gmail.com>

Here is asterisk invite dump:
http://pastebin.com/Me53C1kV
(I can't give any FS dump because there is just nothing)
Fs reg dump:
http://pastebin.com/63z5HPAS
May be problem is in contact header? I think i have seen this problem
somewhere...
2010/10/19 Brian West <brian at freeswitch.org>

> Without sip traces of any other valid information we can only guess...
>
> /b
>
> On Oct 19, 2010, at 9:42 AM, Maxim Balabaev wrote:
>
> > I`m having really strange problem. I connect new provider, changed all
> gateway settings but i can`t take calls. While debuggins external and so as
> global no invite packet is showing up. I even tryed tcpdump - absolutely
> nothing. Just pings. Outcoming are ok, register is ok. If I switch to
> asterisk - no problem occures. Support says that server refuses connection,
> while another gateway works perfectly. I don`t know what to do...
> > May be this can be usefull to. Prov have multiple realms on server and
> accepts only @myprov.com format. While to asterisk invite is thrown with
> sipid at myip. I tryed changing server hostname to ip, but still nothing.
> > Here is my conf:
> > http://pastebin.com/qKqAv38a
> > Really appritiate any help.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rupa at rupa.com  Tue Oct 19 09:56:40 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 19 Oct 2010 11:56:40 -0500
Subject: [Freeswitch-users] Limit [ERROR] question
In-Reply-To: <AANLkTinBexiCrEw0U18tKdQJwinfsRiuA_q_9D-GVKxE@mail.gmail.com>
References: <AANLkTiktTtMkbBB4m+ErkHTvF_xxNtYV-hsTKoJ22wwJ@mail.gmail.com>
	<AANLkTinecv9E6YnfN+8XfKYoBNWaEu4UpxciPD0izAzj@mail.gmail.com>
	<AANLkTin6=rQD3R7LpPG4vR6jJxC=dvjjEmSN5AwJFvQa@mail.gmail.com>
	<AANLkTi=bpNixuziYRCBk-Ap+A8qTx+JsiManGYeKGkfp@mail.gmail.com>
	<AANLkTinBexiCrEw0U18tKdQJwinfsRiuA_q_9D-GVKxE@mail.gmail.com>
Message-ID: <AANLkTik4BMujGUOrnQN72eYwhCiwkHfrUPo=bny1ZHmH@mail.gmail.com>

On Tue, Oct 19, 2010 at 11:19 AM, DJB International <djbinter at gmail.com>wrote:

> Thank you, Rupa.  I will try to change it to hash and see what happen.
>
> Just curiosity, is there any advantage/disadvantage using hash vs. db.  I
> am currently not doing any cluster.  I picked mod_db because it was
> mentioned first in the Wiki/Limit :)
>

If you aren't clustered, always choose hash.

If you are clustered then you want to choose between db and um... whatever
mod_redis provides (distributed hash).

hash is fast, but non persistent and can't be shared between FS instances
(clustered).
db is slow, persistent and can be shared
redis is slower than hash but faster than db and can be shared.


>
> Dorn.
>
>
> --
-Rupa
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From covici at ccs.covici.com  Tue Oct 19 10:11:18 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 19 Oct 2010 13:11:18 -0400
Subject: [Freeswitch-users] Compile on Windows
In-Reply-To: <SNT135-w44528AFDB4D0979738878EB75B0@phx.gbl>
References: <mailman.11271.1287285783.19150.freeswitch-users@lists.freeswitch.org>,
	, <BAY157-w6D3D334579856A7F1E818DE5A0@phx.gbl>,
	<BAY157-w56107C5E9B290FE432741DDE5A0@phx.gbl>
	<SNT135-w44528AFDB4D0979738878EB75B0@phx.gbl>
Message-ID: <13698.1287508278@ccs.covici.com>

I just compiled from git head this morning on a Windows 7 box -- a
couple of modules failed for lack of lame, but otherwise  it seems to
have its default dialplan functioning.

Brian Campbell <bcxml at hotmail.com> wrote:

> 
> Just went through a similar exercise...finally got it to work with the 1.0.6 release
>  
> The only way I could get a clean compile was VS2008 running on a Windows Server 2003 SP2 box
>  
> Hope that helps
>  
>  
> Brian
>  
> 
> 
> From: timb0311 at hotmail.com
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 18 Oct 2010 17:29:17 -0400
> Subject: Re: [Freeswitch-users] Compile on Windows
> 
> 
> 
> 
> 
> 
> 
> Thanks guys... I have tried building...
>  
> 
> Nightly Snapshot of Git Source freeswitch-snapshot.tar.gz 
> the official release http://files.freeswitch.org/freeswitch-1.0.6.tar.gz
> and git clone with autocrlf=false  
> still getting errors and failed builds....
> 
>  
> ========== Rebuild All: 58 succeeded, 76 failed, 18 skipped ==========
>  
>  
> Now is there an specifics I need to set in VS2010 for my c++ options, etc?  I am a c# developer so I don't do much with c++, so I never messed with any of the options there.  I did download the windows 7 sdk so I would have all the windows header files.
>  
>  
> Tim
>  
>  
>  
> 
>  
> 
> 
> 
>  
> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 		 	   		  
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From anthony.minessale at gmail.com  Tue Oct 19 10:18:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 19 Oct 2010 12:18:23 -0500
Subject: [Freeswitch-users] energy level and conference recording
In-Reply-To: <13505.1287507131@ccs.covici.com>
References: <13505.1287507131@ccs.covici.com>
Message-ID: <AANLkTikv3hyfROzsEtctb2EVT7kg4XnKr0218ViHvqyv@mail.gmail.com>

it's not a bug it takes that long to decide someone is talking, it's
the same amount of audio you may miss while actually talking to them.
The answer is not to use energy detection if you treasure those
milliseconds.


On Tue, Oct 19, 2010 at 11:52 AM,  <covici at ccs.covici.com> wrote:
> Hi. ?I have found that if the energy level is not 0 and I try to record
> the conference, the recording loses several milliseconds -- maybe up to
> 20-50 after silence. ?I don't think this should happen, but let me know
> and I can file a bug or is there another work around?
>
> Thanks in advance for any suggestions.
>
> --
> Your life is like a penny. ?You're going to lose it. ?The question is:
> How do
> you spend it?
>
> ? ? ? ? John Covici
> ? ? ? ? covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From null at invalid.name  Tue Oct 19 10:44:56 2010
From: null at invalid.name (Dan Lane)
Date: Tue, 19 Oct 2010 18:44:56 +0100
Subject: [Freeswitch-users] mod_nibblebill hangs up A-Leg and continues
 dialing B-Leg
In-Reply-To: <CB3C8C74-B8AF-4503-B449-22CF2E3E63CB@archerdrive.com>
References: <CB3C8C74-B8AF-4503-B449-22CF2E3E63CB@archerdrive.com>
Message-ID: <AANLkTimhqZ6ZqOR6rru2T+no2LBSFJCeDqof2GCQmahQ@mail.gmail.com>

On Mon, Oct 18, 2010 at 3:47 AM, Ash <ash at archerdrive.com> wrote:
> Hi All,
>
> I am trying to setup mod nibble bill to bill based on the B-Leg. ?I have followed the wiki but have been unable to resolve my issue. ?I first had this issue with 1.0.6 and tried the latest git version and can reproduce the fault on both versions.
>
> When the customers credit is 0 or below the handset I am using to test gets transferred to the hangup destination and the A-Leg is hungup. ?After a couple of seconds my mobile will ring and when I answer it there is silence so a one way call leg. ?As anyone else seen this behaviour?
>

Yup, I believe this is normal for NibbleBill... we worked around it by
checking the balance before attempting to make the call.


From timb0311 at hotmail.com  Tue Oct 19 10:41:18 2010
From: timb0311 at hotmail.com (Tim B)
Date: Tue, 19 Oct 2010 13:41:18 -0400
Subject: [Freeswitch-users] Compile on Windows
In-Reply-To: <mailman.11339.1287497913.19150.freeswitch-users@lists.freeswitch.org>
References: <mailman.11339.1287497913.19150.freeswitch-users@lists.freeswitch.org>
Message-ID: <BAY157-w43DB09B9FC23707A374A5DE5B0@phx.gbl>


Ok and update... I was able to try to compile on an WinXP x86 from GIT using VS2008.  
 
But the following failed.... 
- Win7 x64, GIT, VS2010
- Win Server 2008 x86, GIT, VS2008 and VC++ Express 2010 		 	   		  
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From joedotslash at gmail.com  Tue Oct 19 10:26:56 2010
From: joedotslash at gmail.com (joedotslash)
Date: Tue, 19 Oct 2010 10:26:56 -0700
Subject: [Freeswitch-users] FreeSwitch as SBC, no registration?
In-Reply-To: <7AB39D2F-ECB7-466C-8643-21D354750AE2@ipeva.fr>
References: <C1B9B3DB-066D-4853-AD47-C83139A76211@gmail.com>
	<7AB39D2F-ECB7-466C-8643-21D354750AE2@ipeva.fr>
Message-ID: <FCDC05F5-3DA8-427F-A82A-C97382A49FE4@gmail.com>

Thanks for the info!

B will also need to send calls to A on a different port... so I guess I'll need to create an additional profile so FS will listen on an additional port?

Security isn't needed in this case.

And this will act as an SBC right? None of the IPv6 addresses from A will show up in SIP traffic going to B? (and vice versa)

Joe

On Oct 19, 2010, at 6:38 AM, David Ponzone wrote:

> So you just want to receive calls from A (which is a well-known IPv6 address) and send them to B.
> 
> I would then modify the internal profile to be the exact same than external, except that I would bind internal to 2001::2:5060 and external to 1.1.1.1:5060.
> Then you just need a basic dialplan (public.xml) to send to B calls received from A.
> Of course, some security would be also a plus.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 18/10/2010 ? 21:20, joedotslash a ?crit :
> 
>> Hello all,
>> 
>> I'm looking to implement the following scenario:
>> 
>> Step 1
>> SIP Server A sends INVITE to port 5060 over IPv6 to FreeSwitch:
>> 2001::1 --udp/tcp--> 2001::2:5060
>> 
>> Step 2
>> FreeSwitch SIP NATs the INVITE and sends it out IPv4 to SIP Server B on port 6000
>> 1.1.1.1 --udp/tcp--> 1.1.1.2:6000
>> 
>> Step 3
>> All SIP Server B responses within this SIP dialog are sent back to SIP Server A along the same path.
>> 
>> REGISTER messages WILL NOT be processed by FreeSwitch. They are handle elsewhere.
>> 
>> There is more to this but I want to accomplish this first call flow go from there.
>> 
>> Is this possible with FreeSwitch? Can someone point me toward a resource on how to get this created?
>> 
>> The following looks like a good start but it seems it's setup for users to register. I would prefer not to have this.
>> 
>> http://wiki.freeswitch.org/wiki/SBC_Setup
>> 
>> Thanks!
>> Joe
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From joedotslash at gmail.com  Tue Oct 19 10:29:54 2010
From: joedotslash at gmail.com (joedotslash)
Date: Tue, 19 Oct 2010 10:29:54 -0700
Subject: [Freeswitch-users] FreeSwitch as SBC, no registration?
In-Reply-To: <1564036B-793A-491C-95B7-A1A72A8B29DF@freeswitch.org>
References: <C1B9B3DB-066D-4853-AD47-C83139A76211@gmail.com>
	<7AB39D2F-ECB7-466C-8643-21D354750AE2@ipeva.fr>
	<1564036B-793A-491C-95B7-A1A72A8B29DF@freeswitch.org>
Message-ID: <583E479D-E672-4D18-BE51-92591E0F4E64@gmail.com>

But register messages aren't required right? It's not mandatory that I send them through FreeSwitch is it?

I was planning to setup CIDR ACLs so the A side (another SIP platform) can route traffic through FreeSwitch based on the source IP... and the same thing would happen in the opposite direction. The B side is another SIP platform as well.

Thanks!
Joe

On Oct 19, 2010, at 7:17 AM, Brian West wrote:

> FreeSWITCH will always process the register request... Currently their is NO way to pass it thru.  See jira their is a bounty on adding this functionality.
> 
> /b
> 
> On Oct 19, 2010, at 8:38 AM, David Ponzone wrote:
> 
>> So you just want to receive calls from A (which is a well-known IPv6 address) and send them to B.
>> 
>> I would then modify the internal profile to be the exact same than external, except that I would bind internal to 2001::2:5060 and external to 1.1.1.1:5060.
>> Then you just need a basic dialplan (public.xml) to send to B calls received from A.
>> Of course, some security would be also a plus.
>> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From curriegrad2004 at gmail.com  Tue Oct 19 11:05:03 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Tue, 19 Oct 2010 11:05:03 -0700
Subject: [Freeswitch-users] Compile on Windows
In-Reply-To: <BAY157-w43DB09B9FC23707A374A5DE5B0@phx.gbl>
References: <mailman.11339.1287497913.19150.freeswitch-users@lists.freeswitch.org>
	<BAY157-w43DB09B9FC23707A374A5DE5B0@phx.gbl>
Message-ID: <AANLkTimJyw+OKAjiez+0-9js7MbbZkbkzzZixu+9iD9g@mail.gmail.com>

I'd recommend you to upgrade VS2008 to at least SP1 before attempting
to compile FreeSwitch. VS2008 SP1 seems a lot more mature and stable
than VS2010.

On Tue, Oct 19, 2010 at 10:41 AM, Tim B <timb0311 at hotmail.com> wrote:
> Ok and update... I was able to try to compile on an WinXP x86 from GIT using
> VS2008.
>
> But the following failed....
> - Win7 x64, GIT, VS2010
> - Win Server 2008 x86, GIT, VS2008 and VC++ Express 2010
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From david.ponzone at ipeva.fr  Tue Oct 19 11:10:15 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 19 Oct 2010 20:10:15 +0200
Subject: [Freeswitch-users] FreeSwitch as SBC, no registration?
In-Reply-To: <583E479D-E672-4D18-BE51-92591E0F4E64@gmail.com>
References: <C1B9B3DB-066D-4853-AD47-C83139A76211@gmail.com>
	<7AB39D2F-ECB7-466C-8643-21D354750AE2@ipeva.fr>
	<1564036B-793A-491C-95B7-A1A72A8B29DF@freeswitch.org>
	<583E479D-E672-4D18-BE51-92591E0F4E64@gmail.com>
Message-ID: <9E4A01CE-53E3-40F5-B544-D1D8997715B0@ipeva.fr>

I don't think there is an issue with what you want to do.
You basically just want FS to act as a SBC-switch between switch A and  
switch B, just to hide the IPv6 stuff coming from A.
I think Brian misunderstood you.
Your switch A is not going to send REGISTERs to FS.
Switch A deals with REGISTERs itself, and just sends the call to FS.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 19/10/2010 ? 19:29, joedotslash a ?crit :

> But register messages aren't required right? It's not mandatory that  
> I send them through FreeSwitch is it?
>
> I was planning to setup CIDR ACLs so the A side (another SIP  
> platform) can route traffic through FreeSwitch based on the source  
> IP... and the same thing would happen in the opposite direction. The  
> B side is another SIP platform as well.
>
> Thanks!
> Joe
>
> On Oct 19, 2010, at 7:17 AM, Brian West wrote:
>
>> FreeSWITCH will always process the register request... Currently  
>> their is NO way to pass it thru.  See jira their is a bounty on  
>> adding this functionality.
>>
>> /b
>>
>> On Oct 19, 2010, at 8:38 AM, David Ponzone wrote:
>>
>>> So you just want to receive calls from A (which is a well-known  
>>> IPv6 address) and send them to B.
>>>
>>> I would then modify the internal profile to be the exact same than  
>>> external, except that I would bind internal to 2001::2:5060 and  
>>> external to 1.1.1.1:5060.
>>> Then you just need a basic dialplan (public.xml) to send to B  
>>> calls received from A.
>>> Of course, some security would be also a plus.
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From lists at infosecurity.ch  Tue Oct 19 12:07:16 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Tue, 19 Oct 2010 21:07:16 +0200
Subject: [Freeswitch-users] Are there any rpath based FS distribution?
Message-ID: <4CBDEC64.5070700@infosecurity.ch>

Hi all,

does anyone know if there is some rpath (www.rpath.com) based FS
distribution that can be easily installed as an appliance by using a cdrom?

Fabio


From nico at clickfono.com  Tue Oct 19 12:27:57 2010
From: nico at clickfono.com (Nicolas Brenner)
Date: Tue, 19 Oct 2010 16:27:57 -0300
Subject: [Freeswitch-users] Gateway responding to random port (rport
	problem?)
In-Reply-To: <AANLkTimFmhZKw_boLcvArWBpDcJw-Yz0dN+=bGsG4TcK@mail.gmail.com>
References: <AANLkTi=ErWZP5SEr6p_=rej+xhioBQmxa7cCeg045c3i@mail.gmail.com>
	<AANLkTimFmhZKw_boLcvArWBpDcJw-Yz0dN+=bGsG4TcK@mail.gmail.com>
Message-ID: <AANLkTinGSju0Vsd2nB4s9PaveMdDTe2kLiib4x9v4ckd@mail.gmail.com>

I guess you are right, I just installed Asterisk and tried the exact
configuration suggested by my provider without success. Thanks for the help!


On Tue, Oct 19, 2010 at 11:58 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Do you see a common theme that you have several threads open regarding
> network topology behaviour?
>
> I think you need to audit your network.
>
>
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From peter.olsson at visionutveckling.se  Tue Oct 19 12:37:37 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 19 Oct 2010 21:37:37 +0200
Subject: [Freeswitch-users] Compile on Windows
In-Reply-To: <BAY157-w43DB09B9FC23707A374A5DE5B0@phx.gbl>
References: <mailman.11339.1287497913.19150.freeswitch-users@lists.freeswitch.org>,
	<BAY157-w43DB09B9FC23707A374A5DE5B0@phx.gbl>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57E5ECDC3C@cooper>

If it fails - please file a jira, with complete info - not just "it doesn't work". I'm pretty sure it will be fixed within a day or so :)

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tim B [timb0311 at hotmail.com]
Skickat: den 19 oktober 2010 19:41
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Compile on Windows

Ok and update... I was able to try to compile on an WinXP x86 from GIT using VS2008.

But the following failed....
- Win7 x64, GIT, VS2010
- Win Server 2008 x86, GIT, VS2008 and VC++ Express 2010
!DSPAM:4cbddadc32931503110381!


From nico at clickfono.com  Tue Oct 19 12:50:54 2010
From: nico at clickfono.com (Nicolas Brenner)
Date: Tue, 19 Oct 2010 16:50:54 -0300
Subject: [Freeswitch-users] Gateway responding to random port (rport
	problem?)
In-Reply-To: <AANLkTinGSju0Vsd2nB4s9PaveMdDTe2kLiib4x9v4ckd@mail.gmail.com>
References: <AANLkTi=ErWZP5SEr6p_=rej+xhioBQmxa7cCeg045c3i@mail.gmail.com>
	<AANLkTimFmhZKw_boLcvArWBpDcJw-Yz0dN+=bGsG4TcK@mail.gmail.com>
	<AANLkTinGSju0Vsd2nB4s9PaveMdDTe2kLiib4x9v4ckd@mail.gmail.com>
Message-ID: <AANLkTi=1gA6a17J0JaREmxiJ1k13xK=SzZBxnk6DSEa3@mail.gmail.com>

I was able to solve this by forcing transport to tcp instead of udp (the
default). I just set the following on the gateway's config: <param
name="register-transport" value="tcp"/> and it just worked.


On Tue, Oct 19, 2010 at 4:27 PM, Nicolas Brenner <nico at clickfono.com> wrote:

> I guess you are right, I just installed Asterisk and tried the exact
> configuration suggested by my provider without success. Thanks for the help!
>
>
> On Tue, Oct 19, 2010 at 11:58 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Do you see a common theme that you have several threads open regarding
>> network topology behaviour?
>>
>> I think you need to audit your network.
>>
>>
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From nico at clickfono.com  Tue Oct 19 13:02:01 2010
From: nico at clickfono.com (Nicolas Brenner)
Date: Tue, 19 Oct 2010 17:02:01 -0300
Subject: [Freeswitch-users] Remove rport from Via header
In-Reply-To: <AANLkTimKgR_VMYM_ijxihVC97qKHVsD+fLu59i-0FB+5@mail.gmail.com>
References: <AANLkTiky=9vA5uk-gqKHHjuabF0VQE1tGk0wODPc7Op2@mail.gmail.com>
	<3799D7D4-A663-4C55-A4A5-A63585E413CE@ipeva.fr>
	<AANLkTimiwbf4yRKERy7uQ_gqhW3+N6Zg_Xpmd+J4giGM@mail.gmail.com>
	<AANLkTimKgR_VMYM_ijxihVC97qKHVsD+fLu59i-0FB+5@mail.gmail.com>
Message-ID: <AANLkTincjorVksF-F7oFBwxSRQ3UzbGpd_cXTs5iJ5uX@mail.gmail.com>

Thanks for all the help.

The provider is a cellphone operator in my country, they usually don't
interconnect with anybody (at least not through SIP), but are giving us
access to this only because they want the calls from our service to be
routed directly to them.

Unfortunately I have no access to the router, we rent this server on The
Planet, but we have never had any problems like this with any other
provider, so I'm guessing this doesn't have to do with nat.

Anyway, I was finally able to solve this by setting transport to tcp instead
of udp on the gateway config with: <param name="register-transport"
value="tcp"/> Even though it's working fine now, I'm not that happy, because
I have no idea why it didn't work with udp. Well, hopefully the
provider/customer will be glad this got solved.

Again thanks to everyone for all the help. I'd be happy to answer any
questions if you are curious about this or have a similar problem.


Best,


Nico



On Tue, Oct 19, 2010 at 11:45 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> If the top of their standards are to make it work with asterisk, they
> are clearly not very wise.
>
> Are you using some kind of asymmetric nat? Why is it getting a random port?
> Even with rport enabled it should be the true port the packet is seen
> coming from and a return path should be expected the same way.
>
> Check your router for asymmetric nat or SIP ALG features that you can
> disable in it's config.
>
>
>
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From msc at freeswitch.org  Tue Oct 19 13:06:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 Oct 2010 13:06:16 -0700
Subject: [Freeswitch-users] record on demand for a conference call
In-Reply-To: <AANLkTi=3mHhqtWbLiw+onv8MDAWGLF+0mhKt-M6kiWBe@mail.gmail.com>
References: <6DFF80D26E14467299C2B7789D1D82D6@D810>
	<AANLkTimuurW-3SJX1+JB=rzbgmq1mGd12hQEvzmfD4A6@mail.gmail.com>
	<AANLkTi=3mHhqtWbLiw+onv8MDAWGLF+0mhKt-M6kiWBe@mail.gmail.com>
Message-ID: <AANLkTimLQ_N=Gr+azdnc9Y-ubwN6v5qmhiF8sSX8cKOp@mail.gmail.com>

You can definitely write an ESL app to handle this. It's totally DIY, but FS
gives you all the tools and parts.
-MC

On Tue, Oct 19, 2010 at 2:31 AM, Andy Ayers <andy at fabulous4.co.uk> wrote:

> thanks for your interest Michael, it would be to allow the moderator to
> record an introduction to the call so when it's listened to by others it has
> context. But actually more useful is to be able to start recording when the
> moderator wants it to to cut off the usually messy start to a conference
> call. Is this possible, could I configure a dtmf key press to initiate the
> recording somehow?
>
>
> On 13 October 2010 20:05, Michael Collins <msc at freeswitch.org> wrote:
>
>> Just curious - what value is there to recording the call when only one
>> person is there?
>> -MC
>>
>>  On Wed, Oct 13, 2010 at 3:42 AM, Andy <andy at fabulous4.co.uk> wrote:
>>
>>>   Hi,
>>>
>>> I would like to set up my conferencing system so that it's possible to
>>> initiate the recording of the conference when the moderator presses a key on
>>> their keypad.
>>>
>>> Failing that I would like to know if there is any way of recording the
>>> entire conference call from the moment the first caller arrives rather than
>>> using the auto-record feature which only kicks in when 2 or more people are
>>> connected.
>>>
>>> Can anyone give me some idea how this can be done or point me to the
>>> right wiki pages?
>>>
>>> Many thanks for you help
>>> Andy
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From curriegrad2004 at gmail.com  Tue Oct 19 13:25:20 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Tue, 19 Oct 2010 13:25:20 -0700
Subject: [Freeswitch-users] Compile on Windows
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57E5ECDC3C@cooper>
References: <mailman.11339.1287497913.19150.freeswitch-users@lists.freeswitch.org>
	<BAY157-w43DB09B9FC23707A374A5DE5B0@phx.gbl>
	<549CFEF87AEDE841A38E9D15EAB4C04C57E5ECDC3C@cooper>
Message-ID: <AANLkTi=w6UpyRSxYQFiM8gP5FcRODU9vTcB1G0frE7XR@mail.gmail.com>

And if you can, please include the revision of where it failed.

On Tue, Oct 19, 2010 at 12:37 PM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> If it fails - please file a jira, with complete info - not just "it doesn't work". I'm pretty sure it will be fixed within a day or so :)
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tim B [timb0311 at hotmail.com]
> Skickat: den 19 oktober 2010 19:41
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: Re: [Freeswitch-users] Compile on Windows
>
> Ok and update... I was able to try to compile on an WinXP x86 from GIT using VS2008.
>
> But the following failed....
> - Win7 x64, GIT, VS2010
> - Win Server 2008 x86, GIT, VS2008 and VC++ Express 2010
> !DSPAM:4cbddadc32931503110381!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From oseslija at gmail.com  Tue Oct 19 14:22:56 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Tue, 19 Oct 2010 23:22:56 +0200
Subject: [Freeswitch-users] FreeSwitch and Cisco 7941 IP Phone ( INVITE
 port number )
In-Reply-To: <057bc2928b30a59af5a20d08c3fb5141@astylos.de>
References: <057bc2928b30a59af5a20d08c3fb5141@astylos.de>
Message-ID: <AANLkTik5BkOMcDvaQseJocsp2276_yAa31-PEkpK1kMG@mail.gmail.com>

Please see http://jira.freeswitch.org/browse/FS-2773 I reported.

The mentioned code is now commented in FS, so nat is no longer assumed.

Regards.

On Mon, Oct 18, 2010 at 5:15 PM, Christian Knoblauch <
christian.knoblauch at astylos.de> wrote:

>
> Hello,
>
>
> This is about FreeSwitch 1.0.6 (Windows Build) and Cisco 7941 IP Phone, and
> the same issue was recently reported by another subscriber of this list
> (Michael W. Lucas).
>
> Anyhow, I attach traces so that maybe someone can look into it.
>
>
> CONTEXT:
>
> The 7941 successfull REGISTER in FreeSwitch, and is able to call another
> softphone.
> The softphone is not able to call the 7941, and this is because the INVITE
> from FreeSwitch towards 7941 goes to the UDP port that the 7941 was using
> during REGISTER (high port number, instead of 5060)
>
> I compared this with traces for 3CX where all works fine. 3CX sends the
> INVITE towards 7941 to UDP port 5060, and the 7941 seems to like this :-)
>
>
> Please find attached the REGISTER / INVITE traces for booth 3CX and
> FreeSwitch, and also the FreeSwitch console-output from"sofia profile
> internal siptrace on"
>
>
> Thanks for review !
>
> Regards
> Christian
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From oseslija at gmail.com  Tue Oct 19 14:27:39 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Tue, 19 Oct 2010 23:27:39 +0200
Subject: [Freeswitch-users] SPA921 Problem Bad Session Description
In-Reply-To: <AANLkTi=mh=b6NPDBmdHztYtRY4x95OWN5VrXYQE28be9@mail.gmail.com>
References: <AANLkTikfwv=yiorKvGegPmT408xgSOL_8f=gZ=0WnPKV@mail.gmail.com>
	<AANLkTikM_FJi=vV1zVf+SBJSL-H7Orf9X8t2e7O581ZM@mail.gmail.com>
	<AANLkTi=mh=b6NPDBmdHztYtRY4x95OWN5VrXYQE28be9@mail.gmail.com>
Message-ID: <AANLkTint0xXy27oHpqFO+hYwQGtoFGUji74vknyzvZC8@mail.gmail.com>

Unfortunately, the model in question (921) has latest fw.
For some odd reason, Linksys decided not to issue 5.2.x and 6.x fws for 9x1
models, even though phones are practically same as 9x2 counterparts (PoE is
the difference).
Bad luck.

Maybe you can play with codec name params in web interface of Linksys.


On Tue, Oct 19, 2010 at 4:37 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> We already told him this when he filed a JIRA on it.
> It's clearly broken and probably fixed with a firmware update to the
> device.
>
>
> On Tue, Oct 19, 2010 at 9:23 AM, Kristian Kielhofner <kris at kriskinc.com>
> wrote:
> > Maxim,
> >
> >  That SDP looks nasty:
> >
> >   a=rtpmap:8 /8000
> >   a=rtpmap:0 /8000
> >
> >  These are static payload types and don't require an rtpmap line (the
> > "offer" is in the m= line).  However, when you use an rtpmap you must
> > use the IANA payload type names: PCMU and PCMA in this case.  Either
> > of these two SDPs would be valid:
> >
> >   v=0
> >   o=- 603544 603544 IN IP4 xxx
> >   s=-
> >   c=IN IP4 xxx
> >   t=0 0
> >   m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101
> >   a=rtpmap:2 G726-32/8000
> >   a=rtpmap:4 G723/8000
> >   a=rtpmap:18 G729/8000
> >   a=rtpmap:96 G726-40/8000
> >   a=rtpmap:97 G726-24/8000
> >   a=rtpmap:98 G726-16/8000
> >   a=rtpmap:101 telephone-event/8000
> >   a=fmtp:101 0-15
> >   a=ptime:30
> >   a=sendrecv
> >
> > -or-
> >
> >   v=0
> >   o=- 603544 603544 IN IP4 xxx
> >   s=-
> >   c=IN IP4 xxx
> >   t=0 0
> >   m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101
> >   a=rtpmap:8 PCMA/8000
> >   a=rtpmap:0 PCMU/8000
> >   a=rtpmap:2 G726-32/8000
> >   a=rtpmap:4 G723/8000
> >   a=rtpmap:18 G729/8000
> >   a=rtpmap:96 G726-40/8000
> >   a=rtpmap:97 G726-24/8000
> >   a=rtpmap:98 G726-16/8000
> >   a=rtpmap:101 telephone-event/8000
> >   a=fmtp:101 0-15
> >   a=ptime:30
> >   a=sendrecv
> >
> >  Yours isn't :).
> >
> > http://www.iana.org/assignments/rtp-parameters
> >
> > On Mon, Oct 18, 2010 at 6:32 AM, Maxim Balabaev <balabaev.m at gmail.com>
> wrote:
> >> I can`t make calls from linksys spa921 because of "400 Bad Session
> >> Description", incoming are ok. pap2t works perfectly. freeswitch is rev
> from
> >> git trunk. Here comes logs:
> >>
> ------------------------------------------------------------------------
> >> recv 868 bytes from udp/[xxx]:5060 at 10:27:30.709397:
> >>
> ------------------------------------------------------------------------
> >>    INVITE sip:1001 at xxx SIP/2.0
> >>    Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport
> >>    From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
> >>    To: <sip:1001 at xxx>
> >>    Call-ID: 14d793ed-1abf9d0e at xxx
> >>    CSeq: 101 INVITE
> >>    Max-Forwards: 70
> >>    Contact: <sip:022 at xxx:5060>
> >>    Expires: 240
> >>    User-Agent: Linksys/SPA921-5.1.8
> >>    Content-Length: 386
> >>    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> >>    Supported: replaces
> >>    Content-Type: application/sdp
> >>
> >>    v=0
> >>    o=- 603544 603544 IN IP4 xxx
> >>    s=-
> >>    c=IN IP4 xxx
> >>    t=0 0
> >>    m=audio 16406 RTP/AVP 8 0 2 4 18 96 97 98 101
> >>    a=rtpmap:8 /8000
> >>    a=rtpmap:0 /8000
> >>    a=rtpmap:2 G726-32/8000
> >>    a=rtpmap:4 G723/8000
> >>    a=rtpmap:18 G729/8000
> >>    a=rtpmap:96 G726-40/8000
> >>    a=rtpmap:97 G726-24/8000
> >>    a=rtpmap:98 G726-16/8000
> >>    a=rtpmap:101 telephone-event/8000
> >>    a=fmtp:101 0-15
> >>    a=ptime:30
> >>    a=sendrecv
> >>
> ------------------------------------------------------------------------
> >> send 562 bytes to udp/[xxx]:5060 at 10:27:30.709607:
> >>
> ------------------------------------------------------------------------
> >>    SIP/2.0 400 Bad Session Description
> >>    Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK-ede98b83;rport=5060
> >>    From: <sip:022 at xxx>;tag=8f6dbe61c1d0552ao0
> >>    To: <sip:1001 at xxx>;tag=ytUD271ypvy6r
> >>    Call-ID: 14d793ed-1abf9d0e at xxx
> >>    CSeq: 101 INVITE
> >>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f8d8a39 2010-10-18
> 03-19-16
> >> -0400
> >>    Accept: application/sdp
> >>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> >> REGISTER, REFER, NOTIFY
> >>    Supported: timer, precondition, path, replaces
> >>    Allow-Events: talk, hold, refer
> >>    Content-Length: 0
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
> >
> > --
> > Kristian Kielhofner
> > http://www.astlinux.org
> > http://blog.krisk.org
> > http://www.star2star.com
> > http://www.submityoursip.com
> > http://www.voalte.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Oct 19 16:16:18 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 Oct 2010 16:16:18 -0700
Subject: [Freeswitch-users] Looking for configuration example for a
 Linksys PAP2t-NA
In-Reply-To: <007a01cb6ef3$d8459c20$88d0d460$@accra.ca>
References: <007a01cb6ef3$d8459c20$88d0d460$@accra.ca>
Message-ID: <AANLkTikq7399atF6jw4kftOtKErSqBvLYdPs2cV9UTn7@mail.gmail.com>

Latest PAP2T firmware is not required, but you might as well update.

Factory reset the PAP2T before you do anything else, just in case there are
legacy settings in there. I have only used a PAP2T on a CudaTel and not
actually on a straight FS install. However it did not seem overly
complicated if you are familiar with the PAP2T configs. If someone has a
sample config please post to the list here and we'll put it up on the wiki.

-MC

On Mon, Oct 18, 2010 at 11:39 AM, Charles Bujold <cjbujold at accra.ca> wrote:

> Trying to connect a Linksys PAP2T-NA ATA to Freeswitch and trying to find
> a  configuration example and settings to make it work.  If anybody has a
> suggestion please forward it.  Also do I need to upgrade the firmware of the
> Pap2T to the latest version for it to work with Freeswitch (v5.16)?
>
>
>
> Thanks
>
>
>
> *cjb*
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Oct 19 16:22:15 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 Oct 2010 16:22:15 -0700
Subject: [Freeswitch-users] Early media beginning clipped for certain
	callers
In-Reply-To: <4CBDCA0B.5070900@mgtech.com>
References: <4CBDCA0B.5070900@mgtech.com>
Message-ID: <AANLkTinYpPbhWtb3whS3+LXsGt5BJ7U4eXjS0-fH39wF@mail.gmail.com>

I've seen some carriers do weird things, like wait until they receive audio
before they send audio. Maybe you can test by answering, sleeping for a few
hundred milliseconds, then play a few hundred milliseconds of silence, then
connecting the call through.

-MC

On Tue, Oct 19, 2010 at 9:40 AM, Mario <mario_fs at mgtech.com> wrote:

> I have replaced what the incoming caller hears (ringing) with a
> greeting, instructions, ringing. Works great but for some callers, about
> 1 second is clipped, it seems to do with who is calling:
>
> Caller A cell phone & B land line works every time and is never clipped.
> Caller C cell phone & D land line clips every time.
>
> I added up to a 5 second "sleep" before the media to test but it had no
> effect telling me the other end is clipping the audio when it starts.
> Any ideas on what I might try? I can try to lengthen the audio but I
> didn't want to make the working ones listen to extra silence. Thanks!
> Mario
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bwibowo at gmail.com  Tue Oct 19 16:25:56 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Wed, 20 Oct 2010 06:25:56 +0700
Subject: [Freeswitch-users] mod_nibblebill
Message-ID: <AANLkTi=aTvLC=k6wFnD7VbjAY1q22dgrEfG2QiB4-ojn@mail.gmail.com>

dear all

i install mod_nibblebil on centos 5.5 according to steps in
http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core

odbc.ini config:
[fs_connect]
Driver   = mySQL
SERVER   = localhost
PORT     = 3306
DATABASE = fs
OPTION  = 67108864
Socket   = /var/lib/mysql/mysql.sock

switch.conf.xml i add this

 <param name="odbc-dsn" value="fs_connect:root:rootpass" />

if i execute isql fs_connect root rootpass it's the result
+---------------------------------------+
| Connected!                            |
|                                       |
| sql-statement                         |
| help [tablename]                      |
| quit                                  |
|                                       |
+---------------------------------------+
SQL>


but i found this error, where's my mistake ?

i 2010-10-20 07:20:39.411254 [CRIT] switch_odbc.c:210 The sql server is not
responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR: [unixODBC][Driver
Manager]Connnection does not exist
][169]
2010-10-20 07:20:39.411254 [CRIT] switch_odbc.c:218 The connection could not
be re-established
2010-10-20 07:20:40.412241 [ERR] switch_odbc.c:318 STATE: IM002 CODE 0
ERROR: [unixODBC][Driver Manager]Data source name not found, and no default
driver specified

2010-10-20 07:20:40.412241 [CRIT] switch_odbc.c:210 The sql server is not
responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR: [unixODBC][Driver
Manager]Connnection does not exist
][169]
2010-10-20 07:20:40.412241 [CRIT] switch_odbc.c:218 The connection could not
be re-established
2010-10-20 07:20:41.412257 [ERR] switch_odbc.c:318 STATE: IM002 CODE 0
ERROR: [unixODBC][Driver Manager]Data source name not found, and no default
driver specified

2010-10-20 07:20:41.412257 [CRIT] switch_odbc.c:210 The sql server is not
responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR: [unixODBC][Driver
Manager]Connnection does not exist



regards

budi
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From djbinter at gmail.com  Tue Oct 19 16:47:47 2010
From: djbinter at gmail.com (DJB International)
Date: Tue, 19 Oct 2010 16:47:47 -0700
Subject: [Freeswitch-users] Looking for configuration example for a
 Linksys PAP2t-NA
In-Reply-To: <AANLkTikq7399atF6jw4kftOtKErSqBvLYdPs2cV9UTn7@mail.gmail.com>
References: <007a01cb6ef3$d8459c20$88d0d460$@accra.ca>
	<AANLkTikq7399atF6jw4kftOtKErSqBvLYdPs2cV9UTn7@mail.gmail.com>
Message-ID: <AANLkTimK-B79EjwVncWN0F8kew71BB5VrhNxC7k5Tk2y@mail.gmail.com>

Please see the attached file.

On Tue, Oct 19, 2010 at 4:16 PM, Michael Collins <msc at freeswitch.org> wrote:

> Latest PAP2T firmware is not required, but you might as well update.
>
> Factory reset the PAP2T before you do anything else, just in case there are
> legacy settings in there. I have only used a PAP2T on a CudaTel and not
> actually on a straight FS install. However it did not seem overly
> complicated if you are familiar with the PAP2T configs. If someone has a
> sample config please post to the list here and we'll put it up on the wiki.
>
> -MC
>
> On Mon, Oct 18, 2010 at 11:39 AM, Charles Bujold <cjbujold at accra.ca>wrote:
>
>> Trying to connect a Linksys PAP2T-NA ATA to Freeswitch and trying to find
>> a  configuration example and settings to make it work.  If anybody has a
>> suggestion please forward it.  Also do I need to upgrade the firmware of the
>> Pap2T to the latest version for it to work with Freeswitch (v5.16)?
>>
>>
>>
>> Thanks
>>
>>
>>
>> *cjb*
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue Oct 19 17:20:06 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Oct 2010 19:20:06 -0500
Subject: [Freeswitch-users] Looking for configuration example for a
	Linksys PAP2t-NA
In-Reply-To: <AANLkTikq7399atF6jw4kftOtKErSqBvLYdPs2cV9UTn7@mail.gmail.com>
References: <007a01cb6ef3$d8459c20$88d0d460$@accra.ca>
	<AANLkTikq7399atF6jw4kftOtKErSqBvLYdPs2cV9UTn7@mail.gmail.com>
Message-ID: <8C08C2C0-EF93-4DF8-ADF4-5350FEEF20CB@freeswitch.org>

It should be exactly the same as any other setup.... 

/b

On Oct 19, 2010, at 6:16 PM, Michael Collins wrote:

> Latest PAP2T firmware is not required, but you might as well update.
> 
> Factory reset the PAP2T before you do anything else, just in case there are legacy settings in there. I have only used a PAP2T on a CudaTel and not actually on a straight FS install. However it did not seem overly complicated if you are familiar with the PAP2T configs. If someone has a sample config please post to the list here and we'll put it up on the wiki.
> 
> -MC



From brian at freeswitch.org  Tue Oct 19 17:20:33 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Oct 2010 19:20:33 -0500
Subject: [Freeswitch-users] SPA921 Problem Bad Session Description
In-Reply-To: <AANLkTint0xXy27oHpqFO+hYwQGtoFGUji74vknyzvZC8@mail.gmail.com>
References: <AANLkTikfwv=yiorKvGegPmT408xgSOL_8f=gZ=0WnPKV@mail.gmail.com>
	<AANLkTikM_FJi=vV1zVf+SBJSL-H7Orf9X8t2e7O581ZM@mail.gmail.com>
	<AANLkTi=mh=b6NPDBmdHztYtRY4x95OWN5VrXYQE28be9@mail.gmail.com>
	<AANLkTint0xXy27oHpqFO+hYwQGtoFGUji74vknyzvZC8@mail.gmail.com>
Message-ID: <D741C577-DC0C-41D8-8289-1A1E66C8BFAC@freeswitch.org>

I would factory reset it... could just be a bum config.

/b

On Oct 19, 2010, at 4:27 PM, Ognjen Seslija wrote:

> Unfortunately, the model in question (921) has latest fw.
> For some odd reason, Linksys decided not to issue 5.2.x and 6.x fws for 9x1 models, even though phones are practically same as 9x2 counterparts (PoE is the difference).
> Bad luck.
> 
> Maybe you can play with codec name params in web interface of Linksys.
> 



From brian at freeswitch.org  Tue Oct 19 17:20:54 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Oct 2010 19:20:54 -0500
Subject: [Freeswitch-users] Looking for configuration example for a
	Linksys PAP2t-NA
In-Reply-To: <AANLkTimK-B79EjwVncWN0F8kew71BB5VrhNxC7k5Tk2y@mail.gmail.com>
References: <007a01cb6ef3$d8459c20$88d0d460$@accra.ca>
	<AANLkTikq7399atF6jw4kftOtKErSqBvLYdPs2cV9UTn7@mail.gmail.com>
	<AANLkTimK-B79EjwVncWN0F8kew71BB5VrhNxC7k5Tk2y@mail.gmail.com>
Message-ID: <C8689EE4-5C1D-422A-AB36-18DABE36FBC6@freeswitch.org>

Love the super secret user/pass you use.

/b

On Oct 19, 2010, at 6:47 PM, DJB International wrote:

> Please see the attached file.



From msc at freeswitch.org  Tue Oct 19 17:23:39 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 Oct 2010 17:23:39 -0700
Subject: [Freeswitch-users] transfer problem
In-Reply-To: <13464.1287506974@ccs.covici.com>
References: <13464.1287506974@ccs.covici.com>
Message-ID: <AANLkTikh67cGuMn3v4D3QZ+AuQ0vWrLE_fRhL5+g1fg7@mail.gmail.com>

Please supply the configs and a debug trace. My guess is that there's
probably just a simple config element that is wrong or missing.

-MC

On Tue, Oct 19, 2010 at 9:49 AM, <covici at ccs.covici.com> wrote:

> Hi.  I have an ivr which connects to conferences which I have in their
> own context rather than the default.  The problem is that if I call
> someone else and have a 3-way between the other party and the
> conference, its fine till I hang up and then fs tries to transfer to the
> correct conference name, but uses default for the context instead.  Is
> this expected behavior or should I file a bug?
>
> Thanks.
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue Oct 19 17:28:25 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 Oct 2010 19:28:25 -0500
Subject: [Freeswitch-users] transfer problem
In-Reply-To: <AANLkTikh67cGuMn3v4D3QZ+AuQ0vWrLE_fRhL5+g1fg7@mail.gmail.com>
References: <13464.1287506974@ccs.covici.com>
	<AANLkTikh67cGuMn3v4D3QZ+AuQ0vWrLE_fRhL5+g1fg7@mail.gmail.com>
Message-ID: <BF1F6AD8-04A6-47F4-8110-17849E2D9951@freeswitch.org>

its 100% expected to try to bridge the two channels.

/b

On Oct 19, 2010, at 7:23 PM, Michael Collins wrote:

> Please supply the configs and a debug trace. My guess is that there's probably just a simple config element that is wrong or missing.
> 
> -MC
> 
> On Tue, Oct 19, 2010 at 9:49 AM, <covici at ccs.covici.com> wrote:
> Hi.  I have an ivr which connects to conferences which I have in their
> own context rather than the default.  The problem is that if I call
> someone else and have a 3-way between the other party and the
> conference, its fine till I hang up and then fs tries to transfer to the
> correct conference name, but uses default for the context instead.  Is
> this expected behavior or should I file a bug?
> 
> Thanks.

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From adminjew at gmail.com  Tue Oct 19 17:30:39 2010
From: adminjew at gmail.com (Yitzchok)
Date: Tue, 19 Oct 2010 20:30:39 -0400
Subject: [Freeswitch-users] Looking for configuration example for a
 Linksys PAP2t-NA
In-Reply-To: <007a01cb6ef3$d8459c20$88d0d460$@accra.ca>
References: <Actu89YBNHWEyc0cQ428UlPxV3qggw==>
	<007a01cb6ef3$d8459c20$88d0d460$@accra.ca>
Message-ID: <AANLkTinXvNwddf75oMm29JqKiC0_3BeJf_7Nnemqb=Rr@mail.gmail.com>

Did you setup *stun *on the sip tab? (make sure you are in /admin/advanced)



Yitzchok


On Mon, Oct 18, 2010 at 2:39 PM, Charles Bujold <cjbujold at accra.ca> wrote:

> Trying to connect a Linksys PAP2T-NA ATA to Freeswitch and trying to find
> a  configuration example and settings to make it work.  If anybody has a
> suggestion please forward it.  Also do I need to upgrade the firmware of the
> Pap2T to the latest version for it to work with Freeswitch (v5.16)?
>
>
>
> Thanks
>
>
>
> *cjb*
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Nabble at slickdeals.endjunk.com  Tue Oct 19 19:15:20 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 19 Oct 2010 19:15:20 -0700 (PDT)
Subject: [Freeswitch-users] Looking for configuration example for a
	Linksys PAP2t-NA
In-Reply-To: <AANLkTikq7399atF6jw4kftOtKErSqBvLYdPs2cV9UTn7@mail.gmail.com>
References: <007a01cb6ef3$d8459c20$88d0d460$@accra.ca>
	<AANLkTikq7399atF6jw4kftOtKErSqBvLYdPs2cV9UTn7@mail.gmail.com>
Message-ID: <1287540920634-5653345.post@n2.nabble.com>



mercutioviz wrote:
> If someone has a sample config please post to the list here and we'll put
> it up on the wiki.
I have attached several snapshots from different ATA devices, i.e. Linksys
PAP2v1/v2 and Uniden 
http://www.uniden.com/products/productdetail.cfm?product=UIP1869V UIP1869V .

http://freeswitch-users.2379917.n2.nabble.com/file/n5653345/PAP2v1_SIP.png 

http://freeswitch-users.2379917.n2.nabble.com/file/n5653345/PAP2v1_Line_X.png 

http://freeswitch-users.2379917.n2.nabble.com/file/n5653345/PAP2v2_SIP.png 

http://freeswitch-users.2379917.n2.nabble.com/file/n5653345/PAP2v2_Line_X.png 

http://freeswitch-users.2379917.n2.nabble.com/file/n5653345/UIP1869V_SIP_Configuration.png 

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Looking-for-configuration-example-for-a-Linksys-PAP2t-NA-tp5650817p5653345.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mnhassan at usa.net  Wed Oct 20 00:25:04 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Wed, 20 Oct 2010 13:25:04 +0600
Subject: [Freeswitch-users] mod_nibblebill
In-Reply-To: <AANLkTi=aTvLC=k6wFnD7VbjAY1q22dgrEfG2QiB4-ojn@mail.gmail.com>
References: <AANLkTi=aTvLC=k6wFnD7VbjAY1q22dgrEfG2QiB4-ojn@mail.gmail.com>
Message-ID: <AANLkTi=unUZ8a-pULJeTnNS91wqrJuR90mjRi_qLQyag@mail.gmail.com>

As per the debug logs, the name of the dsn is missing a "n" inside FreeSWITCH.

Regards
HASSAN


On 2010-10-20, budi wibowo <bwibowo at gmail.com> wrote:
> dear all
>
> i install mod_nibblebil on centos 5.5 according to steps in
> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
>
> odbc.ini config:
> [fs_connect]
> Driver   = mySQL
> SERVER   = localhost
> PORT     = 3306
> DATABASE = fs
> OPTION  = 67108864
> Socket   = /var/lib/mysql/mysql.sock
>
> switch.conf.xml i add this
>
>  <param name="odbc-dsn" value="fs_connect:root:rootpass" />
>
> if i execute isql fs_connect root rootpass it's the result
> +---------------------------------------+
> | Connected!                            |
> |                                       |
> | sql-statement                         |
> | help [tablename]                      |
> | quit                                  |
> |                                       |
> +---------------------------------------+
> SQL>
>
>
> but i found this error, where's my mistake ?
>
> i 2010-10-20 07:20:39.411254 [CRIT] switch_odbc.c:210 The sql server is not
> responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR: [unixODBC][Driver
> Manager]Connnection does not exist
> ][169]
> 2010-10-20 07:20:39.411254 [CRIT] switch_odbc.c:218 The connection could not
> be re-established
> 2010-10-20 07:20:40.412241 [ERR] switch_odbc.c:318 STATE: IM002 CODE 0
> ERROR: [unixODBC][Driver Manager]Data source name not found, and no default
> driver specified
>
> 2010-10-20 07:20:40.412241 [CRIT] switch_odbc.c:210 The sql server is not
> responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR: [unixODBC][Driver
> Manager]Connnection does not exist
> ][169]
> 2010-10-20 07:20:40.412241 [CRIT] switch_odbc.c:218 The connection could not
> be re-established
> 2010-10-20 07:20:41.412257 [ERR] switch_odbc.c:318 STATE: IM002 CODE 0
> ERROR: [unixODBC][Driver Manager]Data source name not found, and no default
> driver specified
>
> 2010-10-20 07:20:41.412257 [CRIT] switch_odbc.c:210 The sql server is not
> responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR: [unixODBC][Driver
> Manager]Connnection does not exist
>
>
>
> regards
>
> budi
>

-- 
Sent from my mobile device


From covici at ccs.covici.com  Wed Oct 20 00:44:21 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 20 Oct 2010 03:44:21 -0400
Subject: [Freeswitch-users] energy level and conference recording
In-Reply-To: <AANLkTikv3hyfROzsEtctb2EVT7kg4XnKr0218ViHvqyv@mail.gmail.com>
References: <13505.1287507131@ccs.covici.com>
	<AANLkTikv3hyfROzsEtctb2EVT7kg4XnKr0218ViHvqyv@mail.gmail.com>
Message-ID: <23716.1287560661@ccs.covici.com>

OK, thanks.  The strange thing is that I can hear it much more on the
recording than I can actually hear in the conversation, but this is what
I have done, so its OK.

Anthony Minessale <anthony.minessale at gmail.com> wrote:

> it's not a bug it takes that long to decide someone is talking, it's
> the same amount of audio you may miss while actually talking to them.
> The answer is not to use energy detection if you treasure those
> milliseconds.
> 
> 
> On Tue, Oct 19, 2010 at 11:52 AM,  <covici at ccs.covici.com> wrote:
> > Hi. ?I have found that if the energy level is not 0 and I try to record
> > the conference, the recording loses several milliseconds -- maybe up to
> > 20-50 after silence. ?I don't think this should happen, but let me know
> > and I can file a bug or is there another work around?
> >
> > Thanks in advance for any suggestions.
> >
> > --
> > Your life is like a penny. ?You're going to lose it. ?The question is:
> > How do
> > you spend it?
> >
> > ? ? ? ? John Covici
> > ? ? ? ? covici at ccs.covici.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From bwibowo at gmail.com  Wed Oct 20 00:53:22 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Wed, 20 Oct 2010 14:53:22 +0700
Subject: [Freeswitch-users] mod_nibblebill
In-Reply-To: <AANLkTi=unUZ8a-pULJeTnNS91wqrJuR90mjRi_qLQyag@mail.gmail.com>
References: <AANLkTi=aTvLC=k6wFnD7VbjAY1q22dgrEfG2QiB4-ojn@mail.gmail.com>
	<AANLkTi=unUZ8a-pULJeTnNS91wqrJuR90mjRi_qLQyag@mail.gmail.com>
Message-ID: <AANLkTiksB-RfjGmTundCVgJyqQcG9w5W3iXuYXC9Fard@mail.gmail.com>

hi hassan
dns name is fs_connect, where i hould put that parameter?


regards

budi

On Wed, Oct 20, 2010 at 2:25 PM, Nyamul Hassan <mnhassan at usa.net> wrote:

> As per the debug logs, the name of the dsn is missing a "n" inside
> FreeSWITCH.
>
> Regards
> HASSAN
>
>
> On 2010-10-20, budi wibowo <bwibowo at gmail.com> wrote:
> > dear all
> >
> > i install mod_nibblebil on centos 5.5 according to steps in
> > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
> >
> > odbc.ini config:
> > [fs_connect]
> > Driver   = mySQL
> > SERVER   = localhost
> > PORT     = 3306
> > DATABASE = fs
> > OPTION  = 67108864
> > Socket   = /var/lib/mysql/mysql.sock
> >
> > switch.conf.xml i add this
> >
> >  <param name="odbc-dsn" value="fs_connect:root:rootpass" />
> >
> > if i execute isql fs_connect root rootpass it's the result
> > +---------------------------------------+
> > | Connected!                            |
> > |                                       |
> > | sql-statement                         |
> > | help [tablename]                      |
> > | quit                                  |
> > |                                       |
> > +---------------------------------------+
> > SQL>
> >
> >
> > but i found this error, where's my mistake ?
> >
> > i 2010-10-20 07:20:39.411254 [CRIT] switch_odbc.c:210 The sql server is
> not
> > responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR:
> [unixODBC][Driver
> > Manager]Connnection does not exist
> > ][169]
> > 2010-10-20 07:20:39.411254 [CRIT] switch_odbc.c:218 The connection could
> not
> > be re-established
> > 2010-10-20 07:20:40.412241 [ERR] switch_odbc.c:318 STATE: IM002 CODE 0
> > ERROR: [unixODBC][Driver Manager]Data source name not found, and no
> default
> > driver specified
> >
> > 2010-10-20 07:20:40.412241 [CRIT] switch_odbc.c:210 The sql server is not
> > responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR:
> [unixODBC][Driver
> > Manager]Connnection does not exist
> > ][169]
> > 2010-10-20 07:20:40.412241 [CRIT] switch_odbc.c:218 The connection could
> not
> > be re-established
> > 2010-10-20 07:20:41.412257 [ERR] switch_odbc.c:318 STATE: IM002 CODE 0
> > ERROR: [unixODBC][Driver Manager]Data source name not found, and no
> default
> > driver specified
> >
> > 2010-10-20 07:20:41.412257 [CRIT] switch_odbc.c:210 The sql server is not
> > responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR:
> [unixODBC][Driver
> > Manager]Connnection does not exist
> >
> >
> >
> > regards
> >
> > budi
> >
>
> --
> Sent from my mobile device
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mnhassan at usa.net  Wed Oct 20 01:15:23 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Wed, 20 Oct 2010 14:15:23 +0600
Subject: [Freeswitch-users] mod_nibblebill
In-Reply-To: <AANLkTiksB-RfjGmTundCVgJyqQcG9w5W3iXuYXC9Fard@mail.gmail.com>
References: <AANLkTi=aTvLC=k6wFnD7VbjAY1q22dgrEfG2QiB4-ojn@mail.gmail.com>
	<AANLkTi=unUZ8a-pULJeTnNS91wqrJuR90mjRi_qLQyag@mail.gmail.com>
	<AANLkTiksB-RfjGmTundCVgJyqQcG9w5W3iXuYXC9Fard@mail.gmail.com>
Message-ID: <AANLkTinT24jE84JJ+-VX3wJYQaZpEOPM5H9kJwkFNxqa@mail.gmail.com>

Check the debug log you posted earlier. The first line says all about it.

For nibble bill, you put it in the conf.xml file for nibble bill.

Regards
HASSAN


On 2010-10-20, budi wibowo <bwibowo at gmail.com> wrote:
> hi hassan
> dns name is fs_connect, where i hould put that parameter?
>
>
> regards
>
> budi
>
> On Wed, Oct 20, 2010 at 2:25 PM, Nyamul Hassan <mnhassan at usa.net> wrote:
>
>> As per the debug logs, the name of the dsn is missing a "n" inside
>> FreeSWITCH.
>>
>> Regards
>> HASSAN
>>
>>
>> On 2010-10-20, budi wibowo <bwibowo at gmail.com> wrote:
>> > dear all
>> >
>> > i install mod_nibblebil on centos 5.5 according to steps in
>> > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
>> >
>> > odbc.ini config:
>> > [fs_connect]
>> > Driver   = mySQL
>> > SERVER   = localhost
>> > PORT     = 3306
>> > DATABASE = fs
>> > OPTION  = 67108864
>> > Socket   = /var/lib/mysql/mysql.sock
>> >
>> > switch.conf.xml i add this
>> >
>> >  <param name="odbc-dsn" value="fs_connect:root:rootpass" />
>> >
>> > if i execute isql fs_connect root rootpass it's the result
>> > +---------------------------------------+
>> > | Connected!                            |
>> > |                                       |
>> > | sql-statement                         |
>> > | help [tablename]                      |
>> > | quit                                  |
>> > |                                       |
>> > +---------------------------------------+
>> > SQL>
>> >
>> >
>> > but i found this error, where's my mistake ?
>> >
>> > i 2010-10-20 07:20:39.411254 [CRIT] switch_odbc.c:210 The sql server is
>> not
>> > responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR:
>> [unixODBC][Driver
>> > Manager]Connnection does not exist
>> > ][169]
>> > 2010-10-20 07:20:39.411254 [CRIT] switch_odbc.c:218 The connection could
>> not
>> > be re-established
>> > 2010-10-20 07:20:40.412241 [ERR] switch_odbc.c:318 STATE: IM002 CODE 0
>> > ERROR: [unixODBC][Driver Manager]Data source name not found, and no
>> default
>> > driver specified
>> >
>> > 2010-10-20 07:20:40.412241 [CRIT] switch_odbc.c:210 The sql server is
>> > not
>> > responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR:
>> [unixODBC][Driver
>> > Manager]Connnection does not exist
>> > ][169]
>> > 2010-10-20 07:20:40.412241 [CRIT] switch_odbc.c:218 The connection could
>> not
>> > be re-established
>> > 2010-10-20 07:20:41.412257 [ERR] switch_odbc.c:318 STATE: IM002 CODE 0
>> > ERROR: [unixODBC][Driver Manager]Data source name not found, and no
>> default
>> > driver specified
>> >
>> > 2010-10-20 07:20:41.412257 [CRIT] switch_odbc.c:210 The sql server is
>> > not
>> > responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR:
>> [unixODBC][Driver
>> > Manager]Connnection does not exist
>> >
>> >
>> >
>> > regards
>> >
>> > budi
>> >
>>
>> --
>> Sent from my mobile device
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>

-- 
Sent from my mobile device


From thomas at chaschperli.ch  Wed Oct 20 03:02:55 2010
From: thomas at chaschperli.ch (Thomas Mueller)
Date: Wed, 20 Oct 2010 12:02:55 +0200
Subject: [Freeswitch-users] G.729 & Windows
Message-ID: <4CBEBE4F.8040107@chaschperli.ch>

  hi

I'm reading "Mac OSX and Windows systems will be supported soon" in blog 
post "G.729 Codec Licenses Now Available For FreeSWITCH!" 
(http://freeswitch.org/node/249) from 04/01/2010 - 23:34.

Searching/googling about G.729 and freeswitch did not reveal any news 
about G.729 on Windows.

Is there any estimate about when it will be available for Windows?

- Thomas


-- 
m?ller it gmbh, kanzleistrasse 126, 8004 z?rich

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From norm at voicenetwork.ca  Wed Oct 20 05:17:49 2010
From: norm at voicenetwork.ca (Norman Tomlins)
Date: Wed, 20 Oct 2010 08:17:49 -0400
Subject: [Freeswitch-users] Trouble to start fail2ban
In-Reply-To: <AANLkTi=5nZqN5yE=9Yk80aJuw5h3-UOf6dJTx_x46jrA@mail.gmail.com>
References: <AANLkTi=5nZqN5yE=9Yk80aJuw5h3-UOf6dJTx_x46jrA@mail.gmail.com>
Message-ID: <AANLkTi=bt58wFwnZTutdNwNTC6KmLJ=Wptt8VRVnmrcZ@mail.gmail.com>

Javier,

I had to make some changes to the example on the FreeSwitch wiki to get
fail2ban working correctly.   I also have a step-by-step cut&paste guide on
http://wiki.voicenetwork.ca/wiki/Main_Page#Fail2Ban for CentOS.

[freeswitch-tcp]
enabled  = true
port     = 5060,5061,5080,5081
protocol = tcp
filter   = freeswitch
logpath  = /usr/local/freeswitch/log/freeswitch.log
action   = iptables-allports[name=freeswitch-tcp, protocol=all]
           sendmail-whois[name=FreeSwitch, dest=root,
sender=fail2ban at example.org]

[freeswitch-udp]
enabled  = true
port     = 5060,5061,5080,5081
protocol = udp
filter   = freeswitch
logpath  = /usr/local/freeswitch/log/freeswitch.log
action   = iptables-allports[name=freeswitch-udp, protocol=all]
           sendmail-whois[name=FreeSwitch, dest=root,
sender=fail2ban at example.org]


Norman Tomlins
Voice Network Inc.
http://www.VoiceNetwork.ca


2010/10/15 Javier Aristiz?bal <javieraristizabal at gmail.com>

> Hi folks,
>
> I installed fail2ban CentOS 5.5. And i followed the wiki instructions to
> configure fail2ban with FreeSWITCH. After i edit jail.conf with this config:
>
> [freeswitch-tcp]
>
> enabled  = true
> port     = 5060,5061,5080,5081
> protocol = tcp
> filter   = freeswitch
> logpath  = /usr/local/freeswitch/log/freeswitch.log
>
> [freeswitch-udp]
>
> enabled  = true
> port     = 5060,5061,5080,5081
> protocol = udp
> filter   = freeswitch
> logpath  = /usr/local/freeswitch/log/freeswitch.log
> ######
>
> The fail2ban does not start, and the logs can not get anything.. Did anyone
> have the same thing?
>
> The freeswtich.conf:
>
> # Fail2Ban configuration file
> #
> # Author: Rupa SChomaker
> #
>
> [Definition]
>
> # Option:  failregex
> # Notes.:  regex to match the password failures messages in the logfile.
> The
> #          host must be matched by a group named "host". The tag "<HOST>"
> can
> #          be used for standard IP/hostname matching and is only an alias
> for
> #          (?:::f{4,6}:)?(?P<host>[\w\-.^_]+)
> # Values:  TEXT
> #
> failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(REGISTER\) on
> sofia profile \'\w+\' for \[.*\] from ip <HOST>
>
> # Option:  ignoreregex
> # Notes.:  regex to ignore. If this regex matches, the line is ignored.
> # Values:  TEXT
> #
> ignoreregex =
>
>
> Thanks
>
> --
> Javier Aristiz?bal
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Nabble at slickdeals.endjunk.com  Wed Oct 20 06:44:11 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 20 Oct 2010 06:44:11 -0700 (PDT)
Subject: [Freeswitch-users] G.729 & Windows
In-Reply-To: <4CBEBE4F.8040107@chaschperli.ch>
References: <4CBEBE4F.8040107@chaschperli.ch>
Message-ID: <1287582251418-5654913.post@n2.nabble.com>



Thomas Mueller wrote:
> Searching/googling about G.729 and freeswitch did not reveal any news 
> about G.729 on Windows.
> 
> Is there any estimate about when it will be available for Windows?
Just curious. If a G729 CoDec is a MUST and unless you prefer to use a SIP
compliant ATA device that already comes with a built-in G729 CoDec, then you
can search through Google for some freely available (non-GPLed and perhaps a
3-rd party) SIP softphone for Windows with a built-in G729 CoDec. In this
case, unless you have a plan to configure your FS server with an option to
do a G729 transcoding, there is no need to waste money for a G729 CoDec
license because the endpoints can easily negotiate to use their respective
built-in G729 CoDec.

FYI, I have my FS server hosted on a Seagate 
http://www.seagate.com/www/en-us/products/network_storage/freeagent_dockstar
DockStar  sans a G729 CoDec and set G729 as its defaults CoDec, and all
calls are working perfectly with a G729 CoDec on my Linksys PAP2v1/v2 and/or
my Uniden  http://www.uniden.com/products/productdetail.cfm?product=UIP1869V
UIP1869V  device. Honestly, the reason I use G729 CoDec is to save some
Internet bandwidth, especially communicating with others in some 3-rd world
countries where their broadband Internet connections are still metered per
their usage/month.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/G-729-Windows-tp5654175p5654913.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kond at nstel.ru  Wed Oct 20 07:01:22 2010
From: kond at nstel.ru (Nikolay Kondratyev)
Date: Wed, 20 Oct 2010 18:01:22 +0400
Subject: [Freeswitch-users] recording into perl script
Message-ID: <20101020140122.EA9EF11519@mail.nstel.ru>

Hi all,
Currently i can record a call via 
<action application="record_session" data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
But in this way i can only record to a file.
I have some experience with perl and mysql.
I'd like to record a call into database (mysql or postgre). 
I hope it's possible via mod_perl... is it? 
If yes, how can i pass voice into perl script?
 
I guess that i can just do the following
<action application="record_session" data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
<action application="perl" data="recorder.pl $${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav" />
So that my perl script recorder.pl will get recorded file name as it's first argument.
But when my script starts, will the recording allready be finished?
 
In this way (if it will work at all) i will first record a call in a file, and then put this file into database.
Is there a way to write directly into database? 
Something like <action application="record_session" data=" |recorder.pl" /> ?


Thanks in advance,
Nikolay.
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From steveayre at gmail.com  Wed Oct 20 07:08:31 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 20 Oct 2010 15:08:31 +0100
Subject: [Freeswitch-users] G.729 & Windows
In-Reply-To: <1287582251418-5654913.post@n2.nabble.com>
References: <4CBEBE4F.8040107@chaschperli.ch>
	<1287582251418-5654913.post@n2.nabble.com>
Message-ID: <AANLkTim2emW1C5oeS6hCsrgUSMkgeEudieKWY8AEWfMh@mail.gmail.com>

> Honestly, the reason I use G729 CoDec is to save some
> Internet bandwidth, especially communicating with others in some 3-rd world
> countries where their broadband Internet connections are still metered per
> their usage/month.

Although it's far from the only low bandwidth codec - Speex, GSM etc.

-Steve


On 20 October 2010 14:44, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
>
> Thomas Mueller wrote:
>> Searching/googling about G.729 and freeswitch did not reveal any news
>> about G.729 on Windows.
>>
>> Is there any estimate about when it will be available for Windows?
> Just curious. If a G729 CoDec is a MUST and unless you prefer to use a SIP
> compliant ATA device that already comes with a built-in G729 CoDec, then you
> can search through Google for some freely available (non-GPLed and perhaps a
> 3-rd party) SIP softphone for Windows with a built-in G729 CoDec. In this
> case, unless you have a plan to configure your FS server with an option to
> do a G729 transcoding, there is no need to waste money for a G729 CoDec
> license because the endpoints can easily negotiate to use their respective
> built-in G729 CoDec.
>
> FYI, I have my FS server hosted on a Seagate
> http://www.seagate.com/www/en-us/products/network_storage/freeagent_dockstar
> DockStar ?sans a G729 CoDec and set G729 as its defaults CoDec, and all
> calls are working perfectly with a G729 CoDec on my Linksys PAP2v1/v2 and/or
> my Uniden ?http://www.uniden.com/products/productdetail.cfm?product=UIP1869V
> UIP1869V ?device. Honestly, the reason I use G729 CoDec is to save some
> Internet bandwidth, especially communicating with others in some 3-rd world
> countries where their broadband Internet connections are still metered per
> their usage/month.
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/G-729-Windows-tp5654175p5654913.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From steveu at coppice.org  Wed Oct 20 07:39:34 2010
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 20 Oct 2010 22:39:34 +0800
Subject: [Freeswitch-users] G.729 & Windows
In-Reply-To: <1287582251418-5654913.post@n2.nabble.com>
References: <4CBEBE4F.8040107@chaschperli.ch>
	<1287582251418-5654913.post@n2.nabble.com>
Message-ID: <4CBEFF26.7040506@coppice.org>

  On 10/20/2010 09:44 PM, mazilo wrote:
>
> Thomas Mueller wrote:
>> Searching/googling about G.729 and freeswitch did not reveal any news
>> about G.729 on Windows.
>>
>> Is there any estimate about when it will be available for Windows?
> Just curious. If a G729 CoDec is a MUST and unless you prefer to use a SIP
> compliant ATA device that already comes with a built-in G729 CoDec, then you
> can search through Google for some freely available (non-GPLed and perhaps a
> 3-rd party) SIP softphone for Windows with a built-in G729 CoDec. In this
> case, unless you have a plan to configure your FS server with an option to
> do a G729 transcoding, there is no need to waste money for a G729 CoDec
> license because the endpoints can easily negotiate to use their respective
> built-in G729 CoDec.
>
> FYI, I have my FS server hosted on a Seagate
> http://www.seagate.com/www/en-us/products/network_storage/freeagent_dockstar
> DockStar  sans a G729 CoDec and set G729 as its defaults CoDec, and all
> calls are working perfectly with a G729 CoDec on my Linksys PAP2v1/v2 and/or
> my Uniden  http://www.uniden.com/products/productdetail.cfm?product=UIP1869V
> UIP1869V  device. Honestly, the reason I use G729 CoDec is to save some
> Internet bandwidth, especially communicating with others in some 3-rd world
> countries where their broadband Internet connections are still metered per
> their usage/month.
Those free sip phones with G.729 are not licenced. If you aren't going 
to worry about legal issues, there are many options.

Steve



From Nabble at slickdeals.endjunk.com  Wed Oct 20 07:41:43 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 20 Oct 2010 07:41:43 -0700 (PDT)
Subject: [Freeswitch-users] G.729 & Windows
In-Reply-To: <AANLkTim2emW1C5oeS6hCsrgUSMkgeEudieKWY8AEWfMh@mail.gmail.com>
References: <4CBEBE4F.8040107@chaschperli.ch>
	<1287582251418-5654913.post@n2.nabble.com>
	<AANLkTim2emW1C5oeS6hCsrgUSMkgeEudieKWY8AEWfMh@mail.gmail.com>
Message-ID: <1287585703594-5655162.post@n2.nabble.com>



Steven Ayre wrote:
> 
>> Honestly, the reason I use G729 CoDec is to save some
>> Internet bandwidth, especially communicating with others in some 3-rd
>> world
>> countries where their broadband Internet connections are still metered
>> per
>> their usage/month.
> 
> Although it's far from the only low bandwidth codec - Speex, GSM etc.
This may not be a bad idea for those who can use them. Unfortunately, none
of my ATA devices has a support on these two CoDecs.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/G-729-Windows-tp5654175p5655162.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From covici at ccs.covici.com  Wed Oct 20 08:32:51 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 20 Oct 2010 11:32:51 -0400
Subject: [Freeswitch-users] transfer problem
In-Reply-To: <BF1F6AD8-04A6-47F4-8110-17849E2D9951@freeswitch.org>
References: <13464.1287506974@ccs.covici.com>
	<AANLkTikh67cGuMn3v4D3QZ+AuQ0vWrLE_fRhL5+g1fg7@mail.gmail.com>
	<BF1F6AD8-04A6-47F4-8110-17849E2D9951@freeswitch.org>
Message-ID: <8635.1287588771@ccs.covici.com>

Sure, but it did not remember the correct context, even though it was in
a 3-way call, this is what is weird.

Brian West <brian at freeswitch.org> wrote:

> its 100% expected to try to bridge the two channels.
> 
> /b
> 
> On Oct 19, 2010, at 7:23 PM, Michael Collins wrote:
> 
> > Please supply the configs and a debug trace. My guess is that there's probably just a simple config element that is wrong or missing.
> > 
> > -MC
> > 
> > On Tue, Oct 19, 2010 at 9:49 AM, <covici at ccs.covici.com> wrote:
> > Hi.  I have an ivr which connects to conferences which I have in their
> > own context rather than the default.  The problem is that if I call
> > someone else and have a 3-way between the other party and the
> > conference, its fine till I hang up and then fs tries to transfer to the
> > correct conference name, but uses default for the context instead.  Is
> > this expected behavior or should I file a bug?
> > 
> > Thanks.
> 
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From ash at archerdrive.com  Wed Oct 20 03:57:27 2010
From: ash at archerdrive.com (Ash)
Date: Wed, 20 Oct 2010 21:57:27 +1100
Subject: [Freeswitch-users] mod_nibblebill hangs up A-Leg and continues
	dialing B-Leg
In-Reply-To: <AANLkTimhqZ6ZqOR6rru2T+no2LBSFJCeDqof2GCQmahQ@mail.gmail.com>
References: <CB3C8C74-B8AF-4503-B449-22CF2E3E63CB@archerdrive.com>
	<AANLkTimhqZ6ZqOR6rru2T+no2LBSFJCeDqof2GCQmahQ@mail.gmail.com>
Message-ID: <CEBF899A-C63C-4BEB-AC91-11A8C74B3AFA@archerdrive.com>

Cheers,

Thanks for your reply.  I did the exact same thing after I wrote this email via xml curl.  Reading the wiki I took the assumption that if the credit limit was below nobal_amt then it would not allow the call.

Ash.



On 20/10/2010, at 4:44 AM, Dan Lane wrote:

> On Mon, Oct 18, 2010 at 3:47 AM, Ash <ash at archerdrive.com> wrote:
>> Hi All,
>> 
>> I am trying to setup mod nibble bill to bill based on the B-Leg.  I have followed the wiki but have been unable to resolve my issue.  I first had this issue with 1.0.6 and tried the latest git version and can reproduce the fault on both versions.
>> 
>> When the customers credit is 0 or below the handset I am using to test gets transferred to the hangup destination and the A-Leg is hungup.  After a couple of seconds my mobile will ring and when I answer it there is silence so a one way call leg.  As anyone else seen this behaviour?
>> 
> 
> Yup, I believe this is normal for NibbleBill... we worked around it by
> checking the balance before attempting to make the call.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From balabaev.m at gmail.com  Wed Oct 20 06:29:06 2010
From: balabaev.m at gmail.com (Maxim Balabaev)
Date: Wed, 20 Oct 2010 17:29:06 +0400
Subject: [Freeswitch-users] Sofia gateway incoming problem
In-Reply-To: <AANLkTinPDTshrkSFK8bffkTmHNmCzpy8aT6X-H6zLfKj@mail.gmail.com>
References: <AANLkTim9Zh6rsopD+Tc0krs7+TT2C08cM+mxg-8zA_Yd@mail.gmail.com>
	<BF035383-6CC8-4FEB-AA1A-22F5FD82B767@freeswitch.org>
	<AANLkTi=3n64VgYJbkP2KFDtF0Z_PULES-Ei8Moi+28c6@mail.gmail.com>
	<AANLkTinPDTshrkSFK8bffkTmHNmCzpy8aT6X-H6zLfKj@mail.gmail.com>
Message-ID: <AANLkTi=EYwkoMQ4nfF4+RM=_S9Q-bi20-rjcZwUWN7qR@mail.gmail.com>

Problem solved by <param name="extension-in-contact" value="true"/>
Thx!

2010/10/20 Maxim Balabaev <balabaev.m at gmail.com>

> Here is valid registration log:
> http://pastebin.com/mhkDuCWG
> I really think that the problem can only come from gw+ in fs contact header
>
> 2010/10/19 Maxim Balabaev <balabaev.m at gmail.com>
>
> Here is asterisk invite dump:
>> http://pastebin.com/Me53C1kV
>> (I can't give any FS dump because there is just nothing)
>> Fs reg dump:
>> http://pastebin.com/63z5HPAS
>> May be problem is in contact header? I think i have seen this problem
>> somewhere...
>> 2010/10/19 Brian West <brian at freeswitch.org>
>>
>> Without sip traces of any other valid information we can only guess...
>>>
>>> /b
>>>
>>> On Oct 19, 2010, at 9:42 AM, Maxim Balabaev wrote:
>>>
>>> > I`m having really strange problem. I connect new provider, changed all
>>> gateway settings but i can`t take calls. While debuggins external and so as
>>> global no invite packet is showing up. I even tryed tcpdump - absolutely
>>> nothing. Just pings. Outcoming are ok, register is ok. If I switch to
>>> asterisk - no problem occures. Support says that server refuses connection,
>>> while another gateway works perfectly. I don`t know what to do...
>>> > May be this can be usefull to. Prov have multiple realms on server and
>>> accepts only @myprov.com format. While to asterisk invite is thrown with
>>> sipid at myip. I tryed changing server hostname to ip, but still nothing.
>>> > Here is my conf:
>>> > http://pastebin.com/qKqAv38a
>>> > Really appritiate any help.
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
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From yivzhenko at mksat.net  Wed Oct 20 00:27:59 2010
From: yivzhenko at mksat.net (Yuriy Ivzhenko)
Date: Wed, 20 Oct 2010 10:27:59 +0300
Subject: [Freeswitch-users] mod_nibblebill hangs up A-Leg and continues
	dialing B-Leg
In-Reply-To: <CB3C8C74-B8AF-4503-B449-22CF2E3E63CB@archerdrive.com>
References: <CB3C8C74-B8AF-4503-B449-22CF2E3E63CB@archerdrive.com>
Message-ID: <201010201027.59694.yivzhenko@mksat.net>

On Monday 18 October 2010 05:47:48 Ash wrote:
> Hi All,
> 
> I am trying to setup mod nibble bill to bill based on the B-Leg.  I have
>  followed the wiki but have been unable to resolve my issue.  I first had
>  this issue with 1.0.6 and tried the latest git version and can reproduce
>  the fault on both versions.
> 
> When the customers credit is 0 or below the handset I am using to test gets
>  transferred to the hangup destination and the A-Leg is hungup.  After a
>  couple of seconds my mobile will ring and when I answer it there is
>  silence so a one way call leg.  As anyone else seen this behaviour?
> 

I have seen this problem long ago.
I have replaced 1 line in a code.
It is not assured that this correct decision, but for me it works.

some strings from my patch to mod_nibblebill.c:

        switch_ivr_media(uuid, SMF_REBRIDGE);

        /* Transfer the A leg */
-       switch_ivr_session_transfer(session, argv[0], argv[1], argv[2]);
+//     switch_ivr_session_transfer(session, argv[0], argv[1], argv[2]);
+       switch_channel_hangup(channel, SWITCH_CAUSE_NORMAL_CLEARING);
        free(mydup);
 }


From curriegrad2004 at gmail.com  Wed Oct 20 11:11:16 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Wed, 20 Oct 2010 11:11:16 -0700
Subject: [Freeswitch-users] G.729 & Windows
In-Reply-To: <1287585703594-5655162.post@n2.nabble.com>
References: <4CBEBE4F.8040107@chaschperli.ch>
	<1287582251418-5654913.post@n2.nabble.com>
	<AANLkTim2emW1C5oeS6hCsrgUSMkgeEudieKWY8AEWfMh@mail.gmail.com>
	<1287585703594-5655162.post@n2.nabble.com>
Message-ID: <AANLkTin6gUE9AxmtLeDZ_FQKmeLSwfNyaZSgkntDStNZ@mail.gmail.com>

FreeSwitch does have an option to compile a full g.729 codec for
Windows from the source code, however I doubt that code has even been
tested at all. I've been there and no, so far I haven't been
successful at all on getting it work work properly.

On Wed, Oct 20, 2010 at 7:41 AM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
>
> Steven Ayre wrote:
>>
>>> Honestly, the reason I use G729 CoDec is to save some
>>> Internet bandwidth, especially communicating with others in some 3-rd
>>> world
>>> countries where their broadband Internet connections are still metered
>>> per
>>> their usage/month.
>>
>> Although it's far from the only low bandwidth codec - Speex, GSM etc.
> This may not be a bad idea for those who can use them. Unfortunately, none
> of my ATA devices has a support on these two CoDecs.
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/G-729-Windows-tp5654175p5655162.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Wed Oct 20 11:50:08 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 20 Oct 2010 11:50:08 -0700
Subject: [Freeswitch-users] Call for wiki help: cleanup OpenZAP pages on wiki
Message-ID: <AANLkTiks-kt_wqopLnwbYkX0qf2_btPAJj=6LkCuFPG1@mail.gmail.com>

We need some assistance with the migration from OpenZAP to FreeTDM. In
particular we have some legacy OpenZAP pages that need to be translated into
FreeTDM pages on the wiki. In some cases it may be as simple as swapping
"OpenZAP" with "FreeTDM" and doing a page redirect. In other cases it may be
a bit more involved. I am inviting anyone with OpenZAP and FreeTDM
experience to work on these wiki pages:

http://wiki.freeswitch.org/wiki/Openzap.conf_Examples
http://wiki.freeswitch.org/wiki/Openzap.conf.xml_Examples
http://wiki.freeswitch.org/wiki/Configuration_OpenZAP-_DigiumTDM400P_Example
http://wiki.freeswitch.org/wiki/Openzap.sangoma.libpri
http://wiki.freeswitch.org/wiki/OpenZap_Dahdi
http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2
http://wiki.freeswitch.org/wiki/OpenZAP_Rhino

Please contact me off list if you have any questions. Also, Moises Silva
(IRC: moy) has done a really nice job on the FreeTDM wiki page:
http://wiki.freeswitch.org/wiki/FreeTDM. He is a good source of information.
We have created #freetdm on irc.freenode.net so please don't use #openzap
any more.

Thanks!

-Michael
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From msc at freeswitch.org  Wed Oct 20 12:02:21 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 20 Oct 2010 12:02:21 -0700
Subject: [Freeswitch-users] recording into perl script
In-Reply-To: <20101020140122.EA9EF11519@mail.nstel.ru>
References: <20101020140122.EA9EF11519@mail.nstel.ru>
Message-ID: <AANLkTimQz43oTm8whepWKJr39LrOjb8ofpcEVZbYvdPU@mail.gmail.com>

Definitely record it to a file first and then dump it into your database. I
recommend you record the file with the filename as you have it and then use
api_hangup_hook to launch the cleanup script that puts the recording into
the database.

Note: you probably want to create a chan var with the file name and use the
inline="true" tag so that you pass it to your script:

<action application="set"
data="record_name=$${base_dir}/recordings/${strftime(foo)}.wav"
inline="true"/>
<action application="set" data="api_hangup_hook=perl
my_script.pl${record_name}"/>

-MC

On Wed, Oct 20, 2010 at 7:01 AM, Nikolay Kondratyev <kond at nstel.ru> wrote:

>  Hi all,
> Currently i can record a call via
> <action application="record_session"
> data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
> But in this way i can only record to a file.
> I have some experience with perl and mysql.
> I'd like to record a call into database (mysql or postgre).
> I hope it's possible via mod_perl... is it?
> If yes, how can i pass voice into perl script?
>
> I guess that i can just do the following
> <action application="record_session"
> data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
> <action application="perl" data="recorder.pl$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"
> />
> So that my perl script recorder.pl will get recorded file name as it's
> first argument.
> But when my script starts, will the recording allready be finished?
>
> In this way (if it will work at all) i will first record a call in a file,
> and then put this file into database.
> Is there a way to write directly into database?
> Something like <action application="record_session" data=" |recorder.pl"
> /> ?
>
> Thanks in advance,
> Nikolay.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bwibowo at gmail.com  Wed Oct 20 15:03:24 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Thu, 21 Oct 2010 05:03:24 +0700
Subject: [Freeswitch-users] mod_nibblebill
In-Reply-To: <AANLkTinT24jE84JJ+-VX3wJYQaZpEOPM5H9kJwkFNxqa@mail.gmail.com>
References: <AANLkTi=aTvLC=k6wFnD7VbjAY1q22dgrEfG2QiB4-ojn@mail.gmail.com>
	<AANLkTi=unUZ8a-pULJeTnNS91wqrJuR90mjRi_qLQyag@mail.gmail.com>
	<AANLkTiksB-RfjGmTundCVgJyqQcG9w5W3iXuYXC9Fard@mail.gmail.com>
	<AANLkTinT24jE84JJ+-VX3wJYQaZpEOPM5H9kJwkFNxqa@mail.gmail.com>
Message-ID: <AANLkTinE_aF1f9EzEqS92LLws+PPwG2g2AYMovkT1aWp@mail.gmail.com>

hi
for nibblebill.conf.xml i have this
<configuration name="nibblebill.conf" description="Nibble Billing">
  <settings>
    <!-- See http://wiki.freeswitch.org/wiki/Mod_nibblebill for help with
these options -->

    <!-- Information for connecting to your database -->
    <param name="db_username" value="root"/>
    <param name="db_password" value="rootpass"/>
    <param name="db_dsn" value="fs_conect"/>

also fs_connect is recognized when i issue command isql

please help



On Wed, Oct 20, 2010 at 3:15 PM, Nyamul Hassan <mnhassan at usa.net> wrote:

> Check the debug log you posted earlier. The first line says all about it.
>
> For nibble bill, you put it in the conf.xml file for nibble bill.
>
> Regards
> HASSAN
>
>
> On 2010-10-20, budi wibowo <bwibowo at gmail.com> wrote:
> > hi hassan
> > dns name is fs_connect, where i hould put that parameter?
> >
> >
> > regards
> >
> > budi
> >
> > On Wed, Oct 20, 2010 at 2:25 PM, Nyamul Hassan <mnhassan at usa.net> wrote:
> >
> >> As per the debug logs, the name of the dsn is missing a "n" inside
> >> FreeSWITCH.
> >>
> >> Regards
> >> HASSAN
> >>
> >>
> >> On 2010-10-20, budi wibowo <bwibowo at gmail.com> wrote:
> >> > dear all
> >> >
> >> > i install mod_nibblebil on centos 5.5 according to steps in
> >> > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
> >> >
> >> > odbc.ini config:
> >> > [fs_connect]
> >> > Driver   = mySQL
> >> > SERVER   = localhost
> >> > PORT     = 3306
> >> > DATABASE = fs
> >> > OPTION  = 67108864
> >> > Socket   = /var/lib/mysql/mysql.sock
> >> >
> >> > switch.conf.xml i add this
> >> >
> >> >  <param name="odbc-dsn" value="fs_connect:root:rootpass" />
> >> >
> >> > if i execute isql fs_connect root rootpass it's the result
> >> > +---------------------------------------+
> >> > | Connected!                            |
> >> > |                                       |
> >> > | sql-statement                         |
> >> > | help [tablename]                      |
> >> > | quit                                  |
> >> > |                                       |
> >> > +---------------------------------------+
> >> > SQL>
> >> >
> >> >
> >> > but i found this error, where's my mistake ?
> >> >
> >> > i 2010-10-20 07:20:39.411254 [CRIT] switch_odbc.c:210 The sql server
> is
> >> not
> >> > responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR:
> >> [unixODBC][Driver
> >> > Manager]Connnection does not exist
> >> > ][169]
> >> > 2010-10-20 07:20:39.411254 [CRIT] switch_odbc.c:218 The connection
> could
> >> not
> >> > be re-established
> >> > 2010-10-20 07:20:40.412241 [ERR] switch_odbc.c:318 STATE: IM002 CODE 0
> >> > ERROR: [unixODBC][Driver Manager]Data source name not found, and no
> >> default
> >> > driver specified
> >> >
> >> > 2010-10-20 07:20:40.412241 [CRIT] switch_odbc.c:210 The sql server is
> >> > not
> >> > responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR:
> >> [unixODBC][Driver
> >> > Manager]Connnection does not exist
> >> > ][169]
> >> > 2010-10-20 07:20:40.412241 [CRIT] switch_odbc.c:218 The connection
> could
> >> not
> >> > be re-established
> >> > 2010-10-20 07:20:41.412257 [ERR] switch_odbc.c:318 STATE: IM002 CODE 0
> >> > ERROR: [unixODBC][Driver Manager]Data source name not found, and no
> >> default
> >> > driver specified
> >> >
> >> > 2010-10-20 07:20:41.412257 [CRIT] switch_odbc.c:210 The sql server is
> >> > not
> >> > responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR:
> >> [unixODBC][Driver
> >> > Manager]Connnection does not exist
> >> >
> >> >
> >> >
> >> > regards
> >> >
> >> > budi
> >> >
> >>
> >> --
> >> Sent from my mobile device
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
>
> --
> Sent from my mobile device
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sos at sokhapkin.dyndns.org  Wed Oct 20 15:16:37 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 20 Oct 2010 18:16:37 -0400
Subject: [Freeswitch-users] mod_nibblebill
In-Reply-To: <AANLkTinE_aF1f9EzEqS92LLws+PPwG2g2AYMovkT1aWp@mail.gmail.com>
References: <AANLkTi=aTvLC=k6wFnD7VbjAY1q22dgrEfG2QiB4-ojn@mail.gmail.com>
	<AANLkTinT24jE84JJ+-VX3wJYQaZpEOPM5H9kJwkFNxqa@mail.gmail.com>
	<AANLkTinE_aF1f9EzEqS92LLws+PPwG2g2AYMovkT1aWp@mail.gmail.com>
Message-ID: <201010201816.37895.sos@sokhapkin.dyndns.org>

People told you many times already:

<param name="db_dsn" value="fs_conect"/>

On Wednesday 20 October 2010, budi wibowo wrote:
> hi
> for nibblebill.conf.xml i have this
> <configuration name="nibblebill.conf" description="Nibble Billing">
>   <settings>
>     <!-- See http://wiki.freeswitch.org/wiki/Mod_nibblebill for help with
> these options -->
> 
>     <!-- Information for connecting to your database -->
>     <param name="db_username" value="root"/>
>     <param name="db_password" value="rootpass"/>
>     <param name="db_dsn" value="fs_conect"/>
> 
> also fs_connect is recognized when i issue command isql
> 
> please help
> 
> On Wed, Oct 20, 2010 at 3:15 PM, Nyamul Hassan <mnhassan at usa.net> wrote:
> > Check the debug log you posted earlier. The first line says all about it.
> >
> > For nibble bill, you put it in the conf.xml file for nibble bill.
> >
> > Regards
> > HASSAN
> >
> > On 2010-10-20, budi wibowo <bwibowo at gmail.com> wrote:
> > > hi hassan
> > > dns name is fs_connect, where i hould put that parameter?
> > >
> > >
> > > regards
> > >
> > > budi
> > >
> > > On Wed, Oct 20, 2010 at 2:25 PM, Nyamul Hassan <mnhassan at usa.net> wrote:
> > >> As per the debug logs, the name of the dsn is missing a "n" inside
> > >> FreeSWITCH.
> > >>
> > >> Regards
> > >> HASSAN
> > >>
> > >> On 2010-10-20, budi wibowo <bwibowo at gmail.com> wrote:
> > >> > dear all
> > >> >
> > >> > i install mod_nibblebil on centos 5.5 according to steps in
> > >> > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
> > >> >
> > >> > odbc.ini config:
> > >> > [fs_connect]
> > >> > Driver   = mySQL
> > >> > SERVER   = localhost
> > >> > PORT     = 3306
> > >> > DATABASE = fs
> > >> > OPTION  = 67108864
> > >> > Socket   = /var/lib/mysql/mysql.sock
> > >> >
> > >> > switch.conf.xml i add this
> > >> >
> > >> >  <param name="odbc-dsn" value="fs_connect:root:rootpass" />
> > >> >
> > >> > if i execute isql fs_connect root rootpass it's the result
> > >> > +---------------------------------------+
> > >> >
> > >> > | Connected!                            |
> > >> > |
> > >> > | sql-statement                         |
> > >> > | help [tablename]                      |
> > >> > | quit                                  |
> > >> >
> > >> > +---------------------------------------+
> > >> > SQL>
> > >> >
> > >> >
> > >> > but i found this error, where's my mistake ?
> > >> >
> > >> > i 2010-10-20 07:20:39.411254 [CRIT] switch_odbc.c:210 The sql server
> >
> > is
> >
> > >> not
> > >>
> > >> > responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR:
> > >>
> > >> [unixODBC][Driver
> > >>
> > >> > Manager]Connnection does not exist
> > >> > ][169]
> > >> > 2010-10-20 07:20:39.411254 [CRIT] switch_odbc.c:218 The connection
> >
> > could
> >
> > >> not
> > >>
> > >> > be re-established
> > >> > 2010-10-20 07:20:40.412241 [ERR] switch_odbc.c:318 STATE: IM002 CODE
> > >> > 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and
> > >> > no
> > >>
> > >> default
> > >>
> > >> > driver specified
> > >> >
> > >> > 2010-10-20 07:20:40.412241 [CRIT] switch_odbc.c:210 The sql server
> > >> > is not
> > >> > responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR:
> > >>
> > >> [unixODBC][Driver
> > >>
> > >> > Manager]Connnection does not exist
> > >> > ][169]
> > >> > 2010-10-20 07:20:40.412241 [CRIT] switch_odbc.c:218 The connection
> >
> > could
> >
> > >> not
> > >>
> > >> > be re-established
> > >> > 2010-10-20 07:20:41.412257 [ERR] switch_odbc.c:318 STATE: IM002 CODE
> > >> > 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and
> > >> > no
> > >>
> > >> default
> > >>
> > >> > driver specified
> > >> >
> > >> > 2010-10-20 07:20:41.412257 [CRIT] switch_odbc.c:210 The sql server
> > >> > is not
> > >> > responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR:
> > >>
> > >> [unixODBC][Driver
> > >>
> > >> > Manager]Connnection does not exist
> > >> >
> > >> >
> > >> >
> > >> > regards
> > >> >
> > >> > budi
> > >>
> > >> --
> > >> Sent from my mobile device
> > >>
> > >> _______________________________________________
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >> http://www.freeswitch.org
> >
> > --
> > Sent from my mobile device
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 

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From mnhassan at usa.net  Wed Oct 20 16:40:22 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Thu, 21 Oct 2010 05:40:22 +0600
Subject: [Freeswitch-users] mod_nibblebill
In-Reply-To: <AANLkTinE_aF1f9EzEqS92LLws+PPwG2g2AYMovkT1aWp@mail.gmail.com>
References: <AANLkTi=aTvLC=k6wFnD7VbjAY1q22dgrEfG2QiB4-ojn@mail.gmail.com>
	<AANLkTi=unUZ8a-pULJeTnNS91wqrJuR90mjRi_qLQyag@mail.gmail.com>
	<AANLkTiksB-RfjGmTundCVgJyqQcG9w5W3iXuYXC9Fard@mail.gmail.com>
	<AANLkTinT24jE84JJ+-VX3wJYQaZpEOPM5H9kJwkFNxqa@mail.gmail.com>
	<AANLkTinE_aF1f9EzEqS92LLws+PPwG2g2AYMovkT1aWp@mail.gmail.com>
Message-ID: <AANLkTinrA8C3n0+9-wsRH3niCxcy4mmQ2Lcai4RMJZ2N@mail.gmail.com>

Check the lines you posted. You still have that spelling problem.

Regards
HASSAN


On 2010-10-21, budi wibowo <bwibowo at gmail.com> wrote:
> hi
> for nibblebill.conf.xml i have this
> <configuration name="nibblebill.conf" description="Nibble Billing">
>   <settings>
>     <!-- See http://wiki.freeswitch.org/wiki/Mod_nibblebill for help with
> these options -->
>
>     <!-- Information for connecting to your database -->
>     <param name="db_username" value="root"/>
>     <param name="db_password" value="rootpass"/>
>     <param name="db_dsn" value="fs_conect"/>
>
> also fs_connect is recognized when i issue command isql
>
> please help
>
>
>
> On Wed, Oct 20, 2010 at 3:15 PM, Nyamul Hassan <mnhassan at usa.net> wrote:
>
>> Check the debug log you posted earlier. The first line says all about it.
>>
>> For nibble bill, you put it in the conf.xml file for nibble bill.
>>
>> Regards
>> HASSAN
>>
>>
>> On 2010-10-20, budi wibowo <bwibowo at gmail.com> wrote:
>> > hi hassan
>> > dns name is fs_connect, where i hould put that parameter?
>> >
>> >
>> > regards
>> >
>> > budi
>> >
>> > On Wed, Oct 20, 2010 at 2:25 PM, Nyamul Hassan <mnhassan at usa.net> wrote:
>> >
>> >> As per the debug logs, the name of the dsn is missing a "n" inside
>> >> FreeSWITCH.
>> >>
>> >> Regards
>> >> HASSAN
>> >>
>> >>
>> >> On 2010-10-20, budi wibowo <bwibowo at gmail.com> wrote:
>> >> > dear all
>> >> >
>> >> > i install mod_nibblebil on centos 5.5 according to steps in
>> >> > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
>> >> >
>> >> > odbc.ini config:
>> >> > [fs_connect]
>> >> > Driver   = mySQL
>> >> > SERVER   = localhost
>> >> > PORT     = 3306
>> >> > DATABASE = fs
>> >> > OPTION  = 67108864
>> >> > Socket   = /var/lib/mysql/mysql.sock
>> >> >
>> >> > switch.conf.xml i add this
>> >> >
>> >> >  <param name="odbc-dsn" value="fs_connect:root:rootpass" />
>> >> >
>> >> > if i execute isql fs_connect root rootpass it's the result
>> >> > +---------------------------------------+
>> >> > | Connected!                            |
>> >> > |                                       |
>> >> > | sql-statement                         |
>> >> > | help [tablename]                      |
>> >> > | quit                                  |
>> >> > |                                       |
>> >> > +---------------------------------------+
>> >> > SQL>
>> >> >
>> >> >
>> >> > but i found this error, where's my mistake ?
>> >> >
>> >> > i 2010-10-20 07:20:39.411254 [CRIT] switch_odbc.c:210 The sql server
>> is
>> >> not
>> >> > responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR:
>> >> [unixODBC][Driver
>> >> > Manager]Connnection does not exist
>> >> > ][169]
>> >> > 2010-10-20 07:20:39.411254 [CRIT] switch_odbc.c:218 The connection
>> could
>> >> not
>> >> > be re-established
>> >> > 2010-10-20 07:20:40.412241 [ERR] switch_odbc.c:318 STATE: IM002 CODE
>> >> > 0
>> >> > ERROR: [unixODBC][Driver Manager]Data source name not found, and no
>> >> default
>> >> > driver specified
>> >> >
>> >> > 2010-10-20 07:20:40.412241 [CRIT] switch_odbc.c:210 The sql server is
>> >> > not
>> >> > responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR:
>> >> [unixODBC][Driver
>> >> > Manager]Connnection does not exist
>> >> > ][169]
>> >> > 2010-10-20 07:20:40.412241 [CRIT] switch_odbc.c:218 The connection
>> could
>> >> not
>> >> > be re-established
>> >> > 2010-10-20 07:20:41.412257 [ERR] switch_odbc.c:318 STATE: IM002 CODE
>> >> > 0
>> >> > ERROR: [unixODBC][Driver Manager]Data source name not found, and no
>> >> default
>> >> > driver specified
>> >> >
>> >> > 2010-10-20 07:20:41.412257 [CRIT] switch_odbc.c:210 The sql server is
>> >> > not
>> >> > responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR:
>> >> [unixODBC][Driver
>> >> > Manager]Connnection does not exist
>> >> >
>> >> >
>> >> >
>> >> > regards
>> >> >
>> >> > budi
>> >> >
>> >>
>> >> --
>> >> Sent from my mobile device
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>>
>> --
>> Sent from my mobile device
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>

-- 
Sent from my mobile device


From bwibowo at gmail.com  Wed Oct 20 16:41:51 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Thu, 21 Oct 2010 06:41:51 +0700
Subject: [Freeswitch-users] mod_nibblebill
In-Reply-To: <201010201816.37895.sos@sokhapkin.dyndns.org>
References: <AANLkTi=aTvLC=k6wFnD7VbjAY1q22dgrEfG2QiB4-ojn@mail.gmail.com>
	<AANLkTinT24jE84JJ+-VX3wJYQaZpEOPM5H9kJwkFNxqa@mail.gmail.com>
	<AANLkTinE_aF1f9EzEqS92LLws+PPwG2g2AYMovkT1aWp@mail.gmail.com>
	<201010201816.37895.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTin3YrhLjTMyHeMhcbxD9uLNwXxoMOJ40z=BskMN@mail.gmail.com>

ok thx a lot .. solved now

On Thu, Oct 21, 2010 at 5:16 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

>  People told you many times already:
>
> <param name="db_dsn" value="fs_conect"/>
>
> On Wednesday 20 October 2010, budi wibowo wrote:
>
> > hi
>
> > for nibblebill.conf.xml i have this
>
> > <configuration name="nibblebill.conf" description="Nibble Billing">
>
> > <settings>
>
> > <!-- See http://wiki.freeswitch.org/wiki/Mod_nibblebill for help with
>
> > these options -->
>
> >
>
> > <!-- Information for connecting to your database -->
>
> > <param name="db_username" value="root"/>
>
> > <param name="db_password" value="rootpass"/>
>
> > <param name="db_dsn" value="fs_conect"/>
>
> >
>
> > also fs_connect is recognized when i issue command isql
>
> >
>
> > please help
>
> >
>
> > On Wed, Oct 20, 2010 at 3:15 PM, Nyamul Hassan <mnhassan at usa.net> wrote:
>
> > > Check the debug log you posted earlier. The first line says all about
> it.
>
> > >
>
> > > For nibble bill, you put it in the conf.xml file for nibble bill.
>
> > >
>
> > > Regards
>
> > > HASSAN
>
> > >
>
> > > On 2010-10-20, budi wibowo <bwibowo at gmail.com> wrote:
>
> > > > hi hassan
>
> > > > dns name is fs_connect, where i hould put that parameter?
>
> > > >
>
> > > >
>
> > > > regards
>
> > > >
>
> > > > budi
>
> > > >
>
> > > > On Wed, Oct 20, 2010 at 2:25 PM, Nyamul Hassan <mnhassan at usa.net>
> wrote:
>
> > > >> As per the debug logs, the name of the dsn is missing a "n" inside
>
> > > >> FreeSWITCH.
>
> > > >>
>
> > > >> Regards
>
> > > >> HASSAN
>
> > > >>
>
> > > >> On 2010-10-20, budi wibowo <bwibowo at gmail.com> wrote:
>
> > > >> > dear all
>
> > > >> >
>
> > > >> > i install mod_nibblebil on centos 5.5 according to steps in
>
> > > >> > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
>
> > > >> >
>
> > > >> > odbc.ini config:
>
> > > >> > [fs_connect]
>
> > > >> > Driver = mySQL
>
> > > >> > SERVER = localhost
>
> > > >> > PORT = 3306
>
> > > >> > DATABASE = fs
>
> > > >> > OPTION = 67108864
>
> > > >> > Socket = /var/lib/mysql/mysql.sock
>
> > > >> >
>
> > > >> > switch.conf.xml i add this
>
> > > >> >
>
> > > >> > <param name="odbc-dsn" value="fs_connect:root:rootpass" />
>
> > > >> >
>
> > > >> > if i execute isql fs_connect root rootpass it's the result
>
> > > >> > +---------------------------------------+
>
> > > >> >
>
> > > >> > | Connected! |
>
> > > >> > |
>
> > > >> > | sql-statement |
>
> > > >> > | help [tablename] |
>
> > > >> > | quit |
>
> > > >> >
>
> > > >> > +---------------------------------------+
>
> > > >> > SQL>
>
> > > >> >
>
> > > >> >
>
> > > >> > but i found this error, where's my mistake ?
>
> > > >> >
>
> > > >> > i 2010-10-20 07:20:39.411254 [CRIT] switch_odbc.c:210 The sql
> server
>
> > >
>
> > > is
>
> > >
>
> > > >> not
>
> > > >>
>
> > > >> > responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR:
>
> > > >>
>
> > > >> [unixODBC][Driver
>
> > > >>
>
> > > >> > Manager]Connnection does not exist
>
> > > >> > ][169]
>
> > > >> > 2010-10-20 07:20:39.411254 [CRIT] switch_odbc.c:218 The connection
>
> > >
>
> > > could
>
> > >
>
> > > >> not
>
> > > >>
>
> > > >> > be re-established
>
> > > >> > 2010-10-20 07:20:40.412241 [ERR] switch_odbc.c:318 STATE: IM002
> CODE
>
> > > >> > 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and
>
> > > >> > no
>
> > > >>
>
> > > >> default
>
> > > >>
>
> > > >> > driver specified
>
> > > >> >
>
> > > >> > 2010-10-20 07:20:40.412241 [CRIT] switch_odbc.c:210 The sql server
>
> > > >> > is not
>
> > > >> > responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR:
>
> > > >>
>
> > > >> [unixODBC][Driver
>
> > > >>
>
> > > >> > Manager]Connnection does not exist
>
> > > >> > ][169]
>
> > > >> > 2010-10-20 07:20:40.412241 [CRIT] switch_odbc.c:218 The connection
>
> > >
>
> > > could
>
> > >
>
> > > >> not
>
> > > >>
>
> > > >> > be re-established
>
> > > >> > 2010-10-20 07:20:41.412257 [ERR] switch_odbc.c:318 STATE: IM002
> CODE
>
> > > >> > 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and
>
> > > >> > no
>
> > > >>
>
> > > >> default
>
> > > >>
>
> > > >> > driver specified
>
> > > >> >
>
> > > >> > 2010-10-20 07:20:41.412257 [CRIT] switch_odbc.c:210 The sql server
>
> > > >> > is not
>
> > > >> > responding for DSN fs_conect [STATE: 08003 CODE 0 ERROR:
>
> > > >>
>
> > > >> [unixODBC][Driver
>
> > > >>
>
> > > >> > Manager]Connnection does not exist
>
> > > >> >
>
> > > >> >
>
> > > >> >
>
> > > >> > regards
>
> > > >> >
>
> > > >> > budi
>
> > > >>
>
> > > >> --
>
> > > >> Sent from my mobile device
>
> > > >>
>
> > > >> _______________________________________________
>
> > > >> FreeSWITCH-users mailing list
>
> > > >> FreeSWITCH-users at lists.freeswitch.org
>
> > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> > > >> UNSUBSCRIBE:
>
> > >
>
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> > >
>
> > > >> http://www.freeswitch.org
>
> > >
>
> > > --
>
> > > Sent from my mobile device
>
> > >
>
> > > _______________________________________________
>
> > > FreeSWITCH-users mailing list
>
> > > FreeSWITCH-users at lists.freeswitch.org
>
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> > > http://www.freeswitch.org
>
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mustafa.pk at gmail.com  Wed Oct 20 22:55:59 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Thu, 21 Oct 2010 10:55:59 +0500
Subject: [Freeswitch-users] record on demand for a conference call
In-Reply-To: <AANLkTimLQ_N=Gr+azdnc9Y-ubwN6v5qmhiF8sSX8cKOp@mail.gmail.com>
References: <6DFF80D26E14467299C2B7789D1D82D6@D810>
	<AANLkTimuurW-3SJX1+JB=rzbgmq1mGd12hQEvzmfD4A6@mail.gmail.com>
	<AANLkTi=3mHhqtWbLiw+onv8MDAWGLF+0mhKt-M6kiWBe@mail.gmail.com>
	<AANLkTimLQ_N=Gr+azdnc9Y-ubwN6v5qmhiF8sSX8cKOp@mail.gmail.com>
Message-ID: <AANLkTikApLtk8d+SrYqBk1Fqp8SpF8tzK-tSLw8rbNMo@mail.gmail.com>

you can do it via ESL, whenever a conference is created auto kick in a
*reocording* extension into the conference.


On Wed, Oct 20, 2010 at 1:06 AM, Michael Collins <msc at freeswitch.org> wrote:
> You can definitely write an ESL app to handle this. It's totally DIY, but FS
> gives you all the tools and parts.
> -MC
>
> On Tue, Oct 19, 2010 at 2:31 AM, Andy Ayers <andy at fabulous4.co.uk> wrote:
>>
>> thanks for your interest Michael, it would be to allow the moderator to
>> record an introduction to the call so when it's listened to by others it has
>> context. But actually more useful is to be able to start recording when the
>> moderator wants it to to cut off the usually messy start to a conference
>> call. Is this possible, could I configure a dtmf key press to initiate the
>> recording somehow?
>>
>> On 13 October 2010 20:05, Michael Collins <msc at freeswitch.org> wrote:
>>>
>>> Just curious - what value is there to recording the call when only one
>>> person is there?
>>> -MC
>>>
>>> On Wed, Oct 13, 2010 at 3:42 AM, Andy <andy at fabulous4.co.uk> wrote:
>>>>
>>>> Hi,
>>>>
>>>> I would like to set up my conferencing system so that it's possible to
>>>> initiate the recording of the conference when?the moderator presses a key on
>>>> their keypad.
>>>>
>>>> Failing that I would like to know if there is any way of recording the
>>>> entire conference call from the moment the first caller arrives rather than
>>>> using the auto-record feature which only kicks in when 2 or more people are
>>>> connected.
>>>>
>>>> Can anyone give me some idea how this can be done or point me to the
>>>> right wiki pages?
>>>>
>>>> Many thanks for you help
>>>> Andy
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com


From thomas at chaschperli.ch  Wed Oct 20 23:28:03 2010
From: thomas at chaschperli.ch (Thomas Mueller)
Date: Thu, 21 Oct 2010 08:28:03 +0200
Subject: [Freeswitch-users] G.729 & Windows
In-Reply-To: <1287582251418-5654913.post@n2.nabble.com>
References: <4CBEBE4F.8040107@chaschperli.ch>
	<1287582251418-5654913.post@n2.nabble.com>
Message-ID: <4CBFDD73.1020709@chaschperli.ch>


> Just curious. If a G729 CoDec is a MUST and unless you prefer to use a SIP
> compliant ATA device that already comes with a built-in G729 CoDec
i've had good results with G.729 with audio quality and bandwith - 
especially with ADSL where uplink is the bottleneck. My hardware-sip 
phones (Snom, Siemens) are equiped with the g729 codec, the sip provider 
too.

- Thomas








From mnhassan at usa.net  Thu Oct 21 00:30:13 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Thu, 21 Oct 2010 13:30:13 +0600
Subject: [Freeswitch-users] G.729 & Windows
In-Reply-To: <4CBFDD73.1020709@chaschperli.ch>
References: <4CBEBE4F.8040107@chaschperli.ch>
	<1287582251418-5654913.post@n2.nabble.com>
	<4CBFDD73.1020709@chaschperli.ch>
Message-ID: <AANLkTikCFkD_zstn_eO3uCB9D_H4pEed2ZL2pfcfAU8Z@mail.gmail.com>

Any one know of a comparable low bandwidth codec that works for mobile
based softphones?

Regards
HASSAN


On 2010-10-21, Thomas Mueller <thomas at chaschperli.ch> wrote:
>
>> Just curious. If a G729 CoDec is a MUST and unless you prefer to use a SIP
>> compliant ATA device that already comes with a built-in G729 CoDec
> i've had good results with G.729 with audio quality and bandwith -
> especially with ADSL where uplink is the bottleneck. My hardware-sip
> phones (Snom, Siemens) are equiped with the g729 codec, the sip provider
> too.
>
> - Thomas
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device


From monemran at gmail.com  Thu Oct 21 01:03:01 2010
From: monemran at gmail.com (M.Emran)
Date: Thu, 21 Oct 2010 14:03:01 +0600
Subject: [Freeswitch-users] G.729 & Windows
In-Reply-To: <AANLkTikCFkD_zstn_eO3uCB9D_H4pEed2ZL2pfcfAU8Z@mail.gmail.com>
References: <4CBEBE4F.8040107@chaschperli.ch>
	<1287582251418-5654913.post@n2.nabble.com>
	<4CBFDD73.1020709@chaschperli.ch>
	<AANLkTikCFkD_zstn_eO3uCB9D_H4pEed2ZL2pfcfAU8Z@mail.gmail.com>
Message-ID: <AANLkTikUBrCYTRvm7Mt0_Cw6pRfeTwY94oDRumSd-82v@mail.gmail.com>

in ARM processor, you can do g729.

On Thu, Oct 21, 2010 at 1:30 PM, Nyamul Hassan <mnhassan at usa.net> wrote:

> Any one know of a comparable low bandwidth codec that works for mobile
> based softphones?
>
> Regards
> HASSAN
>
>
> On 2010-10-21, Thomas Mueller <thomas at chaschperli.ch> wrote:
> >
> >> Just curious. If a G729 CoDec is a MUST and unless you prefer to use a
> SIP
> >> compliant ATA device that already comes with a built-in G729 CoDec
> > i've had good results with G.729 with audio quality and bandwith -
> > especially with ADSL where uplink is the bottleneck. My hardware-sip
> > phones (Snom, Siemens) are equiped with the g729 codec, the sip provider
> > too.
> >
> > - Thomas
> >
> >
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> --
> Sent from my mobile device
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Regards
----------
M Emran
Chief Executive Officer
E-SOFT BILLING (pvt). LTD.

HB Tower(3rd Floor)
House No # 1/A
Road No # 23
Gulshan # 1
Dhaka-1212, Bangladesh.
*Phone:* +880-2-8822312,+880-2-8822384
Fax : +880-2-8822254
E-Mail: info at e-softbilling.com
Web: www.e-softbilling.com
         www.isoftswitch.com
         www.howtonix.com
         www.sipmobiledialer.com
-------------- next part --------------
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From m.kelly at mundio.com  Wed Oct 20 15:14:34 2010
From: m.kelly at mundio.com (Magnus Kelly)
Date: Wed, 20 Oct 2010 23:14:34 +0100
Subject: [Freeswitch-users] FS UK Ringback
Message-ID: <8450FE01FF743446A9E2FDCCDE3B323201975CFE4D17@MAIL-ROMEO.squay.com>

Hello all,

Would anyone be able to help  with either a FS friendly audio file for UK ring back, or the definitions to create it natively as my current FS setup is not generating the correct UK tones.

Currently its defined as below, however its sounding like stuttered US dial tone as opposed as normal UK ringback.

<X-PRE-PROCESS cmd="set" data="uk-ring=%(400,200,400,450);%(400,2200,400,450)"/>

Many thanks
Magnus

From mnhassan at usa.net  Thu Oct 21 07:34:31 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Thu, 21 Oct 2010 20:34:31 +0600
Subject: [Freeswitch-users] G.729 & Windows
In-Reply-To: <AANLkTikUBrCYTRvm7Mt0_Cw6pRfeTwY94oDRumSd-82v@mail.gmail.com>
References: <4CBEBE4F.8040107@chaschperli.ch>
	<1287582251418-5654913.post@n2.nabble.com>
	<4CBFDD73.1020709@chaschperli.ch>
	<AANLkTikCFkD_zstn_eO3uCB9D_H4pEed2ZL2pfcfAU8Z@mail.gmail.com>
	<AANLkTikUBrCYTRvm7Mt0_Cw6pRfeTwY94oDRumSd-82v@mail.gmail.com>
Message-ID: <AANLkTi=2jadBQqYxyTqi-ppWu+n2HeWkkv8A6RA4j-5b@mail.gmail.com>

But that doesn't mean that license isn't required, does it?

I was more interested in a free to use codec comparable to G729 in
terms of both bandwidth and quality.

Regards
HASSAN


On 2010-10-21, M.Emran <monemran at gmail.com> wrote:
> in ARM processor, you can do g729.
>
> On Thu, Oct 21, 2010 at 1:30 PM, Nyamul Hassan <mnhassan at usa.net> wrote:
>
>> Any one know of a comparable low bandwidth codec that works for mobile
>> based softphones?
>>
>> Regards
>> HASSAN
>>
>>
>> On 2010-10-21, Thomas Mueller <thomas at chaschperli.ch> wrote:
>> >
>> >> Just curious. If a G729 CoDec is a MUST and unless you prefer to use a
>> SIP
>> >> compliant ATA device that already comes with a built-in G729 CoDec
>> > i've had good results with G.729 with audio quality and bandwith -
>> > especially with ADSL where uplink is the bottleneck. My hardware-sip
>> > phones (Snom, Siemens) are equiped with the g729 codec, the sip provider
>> > too.
>> >
>> > - Thomas
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> --
>> Sent from my mobile device
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Regards
> ----------
> M Emran
> Chief Executive Officer
> E-SOFT BILLING (pvt). LTD.
>
> HB Tower(3rd Floor)
> House No # 1/A
> Road No # 23
> Gulshan # 1
> Dhaka-1212, Bangladesh.
> *Phone:* +880-2-8822312,+880-2-8822384
> Fax : +880-2-8822254
> E-Mail: info at e-softbilling.com
> Web: www.e-softbilling.com
>          www.isoftswitch.com
>          www.howtonix.com
>          www.sipmobiledialer.com
>

-- 
Sent from my mobile device


From fraserredmond at gmail.com  Thu Oct 21 08:10:09 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Thu, 21 Oct 2010 16:10:09 +0100
Subject: [Freeswitch-users] Loss of first second of media
Message-ID: <AANLkTimECwtVmnbFsBJRiN3YSaGyXbvh-sKtKYPey8xx@mail.gmail.com>

I've been trying out a few new termination providers, and am finding that
the first second of media is being lost.

I get early media (ringing), then when the call is answered the first half a
second to a second is lost, then the rest of the call is fine.

The engineers at one of the providers ran a sip trace and a full packet
trace, and it looked like everything was being delivered from the pstn to my
server. They said that none of their other customers have reported this
problem before.

I've tested connecting to pstn's in several different countries, with the
same results. I've used a couple of other termination providers in the past
(1 still in current use) which have been fine, but now have 2 that are
losing the first part of the media.

So it seems like it's specific to Freeswitch, with some subset of conditions
for some providers.

This is running on Ubuntu on a Amazon AWS server, in case that helps.

Any ideas how to narrow it down further?

Cheers,
Fraser
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From steveayre at gmail.com  Thu Oct 21 08:17:08 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 21 Oct 2010 16:17:08 +0100
Subject: [Freeswitch-users] FS UK Ringback
In-Reply-To: <8450FE01FF743446A9E2FDCCDE3B323201975CFE4D17@MAIL-ROMEO.squay.com>
References: <8450FE01FF743446A9E2FDCCDE3B323201975CFE4D17@MAIL-ROMEO.squay.com>
Message-ID: <AANLkTimh=-W6Fdf-ugc+wXgOA+CNh3Hr3_uxLDS4mWZE@mail.gmail.com>

That's the correct UK ringback definition... Are you sure you're using
uk-ring not us-ring?

Does this sound ok?
<action application="playback"
data="tone_stream://%(400,200,400,450);%(400,2200,400,450)"/>

(If it does then you're not using uk-ring).

-Steve


On 20 October 2010 23:14, Magnus Kelly <m.kelly at mundio.com> wrote:
> Hello all,
>
> Would anyone be able to help ?with either a FS friendly audio file for UK ring back, or the definitions to create it natively as my current FS setup is not generating the correct UK tones.
>
> Currently its defined as below, however its sounding like stuttered US dial tone as opposed as normal UK ringback.
>
> <X-PRE-PROCESS cmd="set" data="uk-ring=%(400,200,400,450);%(400,2200,400,450)"/>
>
> Many thanks
> Magnus
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From steveayre at gmail.com  Thu Oct 21 08:20:02 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 21 Oct 2010 16:20:02 +0100
Subject: [Freeswitch-users] FS UK Ringback
In-Reply-To: <8450FE01FF743446A9E2FDCCDE3B323201975CFE4D17@MAIL-ROMEO.squay.com>
References: <8450FE01FF743446A9E2FDCCDE3B323201975CFE4D17@MAIL-ROMEO.squay.com>
Message-ID: <AANLkTi=DWBbgkuPm4iUnSNPm+7OX4UhNA1eTpTVQ8G+H@mail.gmail.com>

Are you setting something in dialplan like:
<action application="set" data="ringback=$${uk-ring}"/

Also, is it possible it's being overridden by your SIP client? Mine
(eyebeam) generates a US ringback tone until it actually starts
receiving early media, so for calls with a long PDD it's noticable
that there's a US, then UK ringback tone.

-Steve


On 20 October 2010 23:14, Magnus Kelly <m.kelly at mundio.com> wrote:
> Hello all,
>
> Would anyone be able to help ?with either a FS friendly audio file for UK ring back, or the definitions to create it natively as my current FS setup is not generating the correct UK tones.
>
> Currently its defined as below, however its sounding like stuttered US dial tone as opposed as normal UK ringback.
>
> <X-PRE-PROCESS cmd="set" data="uk-ring=%(400,200,400,450);%(400,2200,400,450)"/>
>
> Many thanks
> Magnus
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From anthony.minessale at gmail.com  Thu Oct 21 08:28:37 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 21 Oct 2010 10:28:37 -0500
Subject: [Freeswitch-users] Loss of first second of media
In-Reply-To: <AANLkTimECwtVmnbFsBJRiN3YSaGyXbvh-sKtKYPey8xx@mail.gmail.com>
References: <AANLkTimECwtVmnbFsBJRiN3YSaGyXbvh-sKtKYPey8xx@mail.gmail.com>
Message-ID: <AANLkTik5H74MupY8iyx6Ao4=Rv=wCm8jv-w=xCtSnm0Z@mail.gmail.com>

Where is the other side of this call coming from?

[ (   ) ] -> FS -> (PSTN via SIP)

What goes in the empty space above?



On Thu, Oct 21, 2010 at 10:10 AM, Fraser Redmond
<fraserredmond at gmail.com> wrote:
> I've been trying out a few new termination providers, and am finding that
> the first second of media is being lost.
>
> I get early media (ringing), then when the call is answered the first half a
> second to a second is lost, then the rest of the call is fine.
>
> The engineers at one of the providers ran a sip trace and a full packet
> trace, and it looked like everything was being delivered from the pstn to my
> server. They said that none of their other customers have reported this
> problem before.
>
> I've tested connecting to pstn's in several different countries, with the
> same results. I've used a couple of other termination providers in the past
> (1 still in current use) which have been fine, but now have 2 that are
> losing the first part of the media.
>
> So it seems like it's specific to Freeswitch, with some subset of conditions
> for some providers.
>
> This is running on Ubuntu on a Amazon AWS server, in case that helps.
>
> Any ideas how to narrow it down further?
>
> Cheers,
> Fraser
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From covici at ccs.covici.com  Thu Oct 21 08:32:38 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 21 Oct 2010 11:32:38 -0400
Subject: [Freeswitch-users] problem in default dialplan extension 869
Message-ID: <11887.1287675158@ccs.covici.com>

Maybe I am not understanding this, but if I look at the extension for
call return, it has the following:
<action application="transfer"
data="${hash(select/${domain_name}-call_return/${caller_id_number})}"/>
whereas the insert statement when the extension is called has
/${dialed_extension} after the call-return.  So what can I put in the
select to retrieve the correct number?

Thanks for any suggestion.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From fraserredmond at gmail.com  Thu Oct 21 09:12:34 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Thu, 21 Oct 2010 17:12:34 +0100
Subject: [Freeswitch-users] Loss of first second of media
In-Reply-To: <AANLkTik5H74MupY8iyx6Ao4=Rv=wCm8jv-w=xCtSnm0Z@mail.gmail.com>
References: <AANLkTimECwtVmnbFsBJRiN3YSaGyXbvh-sKtKYPey8xx@mail.gmail.com>
	<AANLkTik5H74MupY8iyx6Ao4=Rv=wCm8jv-w=xCtSnm0Z@mail.gmail.com>
Message-ID: <AANLkTikpVH-s6AvJ7fQoZu2PE4dduTg+d-ygyBWSfr4R@mail.gmail.com>

The call is originated from Freeswitch (via CLI) to a softphone, then when
that is connected it bridges out to the gateway.

Cheers,
Fraser




On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Where is the other side of this call coming from?
>
> [ (   ) ] -> FS -> (PSTN via SIP)
>
> What goes in the empty space above?
>
>
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From anthony.minessale at gmail.com  Thu Oct 21 09:35:41 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 21 Oct 2010 11:35:41 -0500
Subject: [Freeswitch-users] Loss of first second of media
In-Reply-To: <AANLkTikpVH-s6AvJ7fQoZu2PE4dduTg+d-ygyBWSfr4R@mail.gmail.com>
References: <AANLkTimECwtVmnbFsBJRiN3YSaGyXbvh-sKtKYPey8xx@mail.gmail.com>
	<AANLkTik5H74MupY8iyx6Ao4=Rv=wCm8jv-w=xCtSnm0Z@mail.gmail.com>
	<AANLkTikpVH-s6AvJ7fQoZu2PE4dduTg+d-ygyBWSfr4R@mail.gmail.com>
Message-ID: <AANLkTi=FZDqev9R1aTMpdvTk8t9NfxiyeisQSRBxfTEb@mail.gmail.com>

how are you accomplishing that? by which technique?

On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond
<fraserredmond at gmail.com> wrote:
> The call is originated from Freeswitch (via CLI) to a softphone, then when
> that is connected it bridges out to the gateway.
>
> Cheers,
> Fraser
>
>
>
>
> On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> Where is the other side of this call coming from?
>>
>> [ ( ? ) ] -> FS -> (PSTN via SIP)
>>
>> What goes in the empty space above?
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From covici at ccs.covici.com  Thu Oct 21 09:42:00 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 21 Oct 2010 12:42:00 -0400
Subject: [Freeswitch-users] Trouble to start fail2ban
In-Reply-To: <AANLkTi=bt58wFwnZTutdNwNTC6KmLJ=Wptt8VRVnmrcZ@mail.gmail.com>
References: <AANLkTi=5nZqN5yE=9Yk80aJuw5h3-UOf6dJTx_x46jrA@mail.gmail.com>
	<AANLkTi=bt58wFwnZTutdNwNTC6KmLJ=Wptt8VRVnmrcZ@mail.gmail.com>
Message-ID: <14388.1287679320@ccs.covici.com>

Your fail2ban jail is working fine, thank goodness it has not found
anything yet.  The only strange thing is that I get a mail message that
the jail stops every night when the cron.daily runs -- maybe the
logrotate is doing something funky.

Norman Tomlins <norm at voicenetwork.ca> wrote:

> Javier,
> 
> I had to make some changes to the example on the FreeSwitch wiki to get
> fail2ban working correctly.   I also have a step-by-step cut&paste guide on
> http://wiki.voicenetwork.ca/wiki/Main_Page#Fail2Ban for CentOS.
> 
> [freeswitch-tcp]
> enabled  = true
> port     = 5060,5061,5080,5081
> protocol = tcp
> filter   = freeswitch
> logpath  = /usr/local/freeswitch/log/freeswitch.log
> action   = iptables-allports[name=freeswitch-tcp, protocol=all]
>            sendmail-whois[name=FreeSwitch, dest=root,
> sender=fail2ban at example.org]
> 
> [freeswitch-udp]
> enabled  = true
> port     = 5060,5061,5080,5081
> protocol = udp
> filter   = freeswitch
> logpath  = /usr/local/freeswitch/log/freeswitch.log
> action   = iptables-allports[name=freeswitch-udp, protocol=all]
>            sendmail-whois[name=FreeSwitch, dest=root,
> sender=fail2ban at example.org]
> 
> 
> Norman Tomlins
> Voice Network Inc.
> http://www.VoiceNetwork.ca
> 
> 
> 2010/10/15 Javier Aristiz?bal <javieraristizabal at gmail.com>
> 
> > Hi folks,
> >
> > I installed fail2ban CentOS 5.5. And i followed the wiki instructions to
> > configure fail2ban with FreeSWITCH. After i edit jail.conf with this config:
> >
> > [freeswitch-tcp]
> >
> > enabled  = true
> > port     = 5060,5061,5080,5081
> > protocol = tcp
> > filter   = freeswitch
> > logpath  = /usr/local/freeswitch/log/freeswitch.log
> >
> > [freeswitch-udp]
> >
> > enabled  = true
> > port     = 5060,5061,5080,5081
> > protocol = udp
> > filter   = freeswitch
> > logpath  = /usr/local/freeswitch/log/freeswitch.log
> > ######
> >
> > The fail2ban does not start, and the logs can not get anything.. Did anyone
> > have the same thing?
> >
> > The freeswtich.conf:
> >
> > # Fail2Ban configuration file
> > #
> > # Author: Rupa SChomaker
> > #
> >
> > [Definition]
> >
> > # Option:  failregex
> > # Notes.:  regex to match the password failures messages in the logfile.
> > The
> > #          host must be matched by a group named "host". The tag "<HOST>"
> > can
> > #          be used for standard IP/hostname matching and is only an alias
> > for
> > #          (?:::f{4,6}:)?(?P<host>[\w\-.^_]+)
> > # Values:  TEXT
> > #
> > failregex = \[WARNING\] sofia_reg.c:\d+ SIP auth failure \(REGISTER\) on
> > sofia profile \'\w+\' for \[.*\] from ip <HOST>
> >
> > # Option:  ignoreregex
> > # Notes.:  regex to ignore. If this regex matches, the line is ignored.
> > # Values:  TEXT
> > #
> > ignoreregex =
> >
> >
> > Thanks
> >
> > --
> > Javier Aristiz?bal
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From fraserredmond at gmail.com  Thu Oct 21 10:00:17 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Thu, 21 Oct 2010 18:00:17 +0100
Subject: [Freeswitch-users] Loss of first second of media
In-Reply-To: <AANLkTi=FZDqev9R1aTMpdvTk8t9NfxiyeisQSRBxfTEb@mail.gmail.com>
References: <AANLkTimECwtVmnbFsBJRiN3YSaGyXbvh-sKtKYPey8xx@mail.gmail.com>
	<AANLkTik5H74MupY8iyx6Ao4=Rv=wCm8jv-w=xCtSnm0Z@mail.gmail.com>
	<AANLkTikpVH-s6AvJ7fQoZu2PE4dduTg+d-ygyBWSfr4R@mail.gmail.com>
	<AANLkTi=FZDqev9R1aTMpdvTk8t9NfxiyeisQSRBxfTEb@mail.gmail.com>
Message-ID: <AANLkTikY68ovuiSO9xkbZZPSskxCNsjrpTSdb-42bwJB@mail.gmail.com>

event_socket:
api originate {vars=values}user/$fromExtn at Domain'set:bLegVars=values,transfer:$toNum
xml outbound_call' inline

then

dialplan:
<action application="set" data="effective_caller_id_number=+1800number"/>
<action application="set" data="effective_caller_id_name="/>
(set and/or export a bunch of other vars too)
<action application="set" data="dial_string=sofia/gateway/
gatewayname.com/00${destination_number}"/>
<action application="bridge" data="${dial_string}"/>

Cheers,
Fraser




On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> how are you accomplishing that? by which technique?
>
> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond
> <fraserredmond at gmail.com> wrote:
> > The call is originated from Freeswitch (via CLI) to a softphone, then
> when
> > that is connected it bridges out to the gateway.
> >
> > Cheers,
> > Fraser
> >
> >
> >
> >
> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> Where is the other side of this call coming from?
> >>
> >> [ (   ) ] -> FS -> (PSTN via SIP)
> >>
> >> What goes in the empty space above?
>
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From anthony.minessale at gmail.com  Thu Oct 21 10:13:28 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 21 Oct 2010 12:13:28 -0500
Subject: [Freeswitch-users] Loss of first second of media
In-Reply-To: <AANLkTikY68ovuiSO9xkbZZPSskxCNsjrpTSdb-42bwJB@mail.gmail.com>
References: <AANLkTimECwtVmnbFsBJRiN3YSaGyXbvh-sKtKYPey8xx@mail.gmail.com>
	<AANLkTik5H74MupY8iyx6Ao4=Rv=wCm8jv-w=xCtSnm0Z@mail.gmail.com>
	<AANLkTikpVH-s6AvJ7fQoZu2PE4dduTg+d-ygyBWSfr4R@mail.gmail.com>
	<AANLkTi=FZDqev9R1aTMpdvTk8t9NfxiyeisQSRBxfTEb@mail.gmail.com>
	<AANLkTikY68ovuiSO9xkbZZPSskxCNsjrpTSdb-42bwJB@mail.gmail.com>
Message-ID: <AANLkTikiG3xPKDVkwU4GDk5t6KDe67AQmM-dkheMCSu7@mail.gmail.com>

are you setting ignore_early_media=true in the first vars=values area?

This looks like you could be calling one leg who is still not answered
and then bridging it to another dest.  The bridge app will wait for
the first leg to answer before bridging.

Also if you have any NAT anywhere, look for an "auto-changing port"
type message which can also be attributed to this due to a detection
period for incorrect ports.



On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond
<fraserredmond at gmail.com> wrote:
> event_socket:
> api originate {vars=values}user/$fromExtn at Domain
> 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline
>
> then
>
> dialplan:
> <action application="set" data="effective_caller_id_number=+1800number"/>
> <action application="set" data="effective_caller_id_name="/>
> (set and/or export a bunch of other vars too)
> <action application="set"
> data="dial_string=sofia/gateway/gatewayname.com/00${destination_number}"/>
> <action application="bridge" data="${dial_string}"/>
>
> Cheers,
> Fraser
>
>
>
>
> On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> how are you accomplishing that? by which technique?
>>
>> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond
>> <fraserredmond at gmail.com> wrote:
>> > The call is originated from Freeswitch (via CLI) to a softphone, then
>> > when
>> > that is connected it bridges out to the gateway.
>> >
>> > Cheers,
>> > Fraser
>> >
>> >
>> >
>> >
>> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> Where is the other side of this call coming from?
>> >>
>> >> [ ( ? ) ] -> FS -> (PSTN via SIP)
>> >>
>> >> What goes in the empty space above?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From fraserredmond at gmail.com  Thu Oct 21 10:32:20 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Thu, 21 Oct 2010 18:32:20 +0100
Subject: [Freeswitch-users] Loss of first second of media
In-Reply-To: <AANLkTikiG3xPKDVkwU4GDk5t6KDe67AQmM-dkheMCSu7@mail.gmail.com>
References: <AANLkTimECwtVmnbFsBJRiN3YSaGyXbvh-sKtKYPey8xx@mail.gmail.com>
	<AANLkTik5H74MupY8iyx6Ao4=Rv=wCm8jv-w=xCtSnm0Z@mail.gmail.com>
	<AANLkTikpVH-s6AvJ7fQoZu2PE4dduTg+d-ygyBWSfr4R@mail.gmail.com>
	<AANLkTi=FZDqev9R1aTMpdvTk8t9NfxiyeisQSRBxfTEb@mail.gmail.com>
	<AANLkTikY68ovuiSO9xkbZZPSskxCNsjrpTSdb-42bwJB@mail.gmail.com>
	<AANLkTikiG3xPKDVkwU4GDk5t6KDe67AQmM-dkheMCSu7@mail.gmail.com>
Message-ID: <AANLkTinJRcpRx_BQyKifECcurYb+vG2P8Nct1xT_X3XO@mail.gmail.com>

Sorry, yes, I am setting ignore_early_media=true in the first area. (Or are
you saying that should be off? I forget now why I needed it on, but there
was a reason I added it.)

Yes, the bridge doesn't start until after the A-leg has answered.

Thanks for the suggestion about nat/auto-changing port, I'll have a look
into that - would that be in the cli output or in a sip trace? I've already
looked and it's not appearing in the CLI output (with loglevel=debug),
haven't looked in the sip trace yet.

Cheers,
Fraser




On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> are you setting ignore_early_media=true in the first vars=values area?
>
> This looks like you could be calling one leg who is still not answered
> and then bridging it to another dest.  The bridge app will wait for
> the first leg to answer before bridging.
>
> Also if you have any NAT anywhere, look for an "auto-changing port"
> type message which can also be attributed to this due to a detection
> period for incorrect ports.
>
>
>
> On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond
> <fraserredmond at gmail.com> wrote:
> > event_socket:
> > api originate {vars=values}user/$fromExtn at Domain
> > 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline
> >
> > then
> >
> > dialplan:
> > <action application="set" data="effective_caller_id_number=+1800number"/>
> > <action application="set" data="effective_caller_id_name="/>
> > (set and/or export a bunch of other vars too)
> > <action application="set"
> > data="dial_string=sofia/gateway/gatewayname.com/00${destination_number}<http://gatewayname.com/00$%7Bdestination_number%7D>
> "/>
> > <action application="bridge" data="${dial_string}"/>
> >
> > Cheers,
> > Fraser
> >
> >
> >
> >
> > On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> how are you accomplishing that? by which technique?
> >>
> >> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond
> >> <fraserredmond at gmail.com> wrote:
> >> > The call is originated from Freeswitch (via CLI) to a softphone, then
> >> > when
> >> > that is connected it bridges out to the gateway.
> >> >
> >> > Cheers,
> >> > Fraser
> >> >
> >> >
> >> >
> >> >
> >> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> Where is the other side of this call coming from?
> >> >>
> >> >> [ (   ) ] -> FS -> (PSTN via SIP)
> >> >>
> >> >> What goes in the empty space above?
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Thu Oct 21 10:44:08 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 21 Oct 2010 12:44:08 -0500
Subject: [Freeswitch-users] Loss of first second of media
In-Reply-To: <AANLkTinJRcpRx_BQyKifECcurYb+vG2P8Nct1xT_X3XO@mail.gmail.com>
References: <AANLkTimECwtVmnbFsBJRiN3YSaGyXbvh-sKtKYPey8xx@mail.gmail.com>
	<AANLkTik5H74MupY8iyx6Ao4=Rv=wCm8jv-w=xCtSnm0Z@mail.gmail.com>
	<AANLkTikpVH-s6AvJ7fQoZu2PE4dduTg+d-ygyBWSfr4R@mail.gmail.com>
	<AANLkTi=FZDqev9R1aTMpdvTk8t9NfxiyeisQSRBxfTEb@mail.gmail.com>
	<AANLkTikY68ovuiSO9xkbZZPSskxCNsjrpTSdb-42bwJB@mail.gmail.com>
	<AANLkTikiG3xPKDVkwU4GDk5t6KDe67AQmM-dkheMCSu7@mail.gmail.com>
	<AANLkTinJRcpRx_BQyKifECcurYb+vG2P8Nct1xT_X3XO@mail.gmail.com>
Message-ID: <AANLkTinpaRivNHteDp4i6WJjmgsPAE6otVx5LjgJcvWr@mail.gmail.com>

its a blue message on cli

It could also be the other side expecting us to send media first or
something silly.
try getting a sip trace and figure out when the rtp starts arriving.


On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond
<fraserredmond at gmail.com> wrote:
> Sorry, yes, I am setting ignore_early_media=true in the first area. (Or are
> you saying that should be off? I forget now why I needed it on, but there
> was a reason I added it.)
>
> Yes, the bridge doesn't start until after the A-leg has answered.
>
> Thanks for the suggestion about nat/auto-changing port, I'll have a look
> into that - would that be in the cli output or in a sip trace? I've already
> looked and it's not appearing in the CLI output (with loglevel=debug),
> haven't looked in the sip trace yet.
>
> Cheers,
> Fraser
>
>
>
>
> On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> are you setting ignore_early_media=true in the first vars=values area?
>>
>> This looks like you could be calling one leg who is still not answered
>> and then bridging it to another dest. ?The bridge app will wait for
>> the first leg to answer before bridging.
>>
>> Also if you have any NAT anywhere, look for an "auto-changing port"
>> type message which can also be attributed to this due to a detection
>> period for incorrect ports.
>>
>>
>>
>> On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond
>> <fraserredmond at gmail.com> wrote:
>> > event_socket:
>> > api originate {vars=values}user/$fromExtn at Domain
>> > 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline
>> >
>> > then
>> >
>> > dialplan:
>> > <action application="set"
>> > data="effective_caller_id_number=+1800number"/>
>> > <action application="set" data="effective_caller_id_name="/>
>> > (set and/or export a bunch of other vars too)
>> > <action application="set"
>> >
>> > data="dial_string=sofia/gateway/gatewayname.com/00${destination_number}"/>
>> > <action application="bridge" data="${dial_string}"/>
>> >
>> > Cheers,
>> > Fraser
>> >
>> >
>> >
>> >
>> > On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> how are you accomplishing that? by which technique?
>> >>
>> >> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond
>> >> <fraserredmond at gmail.com> wrote:
>> >> > The call is originated from Freeswitch (via CLI) to a softphone, then
>> >> > when
>> >> > that is connected it bridges out to the gateway.
>> >> >
>> >> > Cheers,
>> >> > Fraser
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale
>> >> > <anthony.minessale at gmail.com> wrote:
>> >> >>
>> >> >> Where is the other side of this call coming from?
>> >> >>
>> >> >> [ ( ? ) ] -> FS -> (PSTN via SIP)
>> >> >>
>> >> >> What goes in the empty space above?
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From kris at kriskinc.com  Thu Oct 21 10:45:22 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 21 Oct 2010 13:45:22 -0400
Subject: [Freeswitch-users] Accessing fax variables/status after rxfax/txfax
Message-ID: <AANLkTimSPnc9qR3X+WxGaDKPKTDdLn_Y=MiS29x+L12G@mail.gmail.com>

Hello everyone,

  Looking through the code it appears that txfax/rxfax automatically
hangup channels when they're called and set a hangup cause that more
or less makes sense depending on how the fax went.  My question is -
can I have them not hangup the channel automatically so I can continue
execution somehow?  Alternatively, can I somehow continue execution
based on the hangup cause?  I'm thinking something like
api_hangup_hook that has access to the hangup/fax status.

Thanks!

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From mthakershi at gmail.com  Thu Oct 21 10:50:11 2010
From: mthakershi at gmail.com (Malay Thakershi)
Date: Thu, 21 Oct 2010 12:50:11 -0500
Subject: [Freeswitch-users] Simultaneous calls / sound / resources
Message-ID: <AANLkTinhsifthnJcJ+yjZSksCGq7zFCQn4pY5XRwVfUz@mail.gmail.com>

Hello all,

I have few questions about resources when there are multiple in / out calls
going on in FS.

1. I use Cepstral. If I have 20 calls at the same time, does playing media
(from FS) suffer performance issues due to Cepstral licensing. I just have 1
server Cepstral license.

2. There is no sound card on my server (Windows 2008 Standard). Does
functioning of FS media modules / Cepstral suffer because of lack of sound
card? If I install high quality sound device, is it going to make the voice
quality better?

3. How do I make sure FS process has enough resources (processor / memory)
when other taxing tasks are going on the server (e.g. backup)? I have
noticed voice getting broken when someone calls when backup is going on.

Thank you so much for guidance / help.
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From anthony.minessale at gmail.com  Thu Oct 21 11:05:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 21 Oct 2010 13:05:14 -0500
Subject: [Freeswitch-users] Simultaneous calls / sound / resources
In-Reply-To: <AANLkTinhsifthnJcJ+yjZSksCGq7zFCQn4pY5XRwVfUz@mail.gmail.com>
References: <AANLkTinhsifthnJcJ+yjZSksCGq7zFCQn4pY5XRwVfUz@mail.gmail.com>
Message-ID: <AANLkTikNWVjjidi_ZCt4nj5bUC=YoLTmrC5eYWYAWYYd@mail.gmail.com>

1) yes cepstral is no very scalable consider mrcp and an external server.
2) no sound card does not matter.
3) make sure it's a good machine, for real performance a 64 bit OS on
a multi-core 64 bit processor is recommended.  test the timing with
"time_test 1000" looking for as close to 1000 as a final answer as
possible. and "timer_test" looking for 20.00




On Thu, Oct 21, 2010 at 12:50 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
> Hello all,
> I have few questions about resources when there are multiple in / out calls
> going on in FS.
> 1. I use Cepstral. If I have 20 calls at the same time, does playing media
> (from FS) suffer performance issues due to Cepstral licensing. I just have 1
> server Cepstral license.
> 2. There is no sound card on my server (Windows 2008 Standard). Does
> functioning of FS media modules / Cepstral suffer because of lack of sound
> card? If I install high quality sound device, is it going to make the voice
> quality better?
> 3. How do I make sure FS process has enough resources (processor / memory)
> when other taxing tasks are going on the server (e.g. backup)? I have
> noticed voice getting broken when someone calls when backup is going on.
> Thank you so much for guidance / help.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From msc at freeswitch.org  Thu Oct 21 11:07:20 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Oct 2010 11:07:20 -0700
Subject: [Freeswitch-users] problem in default dialplan extension 869
In-Reply-To: <11887.1287675158@ccs.covici.com>
References: <11887.1287675158@ccs.covici.com>
Message-ID: <AANLkTikiUANLBuAKXcbHbCzjVgfPBZ4NYTsBA5LKbhx7@mail.gmail.com>

John,

I'm not sure what you're trying to do. This is a simple hash insertion and
retrieval. When the caller dials someone in the Local_Extension (1000 to
1019 by default) then there's an insert operation. For example, when x1002
calls x1001 this is what happens:
EXECUTE sofia/internal/1002 at 10.15.0.94hash(insert/10.15.0.94-call_return/1001/1002)

At extension 1001, I pick up and dial *69 or 869 and this dialplan action is
set to execute:
Dialplan: sofia/internal/1001 at 10.15.0.94 Action
transfer(${hash(select/${domain_name}-call_return/${caller_id_number})})

The actual execution looks like this:
EXECUTE sofia/internal/1001 at 10.15.0.94 transfer(1002)

So *69/869 simple goes to the hash and effectively says, "Give me the caller
id of the last person to call me." In my example, the "me" is 1001. The last
person to call "me" was 1002. The data is stored in the hash as
"domain-call_return/called_party/calling_party". When the called party dials
*69/869 the dialplan simple retrieves the value from the hash. Now, the
confusing part might be the fact that the called party who is returning the
call is actually the calling party when he/she dials *69/869, therefore the
dialplan must use ${caller_id_number} as the hash key.

I hope this helps.

-MC

On Thu, Oct 21, 2010 at 8:32 AM, <covici at ccs.covici.com> wrote:

> Maybe I am not understanding this, but if I look at the extension for
> call return, it has the following:
> <action application="transfer"
> data="${hash(select/${domain_name}-call_return/${caller_id_number})}"/>
> whereas the insert statement when the extension is called has
> /${dialed_extension} after the call-return.  So what can I put in the
> select to retrieve the correct number?
>
> Thanks for any suggestion.
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From covici at ccs.covici.com  Thu Oct 21 11:25:22 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 21 Oct 2010 14:25:22 -0400
Subject: [Freeswitch-users] Simultaneous calls / sound / resources
In-Reply-To: <AANLkTikNWVjjidi_ZCt4nj5bUC=YoLTmrC5eYWYAWYYd@mail.gmail.com>
References: <AANLkTinhsifthnJcJ+yjZSksCGq7zFCQn4pY5XRwVfUz@mail.gmail.com>
	<AANLkTikNWVjjidi_ZCt4nj5bUC=YoLTmrC5eYWYAWYYd@mail.gmail.com>
Message-ID: <15609.1287685522@ccs.covici.com>

Just wondering I did time_test 1000 and got 999 and timer_test and got
Avg: 20.003ms Total Time: 1000.765ms.  Is that decent for fs?

Anthony Minessale <anthony.minessale at gmail.com> wrote:

> 1) yes cepstral is no very scalable consider mrcp and an external server.
> 2) no sound card does not matter.
> 3) make sure it's a good machine, for real performance a 64 bit OS on
> a multi-core 64 bit processor is recommended.  test the timing with
> "time_test 1000" looking for as close to 1000 as a final answer as
> possible. and "timer_test" looking for 20.00
> 
> 
> 
> 
> On Thu, Oct 21, 2010 at 12:50 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
> > Hello all,
> > I have few questions about resources when there are multiple in / out calls
> > going on in FS.
> > 1. I use Cepstral. If I have 20 calls at the same time, does playing media
> > (from FS) suffer performance issues due to Cepstral licensing. I just have 1
> > server Cepstral license.
> > 2. There is no sound card on my server (Windows 2008 Standard). Does
> > functioning of FS media modules / Cepstral suffer because of lack of sound
> > card? If I install high quality sound device, is it going to make the voice
> > quality better?
> > 3. How do I make sure FS process has enough resources (processor / memory)
> > when other taxing tasks are going on the server (e.g. backup)? I have
> > noticed voice getting broken when someone calls when backup is going on.
> > Thank you so much for guidance / help.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From juanito1982 at gmail.com  Thu Oct 21 11:33:27 2010
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Thu, 21 Oct 2010 20:33:27 +0200
Subject: [Freeswitch-users] FreeTDM error
Message-ID: <AANLkTinqDsAVuV1AGZPuwhcKhNWLSM01pQP+TekzoHae@mail.gmail.com>

Hello!

   I am migrating one server from openzap to freetdm. When I load freetdm I
get:

----------------------
2010-10-21 20:18:58.924598 [ERR] ftdm_io.c:4271 Error loading
/usr/local/freeswitch/mod/ftmod_sangoma_isdn.so [libsng_isdn.so.0: cannot
open shared object file: No such file or directory]
2010-10-21 20:18:58.924598 [ERR] ftdm_io.c:4528 Failed to load module type:
sangoma_isdn
2010-10-21 20:18:58.924598 [ERR] mod_freetdm.c:2478 Error configuring
Sangoma ISDN FreeTDM span 1
----------------------

for each span. /usr/local/freeswitch/mod/ftmod_sangoma_isdn.so file exists
so I don't know why FS fails loading it. Any suggestion?

Regards
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From curriegrad2004 at gmail.com  Thu Oct 21 11:36:47 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Thu, 21 Oct 2010 11:36:47 -0700
Subject: [Freeswitch-users] G.729 & Windows
In-Reply-To: <AANLkTi=2jadBQqYxyTqi-ppWu+n2HeWkkv8A6RA4j-5b@mail.gmail.com>
References: <4CBEBE4F.8040107@chaschperli.ch>
	<1287582251418-5654913.post@n2.nabble.com>
	<4CBFDD73.1020709@chaschperli.ch>
	<AANLkTikCFkD_zstn_eO3uCB9D_H4pEed2ZL2pfcfAU8Z@mail.gmail.com>
	<AANLkTikUBrCYTRvm7Mt0_Cw6pRfeTwY94oDRumSd-82v@mail.gmail.com>
	<AANLkTi=2jadBQqYxyTqi-ppWu+n2HeWkkv8A6RA4j-5b@mail.gmail.com>
Message-ID: <AANLkTimVkcdcz_4WaZd1OazPSmPfOTwafEftk+_fETo9@mail.gmail.com>

Speex would be your best bet. Uses slightly more than what g.729 uses,
but it's completely free and open source. You can use GSM too, but
iirc, the patents with that codec should have been expired quite some
time ago.

On Thu, Oct 21, 2010 at 7:34 AM, Nyamul Hassan <mnhassan at usa.net> wrote:
> But that doesn't mean that license isn't required, does it?
>
> I was more interested in a free to use codec comparable to G729 in
> terms of both bandwidth and quality.
>
> Regards
> HASSAN
>
>
> On 2010-10-21, M.Emran <monemran at gmail.com> wrote:
>> in ARM processor, you can do g729.
>>
>> On Thu, Oct 21, 2010 at 1:30 PM, Nyamul Hassan <mnhassan at usa.net> wrote:
>>
>>> Any one know of a comparable low bandwidth codec that works for mobile
>>> based softphones?
>>>
>>> Regards
>>> HASSAN
>>>
>>>
>>> On 2010-10-21, Thomas Mueller <thomas at chaschperli.ch> wrote:
>>> >
>>> >> Just curious. If a G729 CoDec is a MUST and unless you prefer to use a
>>> SIP
>>> >> compliant ATA device that already comes with a built-in G729 CoDec
>>> > i've had good results with G.729 with audio quality and bandwith -
>>> > especially with ADSL where uplink is the bottleneck. My hardware-sip
>>> > phones (Snom, Siemens) are equiped with the g729 codec, the sip provider
>>> > too.
>>> >
>>> > - Thomas
>>> >
>>> >
>>> >
>>> >
>>> >
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>> --
>>> Sent from my mobile device
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Regards
>> ----------
>> M Emran
>> Chief Executive Officer
>> E-SOFT BILLING (pvt). LTD.
>>
>> HB Tower(3rd Floor)
>> House No # 1/A
>> Road No # 23
>> Gulshan # 1
>> Dhaka-1212, Bangladesh.
>> *Phone:* +880-2-8822312,+880-2-8822384
>> Fax : +880-2-8822254
>> E-Mail: info at e-softbilling.com
>> Web: www.e-softbilling.com
>> ? ? ? ? ?www.isoftswitch.com
>> ? ? ? ? ?www.howtonix.com
>> ? ? ? ? ?www.sipmobiledialer.com
>>
>
> --
> Sent from my mobile device
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From peter.olsson at visionutveckling.se  Thu Oct 21 11:41:50 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 21 Oct 2010 20:41:50 +0200
Subject: [Freeswitch-users] FreeTDM error
In-Reply-To: <AANLkTinqDsAVuV1AGZPuwhcKhNWLSM01pQP+TekzoHae@mail.gmail.com>
References: <AANLkTinqDsAVuV1AGZPuwhcKhNWLSM01pQP+TekzoHae@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57E5ECDC45@cooper>

You're missing some shared object files. Make sure to install the lib-sangoma stuff, according to their wiki.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Juan Antonio Iba?ez Santorum [juanito1982 at gmail.com]
Skickat: den 21 oktober 2010 20:33
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] FreeTDM error

Hello!

   I am migrating one server from openzap to freetdm. When I load freetdm I get:

----------------------
2010-10-21 20:18:58.924598 [ERR] ftdm_io.c:4271 Error loading /usr/local/freeswitch/mod/ftmod_sangoma_isdn.so [libsng_isdn.so.0: cannot open shared object file: No such file or directory]
2010-10-21 20:18:58.924598 [ERR] ftdm_io.c:4528 Failed to load module type: sangoma_isdn
2010-10-21 20:18:58.924598 [ERR] mod_freetdm.c:2478 Error configuring Sangoma ISDN FreeTDM span 1
----------------------

for each span. /usr/local/freeswitch/mod/ftmod_sangoma_isdn.so file exists so I don't know why FS fails loading it. Any suggestion?

Regards
!DSPAM:4cc0893432931434226835!


From Russell.Mosemann at cune.org  Thu Oct 21 11:43:40 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Thu, 21 Oct 2010 18:43:40 -0000
Subject: [Freeswitch-users] FreeTDM error
In-Reply-To: <AANLkTinqDsAVuV1AGZPuwhcKhNWLSM01pQP+TekzoHae@mail.gmail.com>
Message-ID: <20101021184340.B5C982DAB3C@cuneorg-email.cune.pri>

Juan Antonio Iba???ez Santorum <juanito1982 at gmail.com> said:

> /usr/local/freeswitch/mod/ftmod_sangoma_isdn.so file exists
> so I don't know why FS fails loading it. Any suggestion?

File/directory permission problems?

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From covici at ccs.covici.com  Thu Oct 21 11:46:43 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 21 Oct 2010 14:46:43 -0400
Subject: [Freeswitch-users] problem in default dialplan extension 869
In-Reply-To: <AANLkTikiUANLBuAKXcbHbCzjVgfPBZ4NYTsBA5LKbhx7@mail.gmail.com>
References: <11887.1287675158@ccs.covici.com>
	<AANLkTikiUANLBuAKXcbHbCzjVgfPBZ4NYTsBA5LKbhx7@mail.gmail.com>
Message-ID: <15831.1287686803@ccs.covici.com>

Thanks, that clarified things -- I will have to see why it has not
worked for me, I just get an out of order signal, but I will check logs
and see why this happened.

Michael Collins <msc at freeswitch.org> wrote:

> John,
> 
> I'm not sure what you're trying to do. This is a simple hash insertion and
> retrieval. When the caller dials someone in the Local_Extension (1000 to
> 1019 by default) then there's an insert operation. For example, when x1002
> calls x1001 this is what happens:
> EXECUTE sofia/internal/1002 at 10.15.0.94hash(insert/10.15.0.94-call_return/1001/1002)
> 
> At extension 1001, I pick up and dial *69 or 869 and this dialplan action is
> set to execute:
> Dialplan: sofia/internal/1001 at 10.15.0.94 Action
> transfer(${hash(select/${domain_name}-call_return/${caller_id_number})})
> 
> The actual execution looks like this:
> EXECUTE sofia/internal/1001 at 10.15.0.94 transfer(1002)
> 
> So *69/869 simple goes to the hash and effectively says, "Give me the caller
> id of the last person to call me." In my example, the "me" is 1001. The last
> person to call "me" was 1002. The data is stored in the hash as
> "domain-call_return/called_party/calling_party". When the called party dials
> *69/869 the dialplan simple retrieves the value from the hash. Now, the
> confusing part might be the fact that the called party who is returning the
> call is actually the calling party when he/she dials *69/869, therefore the
> dialplan must use ${caller_id_number} as the hash key.
> 
> I hope this helps.
> 
> -MC
> 
> On Thu, Oct 21, 2010 at 8:32 AM, <covici at ccs.covici.com> wrote:
> 
> > Maybe I am not understanding this, but if I look at the extension for
> > call return, it has the following:
> > <action application="transfer"
> > data="${hash(select/${domain_name}-call_return/${caller_id_number})}"/>
> > whereas the insert statement when the extension is called has
> > /${dialed_extension} after the call-return.  So what can I put in the
> > select to retrieve the correct number?
> >
> > Thanks for any suggestion.
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From Russell.Mosemann at cune.org  Thu Oct 21 11:51:58 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Thu, 21 Oct 2010 18:51:58 -0000
Subject: [Freeswitch-users] FreeTDM error
In-Reply-To: <AANLkTinqDsAVuV1AGZPuwhcKhNWLSM01pQP+TekzoHae@mail.gmail.com>
Message-ID: <20101021185159.08E8F390104@cuneorg-email.cune.pri>

> [libsng_isdn.so.0: cannot
> open shared object file: No such file or directory]

Peter is correct. I didn't see the reference to the missing library
above. You have to install the library separately.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From freeswitch-list at puzzled.xs4all.nl  Thu Oct 21 11:54:43 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Thu, 21 Oct 2010 20:54:43 +0200
Subject: [Freeswitch-users] FreeTDM error
In-Reply-To: <AANLkTinqDsAVuV1AGZPuwhcKhNWLSM01pQP+TekzoHae@mail.gmail.com>
References: <AANLkTinqDsAVuV1AGZPuwhcKhNWLSM01pQP+TekzoHae@mail.gmail.com>
Message-ID: <4CC08C73.3040700@puzzled.xs4all.nl>

On 10/21/2010 08:33 PM, Juan Antonio Iba?ez Santorum wrote:
> Hello!
>
>     I am migrating one server from openzap to freetdm. When I load
> freetdm I get:
>
> ----------------------
> 2010-10-21 20:18:58.924598 [ERR] ftdm_io.c:4271 Error loading
> /usr/local/freeswitch/mod/ftmod_sangoma_isdn.so [libsng_isdn.so.0:

                                                    ^^^^^^^^^^^^^^^^^

You need to install libsng_isdn. You can find it on the Sangoma website.

Regards,
Patrick


From juanito1982 at gmail.com  Thu Oct 21 11:58:48 2010
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Thu, 21 Oct 2010 20:58:48 +0200
Subject: [Freeswitch-users] FreeTDM error
In-Reply-To: <4CC08C73.3040700@puzzled.xs4all.nl>
References: <AANLkTinqDsAVuV1AGZPuwhcKhNWLSM01pQP+TekzoHae@mail.gmail.com>
	<4CC08C73.3040700@puzzled.xs4all.nl>
Message-ID: <AANLkTikhqS69L4hZE7ypmQd6D5VP+vr4pu9Z_rAtkoLp@mail.gmail.com>

libsng_isdn-1.2.0.x86_64 installed befor compiling...

2010/10/21 Patrick Lists <freeswitch-list at puzzled.xs4all.nl>

> On 10/21/2010 08:33 PM, Juan Antonio Iba?ez Santorum wrote:
> > Hello!
> >
> >     I am migrating one server from openzap to freetdm. When I load
> > freetdm I get:
> >
> > ----------------------
> > 2010-10-21 20:18:58.924598 [ERR] ftdm_io.c:4271 Error loading
> > /usr/local/freeswitch/mod/ftmod_sangoma_isdn.so [libsng_isdn.so.0:
>
>                                                     ^^^^^^^^^^^^^^^^^
>
> You need to install libsng_isdn. You can find it on the Sangoma website.
>
> Regards,
> Patrick
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Russell.Mosemann at cune.org  Thu Oct 21 12:09:49 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Thu, 21 Oct 2010 19:09:49 -0000
Subject: [Freeswitch-users] FreeTDM error
In-Reply-To: <AANLkTikhqS69L4hZE7ypmQd6D5VP+vr4pu9Z_rAtkoLp@mail.gmail.com>
Message-ID: <20101021190949.1BBDF3E51FD@cuneorg-email.cune.pri>

Juan Antonio Iba???ez Santorum <juanito1982 at gmail.com> said:

> libsng_isdn-1.2.0.x86_64 installed befor compiling...

Check all of the normal things. Are file/directory permissions OK? Does
the library show up in "ldconfig -v"?

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From mthakershi at gmail.com  Thu Oct 21 12:09:50 2010
From: mthakershi at gmail.com (Malay Thakershi)
Date: Thu, 21 Oct 2010 14:09:50 -0500
Subject: [Freeswitch-users] Simultaneous calls / sound / resources
In-Reply-To: <15609.1287685522@ccs.covici.com>
References: <AANLkTinhsifthnJcJ+yjZSksCGq7zFCQn4pY5XRwVfUz@mail.gmail.com>
	<AANLkTikNWVjjidi_ZCt4nj5bUC=YoLTmrC5eYWYAWYYd@mail.gmail.com>
	<15609.1287685522@ccs.covici.com>
Message-ID: <AANLkTimvUzoDB_2+swF=8YYeu=OU5S+r3_WVmNxtapQC@mail.gmail.com>

My output is given below.
time_test gives avg 1953???

While timer_test fails???

Any idea?
--------------------------------------------------------------
freeswitch at CHSERV2> time_test 1000
API CALL [time_test(1000)] output:
test 1 sleep 1000 1953
test 2 sleep 1000 1953
test 3 sleep 1000 1953
test 4 sleep 1000 1953
test 5 sleep 1000 1953
test 6 sleep 1000 1953
test 7 sleep 1000 1953
test 8 sleep 1000 1953
test 9 sleep 1000 1953
test 10 sleep 1000 1953
avg 1953
--------------------------------------
freeswitch at CHSERV2> timer_test
Unknown Command: timer_test


On Thu, Oct 21, 2010 at 1:25 PM, <covici at ccs.covici.com> wrote:

> Just wondering I did time_test 1000 and got 999 and timer_test and got
> Avg: 20.003ms Total Time: 1000.765ms.  Is that decent for fs?
>
> Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
> > 1) yes cepstral is no very scalable consider mrcp and an external server.
> > 2) no sound card does not matter.
> > 3) make sure it's a good machine, for real performance a 64 bit OS on
> > a multi-core 64 bit processor is recommended.  test the timing with
> > "time_test 1000" looking for as close to 1000 as a final answer as
> > possible. and "timer_test" looking for 20.00
> >
> >
> >
> >
> > On Thu, Oct 21, 2010 at 12:50 PM, Malay Thakershi <mthakershi at gmail.com>
> wrote:
> > > Hello all,
> > > I have few questions about resources when there are multiple in / out
> calls
> > > going on in FS.
> > > 1. I use Cepstral. If I have 20 calls at the same time, does playing
> media
> > > (from FS) suffer performance issues due to Cepstral licensing. I just
> have 1
> > > server Cepstral license.
> > > 2. There is no sound card on my server (Windows 2008 Standard). Does
> > > functioning of FS media modules / Cepstral suffer because of lack of
> sound
> > > card? If I install high quality sound device, is it going to make the
> voice
> > > quality better?
> > > 3. How do I make sure FS process has enough resources (processor /
> memory)
> > > when other taxing tasks are going on the server (e.g. backup)? I have
> > > noticed voice getting broken when someone calls when backup is going
> on.
> > > Thank you so much for guidance / help.
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > 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 <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From christian.knoblauch at astylos.de  Thu Oct 21 07:29:42 2010
From: christian.knoblauch at astylos.de (Christian Knoblauch)
Date: Thu, 21 Oct 2010 17:29:42 +0300
Subject: [Freeswitch-users] FreeSwitch and Cisco 7941 IP Phone ( INVITE
	port number )
In-Reply-To: <AANLkTik5BkOMcDvaQseJocsp2276_yAa31-PEkpK1kMG@mail.gmail.com>
References: <057bc2928b30a59af5a20d08c3fb5141@astylos.de>
	<AANLkTik5BkOMcDvaQseJocsp2276_yAa31-PEkpK1kMG@mail.gmail.com>
Message-ID: <001201cb712c$696dbbd0$3c493370$@astylos.de>

Hi,

 

After commenting-out the mentioned code block, the INVITE is send to port
5060 so that the call gets established.

 

But still:

 

After hanging up one of the 7941 (call was established between 2 of them),
the BYE message towards the other 7941 is still send to the higher port
number ( so the corresponding 7941 does not end the line )

 

Thanks and regards

Christian 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ognjen
Seslija
Sent: Wednesday, October 20, 2010 12:23 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FreeSwitch and Cisco 7941 IP Phone ( INVITE
port number )

 

Please see http://jira.freeswitch.org/browse/FS-2773 I reported.

The mentioned code is now commented in FS, so nat is no longer assumed.

Regards.

On Mon, Oct 18, 2010 at 5:15 PM, Christian Knoblauch
<christian.knoblauch at astylos.de> wrote:


Hello,
 
 
This is about FreeSwitch 1.0.6 (Windows Build) and Cisco 7941 IP Phone, and
the same issue was recently reported by another subscriber of this list
(Michael W. Lucas).
 
Anyhow, I attach traces so that maybe someone can look into it.
 
 
CONTEXT:
 
The 7941 successfull REGISTER in FreeSwitch, and is able to call another
softphone.
The softphone is not able to call the 7941, and this is because the INVITE
from FreeSwitch towards 7941 goes to the UDP port that the 7941 was using
during REGISTER (high port number, instead of 5060)
 
I compared this with traces for 3CX where all works fine. 3CX sends the
INVITE towards 7941 to UDP port 5060, and the 7941 seems to like this :-)
 
 
Please find attached the REGISTER / INVITE traces for booth 3CX and
FreeSwitch, and also the FreeSwitch console-output from"sofia profile
internal siptrace on"
 
 
Thanks for review !
 
Regards
Christian
 


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

 

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From renjian at gmail.com  Thu Oct 21 07:21:11 2010
From: renjian at gmail.com (Jian Ren)
Date: Thu, 21 Oct 2010 10:21:11 -0400
Subject: [Freeswitch-users] Help needed about building snd_dummy with the
	dummy.c in skypopen
Message-ID: <AANLkTikQ_vj3Ue1MYs7x1pVj8kiCgFbeKBTL=97PddX=@mail.gmail.com>

Hi,
I am trying to follow this wiki:
http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
to setup FS and Skypopen on Ubuntu 8.04(64bit server inside virualbox).

There is one step asks me to build snd_dummy:
http://wiki.freeswitch.org/wiki/Skypopen_Skype_Endpoint_and_Trunk#ALSA_and_the_custom_snd-dummy
The section is like below:
 Ubuntu, Debian

Note, Ubuntu may have alsa sound drivers installed at /lib/modules/<your
kernel version>/ubuntu/sound/..., you may need to remove the dir to allow
modprobe search from the default place: /lib/modules/<your kernel
verision>/kernel/sound/ (don't forget to rerun /sbin/depmod after removing
the old sound directory. Eg: /sbin/depmod -a 2.6.24-24-server).

./configure --with-redhat=no \
--with-cards=dummy,usb-audio,hda-intel,hrtimer,rtctimer \
--with-card-options=all
make && make install

After the first *make && make install*, copy
mod_skypopen/configs/alsa/dummy.c to alsa-driver-1.0.20sound/drivers/dummy.c
.

make && make install #again :)

The problem is after I did all of them, when I entered modprobe snd_dummy,
it always crashed and returned as "killed" on my terminal, I could see it
dumped out a huge block or data(numbers, etc) on the host but don't
understand the meaning. While if I used the original dummy.c inside
alsa-driver-1.0.20, it worked fine.
Besides, the default ubuntu installation doesn't include kernal dev and
source, so I did one more step(or it cannot build alsa).

Did anyone try the same and get it working?

Thanks!
Jian
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From victor.chukalovskiy at utoronto.ca  Thu Oct 21 08:35:31 2010
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Thu, 21 Oct 2010 11:35:31 -0400
Subject: [Freeswitch-users] FS does not honour sip-force-contact. Advice?
Message-ID: <4CC05DC3.2080109@utoronto.ca>

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From juanito1982 at gmail.com  Thu Oct 21 12:15:53 2010
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Thu, 21 Oct 2010 21:15:53 +0200
Subject: [Freeswitch-users] FreeTDM error
In-Reply-To: <20101021190949.1BBDF3E51FD@cuneorg-email.cune.pri>
References: <AANLkTikhqS69L4hZE7ypmQd6D5VP+vr4pu9Z_rAtkoLp@mail.gmail.com>
	<20101021190949.1BBDF3E51FD@cuneorg-email.cune.pri>
Message-ID: <AANLkTiknR3t7hRrS_Ldthawcaw1=r9zjUHuMyXD7VO66@mail.gmail.com>

FS runs as root so no file permissions problems...

---------------------------
[root at nocheydia03 bin]# ldconfig -v | grep isdn
ldconfig: Path `/usr/local/lib' given more than once
ldconfig: Path `/usr/local/lib' given more than once
ldconfig: Path `/usr/local/lib' given more than once
ldconfig: Path `/usr/local/lib' given more than once
        libsng_isdn.so.0 -> libsng_isdn.so
---------------------------

Regards

2010/10/21 <Russell.Mosemann at cune.org>

> Juan Antonio Iba?ez Santorum <juanito1982 at gmail.com> said:
>
> > libsng_isdn-1.2.0.x86_64 installed befor compiling...
>
> Check all of the normal things. Are file/directory permissions OK? Does
> the library show up in "ldconfig -v"?
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peter.olsson at visionutveckling.se  Thu Oct 21 12:19:52 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 21 Oct 2010 21:19:52 +0200
Subject: [Freeswitch-users] Simultaneous calls / sound / resources
In-Reply-To: <AANLkTimvUzoDB_2+swF=8YYeu=OU5S+r3_WVmNxtapQC@mail.gmail.com>
References: <AANLkTinhsifthnJcJ+yjZSksCGq7zFCQn4pY5XRwVfUz@mail.gmail.com>
	<AANLkTikNWVjjidi_ZCt4nj5bUC=YoLTmrC5eYWYAWYYd@mail.gmail.com>
	<15609.1287685522@ccs.covici.com>,
	<AANLkTimvUzoDB_2+swF=8YYeu=OU5S+r3_WVmNxtapQC@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57E5ECDC46@cooper>

time_test on 1953 _might_ be good enough, 2 ms might be the best resolution your OS/hardware can perform. timer_test should work though, are you using current git?

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Malay Thakershi [mthakershi at gmail.com]
Skickat: den 21 oktober 2010 21:09
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Simultaneous calls / sound / resources

My output is given below.
time_test gives avg 1953???

While timer_test fails???

Any idea?
--------------------------------------------------------------
freeswitch at CHSERV2> time_test 1000
API CALL [time_test(1000)] output:
test 1 sleep 1000 1953
test 2 sleep 1000 1953
test 3 sleep 1000 1953
test 4 sleep 1000 1953
test 5 sleep 1000 1953
test 6 sleep 1000 1953
test 7 sleep 1000 1953
test 8 sleep 1000 1953
test 9 sleep 1000 1953
test 10 sleep 1000 1953
avg 1953
--------------------------------------
freeswitch at CHSERV2> timer_test
Unknown Command: timer_test


On Thu, Oct 21, 2010 at 1:25 PM, <covici at ccs.covici.com<mailto:covici at ccs.covici.com>> wrote:
Just wondering I did time_test 1000 and got 999 and timer_test and got
Avg: 20.003ms Total Time: 1000.765ms.  Is that decent for fs?

Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>> wrote:

> 1) yes cepstral is no very scalable consider mrcp and an external server.
> 2) no sound card does not matter.
> 3) make sure it's a good machine, for real performance a 64 bit OS on
> a multi-core 64 bit processor is recommended.  test the timing with
> "time_test 1000" looking for as close to 1000 as a final answer as
> possible. and "timer_test" looking for 20.00
>
>
>
>
> On Thu, Oct 21, 2010 at 12:50 PM, Malay Thakershi <mthakershi at gmail.com<mailto:mthakershi at gmail.com>> wrote:
> > Hello all,
> > I have few questions about resources when there are multiple in / out calls
> > going on in FS.
> > 1. I use Cepstral. If I have 20 calls at the same time, does playing media
> > (from FS) suffer performance issues due to Cepstral licensing. I just have 1
> > server Cepstral license.
> > 2. There is no sound card on my server (Windows 2008 Standard). Does
> > functioning of FS media modules / Cepstral suffer because of lack of sound
> > card? If I install high quality sound device, is it going to make the voice
> > quality better?
> > 3. How do I make sure FS process has enough resources (processor / memory)
> > when other taxing tasks are going on the server (e.g. backup)? I have
> > noticed voice getting broken when someone calls when backup is going on.
> > Thank you so much for guidance / help.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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<mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

--
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

        John Covici
        covici at ccs.covici.com<mailto:covici at ccs.covici.com>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4cc0918b32935175821050!


From mthakershi at gmail.com  Thu Oct 21 12:33:45 2010
From: mthakershi at gmail.com (Malay Thakershi)
Date: Thu, 21 Oct 2010 14:33:45 -0500
Subject: [Freeswitch-users] Simultaneous calls / sound / resources
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57E5ECDC46@cooper>
References: <AANLkTinhsifthnJcJ+yjZSksCGq7zFCQn4pY5XRwVfUz@mail.gmail.com>
	<AANLkTikNWVjjidi_ZCt4nj5bUC=YoLTmrC5eYWYAWYYd@mail.gmail.com>
	<15609.1287685522@ccs.covici.com>
	<AANLkTimvUzoDB_2+swF=8YYeu=OU5S+r3_WVmNxtapQC@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57E5ECDC46@cooper>
Message-ID: <AANLkTi=g4FJHVW-018WxEvWTsipFYei=P3+kW6sZOQL+@mail.gmail.com>

What does that mean? 2 ms?

I am not using current git.. is there a way to update my current build /
files with current one.. without messing up working system?

On Thu, Oct 21, 2010 at 2:19 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> time_test on 1953 _might_ be good enough, 2 ms might be the best resolution
> your OS/hardware can perform. timer_test should work though, are you using
> current git?
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Malay Thakershi [
> mthakershi at gmail.com]
> Skickat: den 21 oktober 2010 21:09
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] Simultaneous calls / sound / resources
>
> My output is given below.
> time_test gives avg 1953???
>
> While timer_test fails???
>
> Any idea?
> --------------------------------------------------------------
> freeswitch at CHSERV2> time_test 1000
> API CALL [time_test(1000)] output:
> test 1 sleep 1000 1953
> test 2 sleep 1000 1953
> test 3 sleep 1000 1953
> test 4 sleep 1000 1953
> test 5 sleep 1000 1953
> test 6 sleep 1000 1953
> test 7 sleep 1000 1953
> test 8 sleep 1000 1953
> test 9 sleep 1000 1953
> test 10 sleep 1000 1953
> avg 1953
> --------------------------------------
> freeswitch at CHSERV2> timer_test
> Unknown Command: timer_test
>
>
> On Thu, Oct 21, 2010 at 1:25 PM, <covici at ccs.covici.com<mailto:
> covici at ccs.covici.com>> wrote:
> Just wondering I did time_test 1000 and got 999 and timer_test and got
> Avg: 20.003ms Total Time: 1000.765ms.  Is that decent for fs?
>
> Anthony Minessale <anthony.minessale at gmail.com<mailto:
> anthony.minessale at gmail.com>> wrote:
>
> > 1) yes cepstral is no very scalable consider mrcp and an external server.
> > 2) no sound card does not matter.
> > 3) make sure it's a good machine, for real performance a 64 bit OS on
> > a multi-core 64 bit processor is recommended.  test the timing with
> > "time_test 1000" looking for as close to 1000 as a final answer as
> > possible. and "timer_test" looking for 20.00
> >
> >
> >
> >
> > On Thu, Oct 21, 2010 at 12:50 PM, Malay Thakershi <mthakershi at gmail.com
> <mailto:mthakershi at gmail.com>> wrote:
> > > Hello all,
> > > I have few questions about resources when there are multiple in / out
> calls
> > > going on in FS.
> > > 1. I use Cepstral. If I have 20 calls at the same time, does playing
> media
> > > (from FS) suffer performance issues due to Cepstral licensing. I just
> have 1
> > > server Cepstral license.
> > > 2. There is no sound card on my server (Windows 2008 Standard). Does
> > > functioning of FS media modules / Cepstral suffer because of lack of
> sound
> > > card? If I install high quality sound device, is it going to make the
> voice
> > > quality better?
> > > 3. How do I make sure FS process has enough resources (processor /
> memory)
> > > when other taxing tasks are going on the server (e.g. backup)? I have
> > > noticed voice getting broken when someone calls when backup is going
> on.
> > > Thank you so much for guidance / help.
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > 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 <MSN%3Aanthony_minessale at hotmail.com>
> <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>        John Covici
>         covici at ccs.covici.com<mailto:covici at ccs.covici.com>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4cc0918b32935175821050!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Russell.Mosemann at cune.org  Thu Oct 21 12:34:55 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Thu, 21 Oct 2010 19:34:55 -0000
Subject: [Freeswitch-users] FreeTDM error
In-Reply-To: <AANLkTiknR3t7hRrS_Ldthawcaw1=r9zjUHuMyXD7VO66@mail.gmail.com>
Message-ID: <20101021193455.0F29C3F51BF@cuneorg-email.cune.pri>

Juan Antonio Iba???ez Santorum <juanito1982 at gmail.com> said:

> ldconfig: Path `/usr/local/lib' given more than once
>         libsng_isdn.so.0 -> libsng_isdn.so
> ---------------------------

What does "ldd ftmod_sangoma_isdn.so" say?

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From peter.olsson at visionutveckling.se  Thu Oct 21 12:43:29 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 21 Oct 2010 21:43:29 +0200
Subject: [Freeswitch-users] Simultaneous calls / sound / resources
In-Reply-To: <AANLkTi=g4FJHVW-018WxEvWTsipFYei=P3+kW6sZOQL+@mail.gmail.com>
References: <AANLkTinhsifthnJcJ+yjZSksCGq7zFCQn4pY5XRwVfUz@mail.gmail.com>
	<AANLkTikNWVjjidi_ZCt4nj5bUC=YoLTmrC5eYWYAWYYd@mail.gmail.com>
	<15609.1287685522@ccs.covici.com>
	<AANLkTimvUzoDB_2+swF=8YYeu=OU5S+r3_WVmNxtapQC@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57E5ECDC46@cooper>,
	<AANLkTi=g4FJHVW-018WxEvWTsipFYei=P3+kW6sZOQL+@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57E5ECDC47@cooper>

It means FS tries to sleep 1 ms, but the OS returns 2ms at it's best. This is not always a problem, as long as 20ms is 20ms (default packet time for common codecs). This is where timer_test is handy, to use. If you can't run timer_timer, it's probably a really old system you're using.
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Malay Thakershi [mthakershi at gmail.com]
Skickat: den 21 oktober 2010 21:33
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Simultaneous calls / sound / resources

What does that mean? 2 ms?

I am not using current git.. is there a way to update my current build / files with current one.. without messing up working system?

On Thu, Oct 21, 2010 at 2:19 PM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
time_test on 1953 _might_ be good enough, 2 ms might be the best resolution your OS/hardware can perform. timer_test should work though, are you using current git?

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] f&#246;r Malay Thakershi [mthakershi at gmail.com<mailto:mthakershi at gmail.com>]
Skickat: den 21 oktober 2010 21:09
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Simultaneous calls / sound / resources

My output is given below.
time_test gives avg 1953???

While timer_test fails???

Any idea?
--------------------------------------------------------------
freeswitch at CHSERV2> time_test 1000
API CALL [time_test(1000)] output:
test 1 sleep 1000 1953
test 2 sleep 1000 1953
test 3 sleep 1000 1953
test 4 sleep 1000 1953
test 5 sleep 1000 1953
test 6 sleep 1000 1953
test 7 sleep 1000 1953
test 8 sleep 1000 1953
test 9 sleep 1000 1953
test 10 sleep 1000 1953
avg 1953
--------------------------------------
freeswitch at CHSERV2> timer_test
Unknown Command: timer_test


On Thu, Oct 21, 2010 at 1:25 PM, <covici at ccs.covici.com<mailto:covici at ccs.covici.com><mailto:covici at ccs.covici.com<mailto:covici at ccs.covici.com>>> wrote:
Just wondering I did time_test 1000 and got 999 and timer_test and got
Avg: 20.003ms Total Time: 1000.765ms.  Is that decent for fs?

Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com><mailto:anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>>> wrote:

> 1) yes cepstral is no very scalable consider mrcp and an external server.
> 2) no sound card does not matter.
> 3) make sure it's a good machine, for real performance a 64 bit OS on
> a multi-core 64 bit processor is recommended.  test the timing with
> "time_test 1000" looking for as close to 1000 as a final answer as
> possible. and "timer_test" looking for 20.00
>
>
>
>
> On Thu, Oct 21, 2010 at 12:50 PM, Malay Thakershi <mthakershi at gmail.com<mailto:mthakershi at gmail.com><mailto:mthakershi at gmail.com<mailto:mthakershi at gmail.com>>> wrote:
> > Hello all,
> > I have few questions about resources when there are multiple in / out calls
> > going on in FS.
> > 1. I use Cepstral. If I have 20 calls at the same time, does playing media
> > (from FS) suffer performance issues due to Cepstral licensing. I just have 1
> > server Cepstral license.
> > 2. There is no sound card on my server (Windows 2008 Standard). Does
> > functioning of FS media modules / Cepstral suffer because of lack of sound
> > card? If I install high quality sound device, is it going to make the voice
> > quality better?
> > 3. How do I make sure FS process has enough resources (processor / memory)
> > when other taxing tasks are going on the server (e.g. backup)? I have
> > noticed voice getting broken when someone calls when backup is going on.
> > Thank you so much for guidance / help.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org><mailto:FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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<mailto:MSN%3Aanthony_minessale at hotmail.com><mailto:MSN%3Aanthony_minessale at hotmail.com<mailto:MSN%253Aanthony_minessale at hotmail.com>>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com><mailto:PAYPAL%3Aanthony.minessale at gmail.com<mailto:PAYPAL%253Aanthony.minessale at gmail.com>>
> IRC: irc.freenode.net<http://irc.freenode.net><http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org><mailto:sip%3A888 at conference.freeswitch.org<mailto:sip%253A888 at conference.freeswitch.org>>
> googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org><mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org><mailto:FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

--
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

       John Covici
       covici at ccs.covici.com<mailto:covici at ccs.covici.com><mailto:covici at ccs.covici.com<mailto:covici at ccs.covici.com>>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org><mailto:FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4cc097ab32934660745498!


From anthony.minessale at gmail.com  Thu Oct 21 12:56:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 21 Oct 2010 14:56:23 -0500
Subject: [Freeswitch-users] FS does not honour sip-force-contact. Advice?
In-Reply-To: <4CC05DC3.2080109@utoronto.ca>
References: <4CC05DC3.2080109@utoronto.ca>
Message-ID: <AANLkTikqL-+_S5+rcgMTvNbLkq_v23O5pjgZpPKoY0T9@mail.gmail.com>

disable session timers on both sides of the call, they are interfering
with the dialog once it's established.


On Thu, Oct 21, 2010 at 10:35 AM, Victor Chukalovskiy
<victor.chukalovskiy at utoronto.ca> wrote:
> Dear Gurus,
>
> We have a user behind NAT.
> Currently we make him work by using
> ????? <variable name="sip-force-contact"
> value="sip:1234 at 171.134.62.106:5060"/>
> It works since users IP is static and there is a port forwarding setup for
> port 5060 on the users side.
>
> The problem is that after 15 minutes of a call FreeSWITCH attempts re-INVITE
> (RFC 4028 I guess) and does not honour port specified above.
> Instead, it uses some 40,000-ish UDP destination port.? Obviously, this
> never reaches users phone and FS hangups with "expiry on timer" cause.
>
> Is the syntax of sip-force-contact above valid?
>
> Is it normal for re-INVITEs no to honour sip-force-contact?
>
> I'd try NDLB-tls-connectile-dysfunction but I need clarification:
> -will it be honoured by re-INVITEs?
> -will it default to port 5060 or should I specify the port?
>
> Thank you,
> Victor
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From anthony.minessale at gmail.com  Thu Oct 21 13:03:26 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 21 Oct 2010 15:03:26 -0500
Subject: [Freeswitch-users] Simultaneous calls / sound / resources
In-Reply-To: <AANLkTi=g4FJHVW-018WxEvWTsipFYei=P3+kW6sZOQL+@mail.gmail.com>
References: <AANLkTinhsifthnJcJ+yjZSksCGq7zFCQn4pY5XRwVfUz@mail.gmail.com>
	<AANLkTikNWVjjidi_ZCt4nj5bUC=YoLTmrC5eYWYAWYYd@mail.gmail.com>
	<15609.1287685522@ccs.covici.com>
	<AANLkTimvUzoDB_2+swF=8YYeu=OU5S+r3_WVmNxtapQC@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57E5ECDC46@cooper>
	<AANLkTi=g4FJHVW-018WxEvWTsipFYei=P3+kW6sZOQL+@mail.gmail.com>
Message-ID: <AANLkTi=Xq8e0zjeWvvMUR5UGa5Ob5hcKa3oPQ8p5aHNc@mail.gmail.com>

yes you can update, you really should if your code is so old.


On Thu, Oct 21, 2010 at 2:33 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
> What does that mean? 2 ms?
> I am not using current git.. is there a way to update my current build /
> files with current one.. without messing up working system?
>
> On Thu, Oct 21, 2010 at 2:19 PM, Peter Olsson
> <peter.olsson at visionutveckling.se> wrote:
>>
>> time_test on 1953 _might_ be good enough, 2 ms might be the best
>> resolution your OS/hardware can perform. timer_test should work though, are
>> you using current git?
>>
>> /Peter
>> ________________________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Malay Thakershi
>> [mthakershi at gmail.com]
>> Skickat: den 21 oktober 2010 21:09
>> Till: FreeSWITCH Users Help
>> ?mne: Re: [Freeswitch-users] Simultaneous calls / sound / resources
>>
>> My output is given below.
>> time_test gives avg 1953???
>>
>> While timer_test fails???
>>
>> Any idea?
>> --------------------------------------------------------------
>> freeswitch at CHSERV2> time_test 1000
>> API CALL [time_test(1000)] output:
>> test 1 sleep 1000 1953
>> test 2 sleep 1000 1953
>> test 3 sleep 1000 1953
>> test 4 sleep 1000 1953
>> test 5 sleep 1000 1953
>> test 6 sleep 1000 1953
>> test 7 sleep 1000 1953
>> test 8 sleep 1000 1953
>> test 9 sleep 1000 1953
>> test 10 sleep 1000 1953
>> avg 1953
>> --------------------------------------
>> freeswitch at CHSERV2> timer_test
>> Unknown Command: timer_test
>>
>>
>> On Thu, Oct 21, 2010 at 1:25 PM,
>> <covici at ccs.covici.com<mailto:covici at ccs.covici.com>> wrote:
>> Just wondering I did time_test 1000 and got 999 and timer_test and got
>> Avg: 20.003ms Total Time: 1000.765ms. ?Is that decent for fs?
>>
>> Anthony Minessale
>> <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>> wrote:
>>
>> > 1) yes cepstral is no very scalable consider mrcp and an external
>> > server.
>> > 2) no sound card does not matter.
>> > 3) make sure it's a good machine, for real performance a 64 bit OS on
>> > a multi-core 64 bit processor is recommended. ?test the timing with
>> > "time_test 1000" looking for as close to 1000 as a final answer as
>> > possible. and "timer_test" looking for 20.00
>> >
>> >
>> >
>> >
>> > On Thu, Oct 21, 2010 at 12:50 PM, Malay Thakershi
>> > <mthakershi at gmail.com<mailto:mthakershi at gmail.com>> wrote:
>> > > Hello all,
>> > > I have few questions about resources when there are multiple in / out
>> > > calls
>> > > going on in FS.
>> > > 1. I use Cepstral. If I have 20 calls at the same time, does playing
>> > > media
>> > > (from FS) suffer performance issues due to Cepstral licensing. I just
>> > > have 1
>> > > server Cepstral license.
>> > > 2. There is no sound card on my server (Windows 2008 Standard). Does
>> > > functioning of FS media modules / Cepstral suffer because of lack of
>> > > sound
>> > > card? If I install high quality sound device, is it going to make the
>> > > voice
>> > > quality better?
>> > > 3. How do I make sure FS process has enough resources (processor /
>> > > memory)
>> > > when other taxing tasks are going on the server (e.g. backup)? I have
>> > > noticed voice getting broken when someone calls when backup is going
>> > > on.
>> > > Thank you so much for guidance / help.
>> > >
>> > >
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > >
>> > > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >
>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > 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<mailto:MSN%3Aanthony_minessale at hotmail.com>
>> >
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>> > IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> >
>> > sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
>> >
>> > googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> >
>> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> --
>> Your life is like a penny. ?You're going to lose it. ?The question is:
>> How do
>> you spend it?
>>
>> ? ? ? ?John Covici
>> ? ? ? ?covici at ccs.covici.com<mailto:covici at ccs.covici.com>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> !DSPAM:4cc0918b32935175821050!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From gmaruzz at gmail.com  Thu Oct 21 13:45:10 2010
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 21 Oct 2010 22:45:10 +0200
Subject: [Freeswitch-users] Help needed about building snd_dummy with
 the dummy.c in skypopen
In-Reply-To: <AANLkTikQ_vj3Ue1MYs7x1pVj8kiCgFbeKBTL=97PddX=@mail.gmail.com>
References: <AANLkTikQ_vj3Ue1MYs7x1pVj8kiCgFbeKBTL=97PddX=@mail.gmail.com>
Message-ID: <AANLkTik5BEvEi=P_+cEL2kwjfrTQYLGh7MNtF14-8ROE@mail.gmail.com>

Maybe is a problem with virtualbox. Also, is known that
fs+mod_skypopen+skype clients does not works in virtualbox.

Try it on a real (hardware) machine.

Or (but is less popular) in a xen like virtual machine.

-giovanni

On Thu, Oct 21, 2010 at 4:21 PM, Jian Ren <renjian at gmail.com> wrote:
> Hi,
> I am trying to follow this wiki:
> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
> to setup FS and Skypopen on Ubuntu 8.04(64bit server inside virualbox).
>
> There is one step asks me to build snd_dummy:
> http://wiki.freeswitch.org/wiki/Skypopen_Skype_Endpoint_and_Trunk#ALSA_and_the_custom_snd-dummy
> The section is like below:
>
> Ubuntu, Debian
>
> Note, Ubuntu may have alsa sound drivers installed at /lib/modules/<your
> kernel version>/ubuntu/sound/..., you may need to remove the dir to allow
> modprobe search from the default place: /lib/modules/<your kernel
> verision>/kernel/sound/ (don't forget to rerun /sbin/depmod after removing
> the old sound directory. Eg: /sbin/depmod -a 2.6.24-24-server).
>
> ./configure --with-redhat=no \
> --with-cards=dummy,usb-audio,hda-intel,hrtimer,rtctimer \
> --with-card-options=all
> make && make install
>
> After the first make && make install, copy mod_skypopen/configs/alsa/dummy.c
> to alsa-driver-1.0.20sound/drivers/dummy.c .
>
> make && make install #again?:)
>
> The problem is after I did all of them, when I entered modprobe snd_dummy,
> it always crashed and returned as "killed" on my terminal, I could see it
> dumped out a huge block or data(numbers, etc) on the host but don't
> understand the meaning. While if I used the original dummy.c inside
> alsa-driver-1.0.20, it worked fine.
> Besides, the default ubuntu installation doesn't include kernal dev and
> source, so I did one more step(or it cannot build alsa).
>
> Did anyone try the same and get it working?
>
> Thanks!
> Jian
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From juanito1982 at gmail.com  Thu Oct 21 13:51:19 2010
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Thu, 21 Oct 2010 22:51:19 +0200
Subject: [Freeswitch-users] FreeTDM error
In-Reply-To: <20101021193455.0F29C3F51BF@cuneorg-email.cune.pri>
References: <AANLkTiknR3t7hRrS_Ldthawcaw1=r9zjUHuMyXD7VO66@mail.gmail.com>
	<20101021193455.0F29C3F51BF@cuneorg-email.cune.pri>
Message-ID: <AANLkTikpTQR_ODsb=jXuNDf2LtJojDEMZON_vnGYac6C@mail.gmail.com>

------------------
[root at nocheydia03 mod]# ldd ftmod_sangoma_isdn.so
        libsng_isdn.so.0 => /usr/lib64/libsng_isdn.so.0 (0x00002b3c641d3000)
        libfreetdm.so.1 => /usr/local/freeswitch/lib/libfreetdm.so.1
(0x00002b3c645d2000)
        libm.so.6 => /lib64/libm.so.6 (0x00002b3c647f5000)
        libpthread.so.0 => /lib64/libpthread.so.0 (0x00002b3c64a79000)
        libdl.so.2 => /lib64/libdl.so.2 (0x00002b3c64c94000)
        libc.so.6 => /lib64/libc.so.6 (0x00002b3c64e98000)
        libsangoma.so.3 => /usr/lib/libsangoma.so.3 (0x00002b3c651f0000)
        librt.so.1 => /lib64/librt.so.1 (0x00002b3c653f9000)
        /lib64/ld-linux-x86-64.so.2 (0x00000033c6e00000)
------------------

2010/10/21 <Russell.Mosemann at cune.org>

> Juan Antonio Iba?ez Santorum <juanito1982 at gmail.com> said:
>
> > ldconfig: Path `/usr/local/lib' given more than once
> >         libsng_isdn.so.0 -> libsng_isdn.so
> > ---------------------------
>
> What does "ldd ftmod_sangoma_isdn.so" say?
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
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From Russell.Mosemann at cune.org  Thu Oct 21 14:06:59 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Thu, 21 Oct 2010 21:06:59 -0000
Subject: [Freeswitch-users] FreeTDM error
In-Reply-To: <AANLkTikpTQR_ODsb=jXuNDf2LtJojDEMZON_vnGYac6C@mail.gmail.com>
Message-ID: <20101021210700.097723F4779@cuneorg-email.cune.pri>

Juan Antonio Iba???ez Santorum <juanito1982 at gmail.com> said:

> ------------------
> [root at nocheydia03 mod]# ldd ftmod_sangoma_isdn.so
>         libsng_isdn.so.0 => /usr/lib64/libsng_isdn.so.0 

Keep going. You have to track down the problem. Does
/usr/lib64/libsng_isdn.so.0 exist? Does it point to a file that exists in
the same directory? You showed that there was some kind of isdn library
from ldconfig, but you didn't show that the library was listed in
/usr/lib64 when running ldconfig. Are you running on a 64-bit OS of some
kind?

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From mthakershi at gmail.com  Thu Oct 21 15:17:29 2010
From: mthakershi at gmail.com (Malay Thakershi)
Date: Thu, 21 Oct 2010 17:17:29 -0500
Subject: [Freeswitch-users] Simultaneous calls / sound / resources
In-Reply-To: <AANLkTi=Xq8e0zjeWvvMUR5UGa5Ob5hcKa3oPQ8p5aHNc@mail.gmail.com>
References: <AANLkTinhsifthnJcJ+yjZSksCGq7zFCQn4pY5XRwVfUz@mail.gmail.com>
	<AANLkTikNWVjjidi_ZCt4nj5bUC=YoLTmrC5eYWYAWYYd@mail.gmail.com>
	<15609.1287685522@ccs.covici.com>
	<AANLkTimvUzoDB_2+swF=8YYeu=OU5S+r3_WVmNxtapQC@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57E5ECDC46@cooper>
	<AANLkTi=g4FJHVW-018WxEvWTsipFYei=P3+kW6sZOQL+@mail.gmail.com>
	<AANLkTi=Xq8e0zjeWvvMUR5UGa5Ob5hcKa3oPQ8p5aHNc@mail.gmail.com>
Message-ID: <AANLkTimP7E8+kdcq0jkZrRm8zyP1HOJqY7aFkMCQOeR-@mail.gmail.com>

I am really afraid.. if I update the system.. my current settings won't
work. Is there a safe way to go from one version to other? Any method?

On Thu, Oct 21, 2010 at 3:03 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> yes you can update, you really should if your code is so old.
>
>
> On Thu, Oct 21, 2010 at 2:33 PM, Malay Thakershi <mthakershi at gmail.com>
> wrote:
> > What does that mean? 2 ms?
> > I am not using current git.. is there a way to update my current build /
> > files with current one.. without messing up working system?
> >
> > On Thu, Oct 21, 2010 at 2:19 PM, Peter Olsson
> > <peter.olsson at visionutveckling.se> wrote:
> >>
> >> time_test on 1953 _might_ be good enough, 2 ms might be the best
> >> resolution your OS/hardware can perform. timer_test should work though,
> are
> >> you using current git?
> >>
> >> /Peter
> >> ________________________________________
> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> >> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Malay
> Thakershi
> >> [mthakershi at gmail.com]
> >> Skickat: den 21 oktober 2010 21:09
> >> Till: FreeSWITCH Users Help
> >> ?mne: Re: [Freeswitch-users] Simultaneous calls / sound / resources
> >>
> >> My output is given below.
> >> time_test gives avg 1953???
> >>
> >> While timer_test fails???
> >>
> >> Any idea?
> >> --------------------------------------------------------------
> >> freeswitch at CHSERV2> time_test 1000
> >> API CALL [time_test(1000)] output:
> >> test 1 sleep 1000 1953
> >> test 2 sleep 1000 1953
> >> test 3 sleep 1000 1953
> >> test 4 sleep 1000 1953
> >> test 5 sleep 1000 1953
> >> test 6 sleep 1000 1953
> >> test 7 sleep 1000 1953
> >> test 8 sleep 1000 1953
> >> test 9 sleep 1000 1953
> >> test 10 sleep 1000 1953
> >> avg 1953
> >> --------------------------------------
> >> freeswitch at CHSERV2> timer_test
> >> Unknown Command: timer_test
> >>
> >>
> >> On Thu, Oct 21, 2010 at 1:25 PM,
> >> <covici at ccs.covici.com<mailto:covici at ccs.covici.com>> wrote:
> >> Just wondering I did time_test 1000 and got 999 and timer_test and got
> >> Avg: 20.003ms Total Time: 1000.765ms.  Is that decent for fs?
> >>
> >> Anthony Minessale
> >> <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>>
> wrote:
> >>
> >> > 1) yes cepstral is no very scalable consider mrcp and an external
> >> > server.
> >> > 2) no sound card does not matter.
> >> > 3) make sure it's a good machine, for real performance a 64 bit OS on
> >> > a multi-core 64 bit processor is recommended.  test the timing with
> >> > "time_test 1000" looking for as close to 1000 as a final answer as
> >> > possible. and "timer_test" looking for 20.00
> >> >
> >> >
> >> >
> >> >
> >> > On Thu, Oct 21, 2010 at 12:50 PM, Malay Thakershi
> >> > <mthakershi at gmail.com<mailto:mthakershi at gmail.com>> wrote:
> >> > > Hello all,
> >> > > I have few questions about resources when there are multiple in /
> out
> >> > > calls
> >> > > going on in FS.
> >> > > 1. I use Cepstral. If I have 20 calls at the same time, does playing
> >> > > media
> >> > > (from FS) suffer performance issues due to Cepstral licensing. I
> just
> >> > > have 1
> >> > > server Cepstral license.
> >> > > 2. There is no sound card on my server (Windows 2008 Standard). Does
> >> > > functioning of FS media modules / Cepstral suffer because of lack of
> >> > > sound
> >> > > card? If I install high quality sound device, is it going to make
> the
> >> > > voice
> >> > > quality better?
> >> > > 3. How do I make sure FS process has enough resources (processor /
> >> > > memory)
> >> > > when other taxing tasks are going on the server (e.g. backup)? I
> have
> >> > > noticed voice getting broken when someone calls when backup is going
> >> > > on.
> >> > > Thank you so much for guidance / help.
> >> > >
> >> > >
> >> > > _______________________________________________
> >> > > FreeSWITCH-users mailing list
> >> > >
> >> > > FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > >
> >> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > > 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<MSN%3Aanthony_minessale at hotmail.com>
> <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> >> >
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> >> > IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> >
> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> >> >
> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> >
> >> > FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >>
> >> --
> >> Your life is like a penny.  You're going to lose it.  The question is:
> >> How do
> >> you spend it?
> >>
> >>        John Covici
> >>        covici at ccs.covici.com<mailto:covici at ccs.covici.com>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >>
> >> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >> !DSPAM:4cc0918b32935175821050!
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bwibowo at gmail.com  Thu Oct 21 16:28:19 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Fri, 22 Oct 2010 06:28:19 +0700
Subject: [Freeswitch-users] mod_dingaling node error
Message-ID: <AANLkTikYuDcOHoXay+pexueun1ruZg2HPEBT2DXo_KLk@mail.gmail.com>

dear all
i try to install mod_dingaling following the wiki guide.
client.xml
<profile type="client">
   <param name="name" value="gmail.com"/>
   <param name="login" value="xxx at gmail.com/gtalk"/>
   <param name="password" value="xxxpass"/>
   <param name="server" value="talk.google.com"/>
   <param name="dialplan" value="XML"/>
   <param name="message" value="FreeSwitch (The Future of Telephony)"/>
   <param name="rtp-ip" value="auto"/>
   <!-- if you are behind NAT, you should probably use stun -->
   <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
   <param name="auto-login" value="true"/>
   <param name="auto-reply" value="Press *Call* to join my conference"/>
   <param name="sasl" value="plain"/>
   <param name="tls" value="true"/>
   <param name="use-rtp-timer" value="true"/>
   <param name="exten" value="1000"/>
 </profile>

in dialplan i add
 <extension name="gmail.com">
      <condition field="destination_number" expression="^50(1\d{10})$">
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="bridge" data="dingaling/
gmail.com/+$1 at voice.google.com"/>
       </condition>
   </extension>
</include>

after i load mod_dingaling i got this error

2010-10-22 07:21:45.483898 [ERR] libdingaling.c:1205 NODE ERROR!

2010-10-22 07:21:45.720897 [ERR] libdingaling.c:1205 NODE ERROR!

2010-10-22 07:21:45.956898 [ERR] libdingaling.c:1205 NODE ERROR!

2010-10-22 07:21:47.427901 [ERR] libdingaling.c:1205 NODE ERROR!

2010-10-22 07:21:50.147899 [ERR] libdingaling.c:1205 NODE ERROR!

2010-10-22 07:21:52.861897 [ERR] libdingaling.c:1205 NODE ERROR!

2010-10-22 07:21:53.097897 [ERR] libdingaling.c:1205 NODE ERROR!

2010-10-22 07:21:53.333896 [ERR] libdingaling.c:1205 NODE ERROR!

2010-10-22 07:21:53.560897 [ERR] libdingaling.c:1205 NODE ERROR!

have been trying to google but no luck related with that error.

any help is appreciated, and i use the git version



regards

budi
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From msc at freeswitch.org  Thu Oct 21 17:57:59 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Oct 2010 17:57:59 -0700
Subject: [Freeswitch-users] Accessing fax variables/status after
	rxfax/txfax
In-Reply-To: <AANLkTimSPnc9qR3X+WxGaDKPKTDdLn_Y=MiS29x+L12G@mail.gmail.com>
References: <AANLkTimSPnc9qR3X+WxGaDKPKTDdLn_Y=MiS29x+L12G@mail.gmail.com>
Message-ID: <AANLkTi=vXgXV7E4GAKEPUTYEEqYaCTJPks8=SOsikoaB@mail.gmail.com>

On Thu, Oct 21, 2010 at 10:45 AM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> Hello everyone,
>
>  Looking through the code it appears that txfax/rxfax automatically
> hangup channels when they're called and set a hangup cause that more
> or less makes sense depending on how the fax went.  My question is -
> can I have them not hangup the channel automatically so I can continue
> execution somehow?  Alternatively, can I somehow continue execution
> based on the hangup cause?  I'm thinking something like
> api_hangup_hook that has access to the hangup/fax status.
>
Kristian,

I think you need this:
http://wiki.freeswitch.org/wiki/Channel_Variables#session_in_hangup_hook

It will let the call hang up but a copy of the session will be available to
your hangup hook so you can access all the channel variables even after the
channel has actually hung up.
-MC
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From msc at freeswitch.org  Thu Oct 21 18:06:44 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Oct 2010 18:06:44 -0700
Subject: [Freeswitch-users] Simultaneous calls / sound / resources
In-Reply-To: <AANLkTimP7E8+kdcq0jkZrRm8zyP1HOJqY7aFkMCQOeR-@mail.gmail.com>
References: <AANLkTinhsifthnJcJ+yjZSksCGq7zFCQn4pY5XRwVfUz@mail.gmail.com>
	<AANLkTikNWVjjidi_ZCt4nj5bUC=YoLTmrC5eYWYAWYYd@mail.gmail.com>
	<15609.1287685522@ccs.covici.com>
	<AANLkTimvUzoDB_2+swF=8YYeu=OU5S+r3_WVmNxtapQC@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57E5ECDC46@cooper>
	<AANLkTi=g4FJHVW-018WxEvWTsipFYei=P3+kW6sZOQL+@mail.gmail.com>
	<AANLkTi=Xq8e0zjeWvvMUR5UGa5Ob5hcKa3oPQ8p5aHNc@mail.gmail.com>
	<AANLkTimP7E8+kdcq0jkZrRm8zyP1HOJqY7aFkMCQOeR-@mail.gmail.com>
Message-ID: <AANLkTimKRn6E3h6rvK24veAy6e0RtBCcWVypn8nopMbf@mail.gmail.com>

On Thu, Oct 21, 2010 at 3:17 PM, Malay Thakershi <mthakershi at gmail.com>wrote:

> I am really afraid.. if I update the system.. my current settings won't
> work. Is there a safe way to go from one version to other? Any method?
>
> If you are in production then you really should have a second machine
explicitly for testing. If you don't have a second machine then updating can
be an adventure. However, if you completely backup your
/usr/local/freeswitch directory and also your /usr/src/freeswitch directory
(whatever you named it) then you will have a snapshot of exactly what you
are running. You could restore those if the update goes bad for any reason.
(WARNING: that is a risky thing to do with a production system!)

The trick to updating and old system is to install fresh (preferably on your
test system) and then integrate your custom changes to the default configs.
Run FS, look for errors, fix 'em, repeat. I will never recommend you do this
on a production server, but some folks like to live dangerously. If you
don't have a test system then you have 3 choices: get a test server and
experiment on it, do your experiments on your production server (being VERY
careful to backup everything first), or live with the old version.

I think you'll agree that having a test system is the best option available.
-MC
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From pjintheusa at gmail.com  Thu Oct 21 18:23:58 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 21 Oct 2010 21:23:58 -0400
Subject: [Freeswitch-users] Using the bridge | operator for Implementing
	Failover
Message-ID: <AANLkTimZonOAnpKJKyy24ZnyJD9HnDPYMju40MYL6CvH@mail.gmail.com>

Hi there,
>From the WIKI

Implementing Failover

Failover for your outbound gateway is easy to implement at bridge time using
the | separator:

<action application="bridge"
data="sofia/gateway/primary/dialstring|sofia/gateway/secondary/dialstring"/>


In the above example though, if the user does not pick the call, that bridge
will try again using the secondary gateway, which is not desired. The second
attempt should only be made in there is a problem with the gateway.

How do you control that? .


Thanks


Pj
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From anthony.minessale at gmail.com  Thu Oct 21 20:55:21 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 21 Oct 2010 22:55:21 -0500
Subject: [Freeswitch-users] Using the bridge | operator for Implementing
	Failover
In-Reply-To: <AANLkTimZonOAnpKJKyy24ZnyJD9HnDPYMju40MYL6CvH@mail.gmail.com>
References: <AANLkTimZonOAnpKJKyy24ZnyJD9HnDPYMju40MYL6CvH@mail.gmail.com>
Message-ID: <AANLkTimoE7t0LQwK3Nmv8Km-KSHHAtk+n-Teeiv7zhoQ@mail.gmail.com>

put this before the bridge call

<!-- add as many , sep cause codes which qualify to stop trying the
pipe sep list -->
<action application="set" data="fail_on_single_reject=no_answer"/>


On Thu, Oct 21, 2010 at 8:23 PM, Phillip Jones <pjintheusa at gmail.com> wrote:
> Hi there,
> From the WIKI
> Implementing Failover
>
> Failover for your outbound gateway is easy to implement at bridge time using
> the | separator:
>
> <action application="bridge"
> data="sofia/gateway/primary/dialstring|sofia/gateway/secondary/dialstring"/>
>
>
> In the above example though, if the user does not pick the call, that bridge
> will try again using?the secondary gateway, which is not desired. The second
> attempt should only be made in there is a problem with the gateway.
>
> How do you control that? .
>
>
> Thanks
>
>
> Pj
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From anthony.minessale at gmail.com  Thu Oct 21 20:57:42 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 21 Oct 2010 22:57:42 -0500
Subject: [Freeswitch-users] Accessing fax variables/status after
	rxfax/txfax
In-Reply-To: <AANLkTi=vXgXV7E4GAKEPUTYEEqYaCTJPks8=SOsikoaB@mail.gmail.com>
References: <AANLkTimSPnc9qR3X+WxGaDKPKTDdLn_Y=MiS29x+L12G@mail.gmail.com>
	<AANLkTi=vXgXV7E4GAKEPUTYEEqYaCTJPks8=SOsikoaB@mail.gmail.com>
Message-ID: <AANLkTi=DUnXPcFHLJfmAawfzahdM4S_A=3muy-me7wys@mail.gmail.com>

Dang, that's a big page! Did I really make that many vars =0


On Thu, Oct 21, 2010 at 7:57 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Thu, Oct 21, 2010 at 10:45 AM, Kristian Kielhofner <kris at kriskinc.com>
> wrote:
>>
>> Hello everyone,
>>
>> ?Looking through the code it appears that txfax/rxfax automatically
>> hangup channels when they're called and set a hangup cause that more
>> or less makes sense depending on how the fax went. ?My question is -
>> can I have them not hangup the channel automatically so I can continue
>> execution somehow? ?Alternatively, can I somehow continue execution
>> based on the hangup cause? ?I'm thinking something like
>> api_hangup_hook that has access to the hangup/fax status.
>
> Kristian,
>
> I think you need this:
> http://wiki.freeswitch.org/wiki/Channel_Variables#session_in_hangup_hook
>
> It will let the call hang up but a copy of the session will be available to
> your hangup hook so you can access all the channel variables even after the
> channel has actually hung up.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From juanito1982 at gmail.com  Thu Oct 21 23:25:16 2010
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Fri, 22 Oct 2010 08:25:16 +0200
Subject: [Freeswitch-users] FreeTDM error
In-Reply-To: <20101021210700.097723F4779@cuneorg-email.cune.pri>
References: <AANLkTikpTQR_ODsb=jXuNDf2LtJojDEMZON_vnGYac6C@mail.gmail.com>
	<20101021210700.097723F4779@cuneorg-email.cune.pri>
Message-ID: <AANLkTim8wLD2Jj+_LQ-tv7xKfp-h7Cx7HYok9YPc9+oT@mail.gmail.com>

2010/10/21 <Russell.Mosemann at cune.org>

> Juan Antonio Iba?ez Santorum <juanito1982 at gmail.com> said:
>
> > ------------------
> > [root at nocheydia03 mod]# ldd ftmod_sangoma_isdn.so
> >         libsng_isdn.so.0 => /usr/lib64/libsng_isdn.so.0
>
> Keep going. You have to track down the problem. Does
> /usr/lib64/libsng_isdn.so.0 exist?


Yes


> Does it point to a file that exists in
> the same directory?


Yes,
-----------------------
[root at nocheydia03 lib64]# ls -l libsng*
-rw-r--r-- 1 root root 2177181 oct 21 19:40 libsng_isdn.so
lrwxrwxrwx 1 root root      14 oct 21 21:13 libsng_isdn.so.0 ->
libsng_isdn.so
-----------------------


> You showed that there was some kind of isdn library
> from ldconfig, but you didn't show that the library was listed in
> /usr/lib64 when running ldconfig.


Yes, it is under /usr/lib64:

http://pastebin.freeswitch.org/14288


> Are you running on a 64-bit OS of some
> kind?
>

Centos 5.4


>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From covici at ccs.covici.com  Fri Oct 22 01:00:11 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 22 Oct 2010 04:00:11 -0400
Subject: [Freeswitch-users] extension in use
Message-ID: <25918.1287734411@ccs.covici.com>

Is there any variable or function so I can see if an extension is in use
before dialing?  What I want to do is implement call waiting control or
maybe other things.

Thanks.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From juanbackson at gmail.com  Fri Oct 22 01:31:15 2010
From: juanbackson at gmail.com (Juan Backson)
Date: Fri, 22 Oct 2010 16:31:15 +0800
Subject: [Freeswitch-users] unable to compile fscomm
Message-ID: <AANLkTino2Ohfj6YX5ObFGooJJHOu91U_sZkNVK7ZEaDS@mail.gmail.com>

Hello

I am having difficulty compiling fscomm.  Can someone please help me out?
How can I resolve these compilation errors?


Makefile:682: warning: ignoring old commands for target
`debugtools/moc_statedebugdialog.cpp'
g++ -c -pipe -Wall -W -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2
-fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64
-mtune=generic  -DQT_NO_DEBUG -DQT_SHARED -DQT_THREAD_SUPPORT
-I/usr/lib64/qt-3.3/mkspecs/default -I. -I../src/include
-I../libs/apr/include -I../libs/libteletone/src -I/usr/lib64/qt-3.3/include
-o main.o main.cpp
main.cpp:30:30: error: QtGui/QApplication: No such file or directory
main.cpp:31:25: error: QSplashScreen: No such file or directory
In file included from main.cpp:32:
mainwindow.h:34:23: error: QMainWindow: No such file or directory
mainwindow.h:35:28: error: QTableWidgetItem: No such file or directory
mainwindow.h:36:25: error: QSignalMapper: No such file or directory
mainwindow.h:37:27: error: QSystemTrayIcon: No such file or directory
In file included from mainwindow.h:39,
                 from main.cpp:32:
./fshost.h:32:19: error: QThread: No such file or directory
./fshost.h:33:18: error: QColor: No such file or directory
./fshost.h:34:17: error: QHash: No such file or directory
./fshost.h:35:26: error: QSharedPointer: No such file or directory
In file included from ./fshost.h:37,
                 from mainwindow.h:39,
                 from main.cpp:32:
./call.h:32:18: error: QtCore: No such file or directory
./call.h:33:19: error: QString: No such file or directory
In file included from ./fscomm.h:5,
                 from ./fshost.h:40,
                 from mainwindow.h:39,
                 from main.cpp:32:
./isettings.h:4:19: error: QObject: No such file or directory
./isettings.h:5:17: error: QtXml: No such file or directory
In file included from mainwindow.h:42,
                 from main.cpp:32:
preferences/prefdialog.h:4:19: error: QDialog: No such file or directory
preferences/prefdialog.h:5:24: error: QDomDocument: No such file or
directory
In file included from mainwindow.h:43,
                 from main.cpp:32:
debugtools/consolewindow.h:4:17: error: QtGui: No such file or directory
In file included from debugtools/consolewindow.h:6,
                 from mainwindow.h:43,
                 from main.cpp:32:
debugtools/sortfilterproxymodel.h:41:33: error: QSortFilterProxyModel: No
such file or directory
debugtools/sortfilterproxymodel.h:42:31: error: QAbstractTableModel: No such
file or directory
debugtools/sortfilterproxymodel.h:43:19: error: QVector: No such file or
directory
debugtools/sortfilterproxymodel.h:44:17: error: QList: No such file or
directory
../src/include/switch_utils.h:379: warning: unused parameter ?number?
./channel.h:10: error: expected `)' before ?uuid?
./channel.h:11: error: ?QString? does not name a type
./channel.h:12: error: ?QString? has not been declared
./channel.h:13: error: ?QString? does not name a type
./channel.h:14: error: ?QString? has not been declared
./channel.h:15: error: ?QString? does not name a type
./channel.h:16: error: ?QString? has not been declared
./channel.h:17: error: ?QString? does not name a type
./channel.h:22: error: ?qulonglong? has not been declared
./channel.h:23: error: ?qulonglong? does not name a type
./channel.h:24: error: ?qulonglong? has not been declared
./channel.h:25: error: ?qulonglong? does not name a type
./channel.h:26: error: ?qulonglong? has not been declared
./channel.h:27: error: ?qulonglong? does not name a type
./channel.h:30: error: ?QString? does not name a type
./channel.h:31: error: ?QString? does not name a type
./channel.h:32: error: ?QString? does not name a type
./channel.h:33: error: ?QString? does not name a type
./channel.h:35: error: ?qulonglong? does not name a type
./channel.h:36: error: ?qulonglong? does not name a type
./channel.h:37: error: ?qulonglong? does not name a type
./channel.h: In member function ?void Channel::setCidName(int)?:
./channel.h:12: error: ?_cidName? was not declared in this scope
./channel.h: In member function ?void Channel::setCidNumber(int)?:
./channel.h:14: error: ?_cidNumber? was not declared in this scope
./channel.h: In member function ?void Channel::setDestinatinonNumber(int)?:
./channel.h:16: error: ?_destinationNumber? was not declared in this scope
./channel.h: In member function ?void Channel::setProgressEpoch(int)?:
./channel.h:22: error: ?_progressEpoch? was not declared in this scope
./channel.h: In member function ?void Channel::setProgressMediaEpoch(int)?:
./channel.h:24: error: ?_progressMediaEpoch? was not declared in this scope
./channel.h: In member function ?void Channel::setCreatedEpoch(int)?:
./channel.h:26: error: ?_createdEpoch? was not declared in this scope
./call.h: At global scope:
./call.h:37: error: expected constructor, destructor, or type conversion
before ?typedef?
./call.h:43: error: expected constructor, destructor, or type conversion
before ?;? token
./call.h:54: error: ?QString? does not name a type
./call.h:55: error: ?QString? does not name a type
./call.h:56: error: ?QString? does not name a type
./call.h:58: error: ?QSharedPointer? has not been declared
./call.h:58: error: expected ?,? or ?...? before ?<? token
./call.h:59: error: ISO C++ forbids declaration of ?QSharedPointer? with no
type
./call.h:59: error: expected ?;? before ?<? token
./call.h:60: error: expected `;' before ?void?
./call.h:60: error: ?QSharedPointer? has not been declared
./call.h:60: error: expected ?,? or ?...? before ?<? token
./call.h:61: error: ISO C++ forbids declaration of ?QSharedPointer? with no
type
./call.h:61: error: expected ?;? before ?<? token
./call.h:63: error: expected `;' before ?QString?
./call.h:63: error: ?QString? does not name a type
./call.h:64: error: ?QString? does not name a type
./call.h:68: error: ?fscomm_call_state_t? does not name a type
./call.h:69: error: ?fscomm_call_state_t? has not been declared
./call.h:70: error: ?QString? has not been declared
./call.h:71: error: ?QString? does not name a type
./call.h:76: error: ?QString? has not been declared
./call.h:77: error: ?qulonglong? has not been declared
./call.h:78: error: ?QTime? does not name a type
./call.h:83: error: ISO C++ forbids declaration of ?QSharedPointer? with no
type
./call.h:83: error: expected ?;? before ?<? token
./call.h:84: error: ISO C++ forbids declaration of ?QSharedPointer? with no
type
./call.h:84: error: expected ?;? before ?<? token
./call.h:86: error: ?QString? does not name a type
./call.h:90: error: ?QString? does not name a type
./call.h:91: error: ?fscomm_call_state_t? does not name a type
./call.h:92: error: ?qulonglong? does not name a type
./call.h: In member function ?void Call::setChannel(int)?:
./call.h:58: error: ?_channel? was not declared in this scope
./call.h:58: error: ?channel? was not declared in this scope
./call.h: In member function ?void Call::setOtherLegChannel(int)?:
./call.h:60: error: ?_otherLegChannel? was not declared in this scope
./call.h:60: error: ?channel? was not declared in this scope
./call.h: In member function ?int Call::getCallID()?:
./call.h:66: error: ?_channel? was not declared in this scope
./call.h: In member function ?void Call::setState(int)?:
./call.h:69: error: ?_state? was not declared in this scope
./call.h: In member function ?void Call::setCause(int)?:
./call.h:70: error: ?_cause? was not declared in this scope
./call.h:70: error: ?qDebug? was not declared in this scope
./call.h: In member function ?void Call::setAnsweredEpoch(int)?:
./call.h:77: error: ?_answeredEpoch? was not declared in this scope
./account.h: At global scope:
./account.h:7: error: expected constructor, destructor, or type conversion
before ?class?
make: *** [main.o] Error 1
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From dujinfang at gmail.com  Fri Oct 22 02:12:52 2010
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 22 Oct 2010 17:12:52 +0800
Subject: [Freeswitch-users] unable to compile fscomm
In-Reply-To: <AANLkTino2Ohfj6YX5ObFGooJJHOu91U_sZkNVK7ZEaDS@mail.gmail.com>
References: <AANLkTino2Ohfj6YX5ObFGooJJHOu91U_sZkNVK7ZEaDS@mail.gmail.com>
Message-ID: <AANLkTin-QaaGdJX-meStRNwH_2sQtfpy8BMQj0QEu6AT@mail.gmail.com>

You need install QT. Look the FSComm wiki page for instructions.

On Fri, Oct 22, 2010 at 4:31 PM, Juan Backson <juanbackson at gmail.com> wrote:
> Hello
>
> I am having difficulty compiling fscomm.? Can someone please help me out?
> How can I resolve these compilation errors?
>
>
> Makefile:682: warning: ignoring old commands for target
> `debugtools/moc_statedebugdialog.cpp'
> g++ -c -pipe -Wall -W -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2
> -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64
> -mtune=generic? -DQT_NO_DEBUG -DQT_SHARED -DQT_THREAD_SUPPORT
> -I/usr/lib64/qt-3.3/mkspecs/default -I. -I../src/include
> -I../libs/apr/include -I../libs/libteletone/src -I/usr/lib64/qt-3.3/include
> -o main.o main.cpp
> main.cpp:30:30: error: QtGui/QApplication: No such file or directory
> main.cpp:31:25: error: QSplashScreen: No such file or directory
> In file included from main.cpp:32:
> mainwindow.h:34:23: error: QMainWindow: No such file or directory
> mainwindow.h:35:28: error: QTableWidgetItem: No such file or directory
> mainwindow.h:36:25: error: QSignalMapper: No such file or directory
> mainwindow.h:37:27: error: QSystemTrayIcon: No such file or directory
> In file included from mainwindow.h:39,
> ???????????????? from main.cpp:32:
> ./fshost.h:32:19: error: QThread: No such file or directory
> ./fshost.h:33:18: error: QColor: No such file or directory
> ./fshost.h:34:17: error: QHash: No such file or directory
> ./fshost.h:35:26: error: QSharedPointer: No such file or directory
> In file included from ./fshost.h:37,
> ???????????????? from mainwindow.h:39,
> ???????????????? from main.cpp:32:
> ./call.h:32:18: error: QtCore: No such file or directory
> ./call.h:33:19: error: QString: No such file or directory
> In file included from ./fscomm.h:5,
> ???????????????? from ./fshost.h:40,
> ???????????????? from mainwindow.h:39,
> ???????????????? from main.cpp:32:
> ./isettings.h:4:19: error: QObject: No such file or directory
> ./isettings.h:5:17: error: QtXml: No such file or directory
> In file included from mainwindow.h:42,
> ???????????????? from main.cpp:32:
> preferences/prefdialog.h:4:19: error: QDialog: No such file or directory
> preferences/prefdialog.h:5:24: error: QDomDocument: No such file or
> directory
> In file included from mainwindow.h:43,
> ???????????????? from main.cpp:32:
> debugtools/consolewindow.h:4:17: error: QtGui: No such file or directory
> In file included from debugtools/consolewindow.h:6,
> ???????????????? from mainwindow.h:43,
> ???????????????? from main.cpp:32:
> debugtools/sortfilterproxymodel.h:41:33: error: QSortFilterProxyModel: No
> such file or directory
> debugtools/sortfilterproxymodel.h:42:31: error: QAbstractTableModel: No such
> file or directory
> debugtools/sortfilterproxymodel.h:43:19: error: QVector: No such file or
> directory
> debugtools/sortfilterproxymodel.h:44:17: error: QList: No such file or
> directory
> ../src/include/switch_utils.h:379: warning: unused parameter ?number?
> ./channel.h:10: error: expected `)' before ?uuid?
> ./channel.h:11: error: ?QString? does not name a type
> ./channel.h:12: error: ?QString? has not been declared
> ./channel.h:13: error: ?QString? does not name a type
> ./channel.h:14: error: ?QString? has not been declared
> ./channel.h:15: error: ?QString? does not name a type
> ./channel.h:16: error: ?QString? has not been declared
> ./channel.h:17: error: ?QString? does not name a type
> ./channel.h:22: error: ?qulonglong? has not been declared
> ./channel.h:23: error: ?qulonglong? does not name a type
> ./channel.h:24: error: ?qulonglong? has not been declared
> ./channel.h:25: error: ?qulonglong? does not name a type
> ./channel.h:26: error: ?qulonglong? has not been declared
> ./channel.h:27: error: ?qulonglong? does not name a type
> ./channel.h:30: error: ?QString? does not name a type
> ./channel.h:31: error: ?QString? does not name a type
> ./channel.h:32: error: ?QString? does not name a type
> ./channel.h:33: error: ?QString? does not name a type
> ./channel.h:35: error: ?qulonglong? does not name a type
> ./channel.h:36: error: ?qulonglong? does not name a type
> ./channel.h:37: error: ?qulonglong? does not name a type
> ./channel.h: In member function ?void Channel::setCidName(int)?:
> ./channel.h:12: error: ?_cidName? was not declared in this scope
> ./channel.h: In member function ?void Channel::setCidNumber(int)?:
> ./channel.h:14: error: ?_cidNumber? was not declared in this scope
> ./channel.h: In member function ?void Channel::setDestinatinonNumber(int)?:
> ./channel.h:16: error: ?_destinationNumber? was not declared in this scope
> ./channel.h: In member function ?void Channel::setProgressEpoch(int)?:
> ./channel.h:22: error: ?_progressEpoch? was not declared in this scope
> ./channel.h: In member function ?void Channel::setProgressMediaEpoch(int)?:
> ./channel.h:24: error: ?_progressMediaEpoch? was not declared in this scope
> ./channel.h: In member function ?void Channel::setCreatedEpoch(int)?:
> ./channel.h:26: error: ?_createdEpoch? was not declared in this scope
> ./call.h: At global scope:
> ./call.h:37: error: expected constructor, destructor, or type conversion
> before ?typedef?
> ./call.h:43: error: expected constructor, destructor, or type conversion
> before ?;? token
> ./call.h:54: error: ?QString? does not name a type
> ./call.h:55: error: ?QString? does not name a type
> ./call.h:56: error: ?QString? does not name a type
> ./call.h:58: error: ?QSharedPointer? has not been declared
> ./call.h:58: error: expected ?,? or ?...? before ?<? token
> ./call.h:59: error: ISO C++ forbids declaration of ?QSharedPointer? with no
> type
> ./call.h:59: error: expected ?;? before ?<? token
> ./call.h:60: error: expected `;' before ?void?
> ./call.h:60: error: ?QSharedPointer? has not been declared
> ./call.h:60: error: expected ?,? or ?...? before ?<? token
> ./call.h:61: error: ISO C++ forbids declaration of ?QSharedPointer? with no
> type
> ./call.h:61: error: expected ?;? before ?<? token
> ./call.h:63: error: expected `;' before ?QString?
> ./call.h:63: error: ?QString? does not name a type
> ./call.h:64: error: ?QString? does not name a type
> ./call.h:68: error: ?fscomm_call_state_t? does not name a type
> ./call.h:69: error: ?fscomm_call_state_t? has not been declared
> ./call.h:70: error: ?QString? has not been declared
> ./call.h:71: error: ?QString? does not name a type
> ./call.h:76: error: ?QString? has not been declared
> ./call.h:77: error: ?qulonglong? has not been declared
> ./call.h:78: error: ?QTime? does not name a type
> ./call.h:83: error: ISO C++ forbids declaration of ?QSharedPointer? with no
> type
> ./call.h:83: error: expected ?;? before ?<? token
> ./call.h:84: error: ISO C++ forbids declaration of ?QSharedPointer? with no
> type
> ./call.h:84: error: expected ?;? before ?<? token
> ./call.h:86: error: ?QString? does not name a type
> ./call.h:90: error: ?QString? does not name a type
> ./call.h:91: error: ?fscomm_call_state_t? does not name a type
> ./call.h:92: error: ?qulonglong? does not name a type
> ./call.h: In member function ?void Call::setChannel(int)?:
> ./call.h:58: error: ?_channel? was not declared in this scope
> ./call.h:58: error: ?channel? was not declared in this scope
> ./call.h: In member function ?void Call::setOtherLegChannel(int)?:
> ./call.h:60: error: ?_otherLegChannel? was not declared in this scope
> ./call.h:60: error: ?channel? was not declared in this scope
> ./call.h: In member function ?int Call::getCallID()?:
> ./call.h:66: error: ?_channel? was not declared in this scope
> ./call.h: In member function ?void Call::setState(int)?:
> ./call.h:69: error: ?_state? was not declared in this scope
> ./call.h: In member function ?void Call::setCause(int)?:
> ./call.h:70: error: ?_cause? was not declared in this scope
> ./call.h:70: error: ?qDebug? was not declared in this scope
> ./call.h: In member function ?void Call::setAnsweredEpoch(int)?:
> ./call.h:77: error: ?_answeredEpoch? was not declared in this scope
> ./account.h: At global scope:
> ./account.h:7: error: expected constructor, destructor, or type conversion
> before ?class?
> make: *** [main.o] Error 1
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From yehavi.bourvine at gmail.com  Fri Oct 22 04:36:50 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 22 Oct 2010 13:36:50 +0200
Subject: [Freeswitch-users] extension in use
In-Reply-To: <25918.1287734411@ccs.covici.com>
References: <25918.1287734411@ccs.covici.com>
Message-ID: <AANLkTinMgEbdvmU05yB1Fg2X4ijaPrFmJuQ1RXYWBZRA@mail.gmail.com>

Hi,

  We've implemented it using mod_limit. However, there are two issues you
should be aware of:

- When the call is transfered to voicemail the extension is still thought as
busy until the caller hangs up.
  We have been suggested to set limit_ignore_transfer  to false, but it
doesn't help (at least on 1.0.6).
  We are waiting fir the next release to try it again (production system
which we have un-tolerant users...).
- If another user picks the call (using intercept application) the original
callee is still thought as busy unless released
  manually. This causes problems when someone does a directed pickup of the
call and then wants to transfer it
  back to the orignal callee.

Besides that it works fine. We use it with a database field to decide what
to do when there is a call to a busy user:
reject with busy tone, play a "call waiting" tone or transfer to voicemail.
                   Regards, __Yehavi:

2010/10/22 <covici at ccs.covici.com>

> Is there any variable or function so I can see if an extension is in use
> before dialing?  What I want to do is implement call waiting control or
> maybe other things.
>
> Thanks.
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jmesquita at freeswitch.org  Fri Oct 22 05:13:02 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 22 Oct 2010 09:13:02 -0300
Subject: [Freeswitch-users] unable to compile fscomm
In-Reply-To: <AANLkTin-QaaGdJX-meStRNwH_2sQtfpy8BMQj0QEu6AT@mail.gmail.com>
References: <AANLkTino2Ohfj6YX5ObFGooJJHOu91U_sZkNVK7ZEaDS@mail.gmail.com>
	<AANLkTin-QaaGdJX-meStRNwH_2sQtfpy8BMQj0QEu6AT@mail.gmail.com>
Message-ID: <AANLkTiksuFddrCuCFdJ+AA20eW6u8C0oejMG6d-W4k3S@mail.gmail.com>

Thank you Seven.

Regards.
Jo?o Mesquita


On Fri, Oct 22, 2010 at 6:12 AM, Seven Du <dujinfang at gmail.com> wrote:

> You need install QT. Look the FSComm wiki page for instructions.
>
> On Fri, Oct 22, 2010 at 4:31 PM, Juan Backson <juanbackson at gmail.com>
> wrote:
> > Hello
> >
> > I am having difficulty compiling fscomm.  Can someone please help me out?
> > How can I resolve these compilation errors?
> >
> >
> > Makefile:682: warning: ignoring old commands for target
> > `debugtools/moc_statedebugdialog.cpp'
> > g++ -c -pipe -Wall -W -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2
> > -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64
> > -mtune=generic  -DQT_NO_DEBUG -DQT_SHARED -DQT_THREAD_SUPPORT
> > -I/usr/lib64/qt-3.3/mkspecs/default -I. -I../src/include
> > -I../libs/apr/include -I../libs/libteletone/src
> -I/usr/lib64/qt-3.3/include
> > -o main.o main.cpp
> > main.cpp:30:30: error: QtGui/QApplication: No such file or directory
> > main.cpp:31:25: error: QSplashScreen: No such file or directory
> > In file included from main.cpp:32:
> > mainwindow.h:34:23: error: QMainWindow: No such file or directory
> > mainwindow.h:35:28: error: QTableWidgetItem: No such file or directory
> > mainwindow.h:36:25: error: QSignalMapper: No such file or directory
> > mainwindow.h:37:27: error: QSystemTrayIcon: No such file or directory
> > In file included from mainwindow.h:39,
> >                  from main.cpp:32:
> > ./fshost.h:32:19: error: QThread: No such file or directory
> > ./fshost.h:33:18: error: QColor: No such file or directory
> > ./fshost.h:34:17: error: QHash: No such file or directory
> > ./fshost.h:35:26: error: QSharedPointer: No such file or directory
> > In file included from ./fshost.h:37,
> >                  from mainwindow.h:39,
> >                  from main.cpp:32:
> > ./call.h:32:18: error: QtCore: No such file or directory
> > ./call.h:33:19: error: QString: No such file or directory
> > In file included from ./fscomm.h:5,
> >                  from ./fshost.h:40,
> >                  from mainwindow.h:39,
> >                  from main.cpp:32:
> > ./isettings.h:4:19: error: QObject: No such file or directory
> > ./isettings.h:5:17: error: QtXml: No such file or directory
> > In file included from mainwindow.h:42,
> >                  from main.cpp:32:
> > preferences/prefdialog.h:4:19: error: QDialog: No such file or directory
> > preferences/prefdialog.h:5:24: error: QDomDocument: No such file or
> > directory
> > In file included from mainwindow.h:43,
> >                  from main.cpp:32:
> > debugtools/consolewindow.h:4:17: error: QtGui: No such file or directory
> > In file included from debugtools/consolewindow.h:6,
> >                  from mainwindow.h:43,
> >                  from main.cpp:32:
> > debugtools/sortfilterproxymodel.h:41:33: error: QSortFilterProxyModel: No
> > such file or directory
> > debugtools/sortfilterproxymodel.h:42:31: error: QAbstractTableModel: No
> such
> > file or directory
> > debugtools/sortfilterproxymodel.h:43:19: error: QVector: No such file or
> > directory
> > debugtools/sortfilterproxymodel.h:44:17: error: QList: No such file or
> > directory
> > ../src/include/switch_utils.h:379: warning: unused parameter ?number?
> > ./channel.h:10: error: expected `)' before ?uuid?
> > ./channel.h:11: error: ?QString? does not name a type
> > ./channel.h:12: error: ?QString? has not been declared
> > ./channel.h:13: error: ?QString? does not name a type
> > ./channel.h:14: error: ?QString? has not been declared
> > ./channel.h:15: error: ?QString? does not name a type
> > ./channel.h:16: error: ?QString? has not been declared
> > ./channel.h:17: error: ?QString? does not name a type
> > ./channel.h:22: error: ?qulonglong? has not been declared
> > ./channel.h:23: error: ?qulonglong? does not name a type
> > ./channel.h:24: error: ?qulonglong? has not been declared
> > ./channel.h:25: error: ?qulonglong? does not name a type
> > ./channel.h:26: error: ?qulonglong? has not been declared
> > ./channel.h:27: error: ?qulonglong? does not name a type
> > ./channel.h:30: error: ?QString? does not name a type
> > ./channel.h:31: error: ?QString? does not name a type
> > ./channel.h:32: error: ?QString? does not name a type
> > ./channel.h:33: error: ?QString? does not name a type
> > ./channel.h:35: error: ?qulonglong? does not name a type
> > ./channel.h:36: error: ?qulonglong? does not name a type
> > ./channel.h:37: error: ?qulonglong? does not name a type
> > ./channel.h: In member function ?void Channel::setCidName(int)?:
> > ./channel.h:12: error: ?_cidName? was not declared in this scope
> > ./channel.h: In member function ?void Channel::setCidNumber(int)?:
> > ./channel.h:14: error: ?_cidNumber? was not declared in this scope
> > ./channel.h: In member function ?void
> Channel::setDestinatinonNumber(int)?:
> > ./channel.h:16: error: ?_destinationNumber? was not declared in this
> scope
> > ./channel.h: In member function ?void Channel::setProgressEpoch(int)?:
> > ./channel.h:22: error: ?_progressEpoch? was not declared in this scope
> > ./channel.h: In member function ?void
> Channel::setProgressMediaEpoch(int)?:
> > ./channel.h:24: error: ?_progressMediaEpoch? was not declared in this
> scope
> > ./channel.h: In member function ?void Channel::setCreatedEpoch(int)?:
> > ./channel.h:26: error: ?_createdEpoch? was not declared in this scope
> > ./call.h: At global scope:
> > ./call.h:37: error: expected constructor, destructor, or type conversion
> > before ?typedef?
> > ./call.h:43: error: expected constructor, destructor, or type conversion
> > before ?;? token
> > ./call.h:54: error: ?QString? does not name a type
> > ./call.h:55: error: ?QString? does not name a type
> > ./call.h:56: error: ?QString? does not name a type
> > ./call.h:58: error: ?QSharedPointer? has not been declared
> > ./call.h:58: error: expected ?,? or ?...? before ?<? token
> > ./call.h:59: error: ISO C++ forbids declaration of ?QSharedPointer? with
> no
> > type
> > ./call.h:59: error: expected ?;? before ?<? token
> > ./call.h:60: error: expected `;' before ?void?
> > ./call.h:60: error: ?QSharedPointer? has not been declared
> > ./call.h:60: error: expected ?,? or ?...? before ?<? token
> > ./call.h:61: error: ISO C++ forbids declaration of ?QSharedPointer? with
> no
> > type
> > ./call.h:61: error: expected ?;? before ?<? token
> > ./call.h:63: error: expected `;' before ?QString?
> > ./call.h:63: error: ?QString? does not name a type
> > ./call.h:64: error: ?QString? does not name a type
> > ./call.h:68: error: ?fscomm_call_state_t? does not name a type
> > ./call.h:69: error: ?fscomm_call_state_t? has not been declared
> > ./call.h:70: error: ?QString? has not been declared
> > ./call.h:71: error: ?QString? does not name a type
> > ./call.h:76: error: ?QString? has not been declared
> > ./call.h:77: error: ?qulonglong? has not been declared
> > ./call.h:78: error: ?QTime? does not name a type
> > ./call.h:83: error: ISO C++ forbids declaration of ?QSharedPointer? with
> no
> > type
> > ./call.h:83: error: expected ?;? before ?<? token
> > ./call.h:84: error: ISO C++ forbids declaration of ?QSharedPointer? with
> no
> > type
> > ./call.h:84: error: expected ?;? before ?<? token
> > ./call.h:86: error: ?QString? does not name a type
> > ./call.h:90: error: ?QString? does not name a type
> > ./call.h:91: error: ?fscomm_call_state_t? does not name a type
> > ./call.h:92: error: ?qulonglong? does not name a type
> > ./call.h: In member function ?void Call::setChannel(int)?:
> > ./call.h:58: error: ?_channel? was not declared in this scope
> > ./call.h:58: error: ?channel? was not declared in this scope
> > ./call.h: In member function ?void Call::setOtherLegChannel(int)?:
> > ./call.h:60: error: ?_otherLegChannel? was not declared in this scope
> > ./call.h:60: error: ?channel? was not declared in this scope
> > ./call.h: In member function ?int Call::getCallID()?:
> > ./call.h:66: error: ?_channel? was not declared in this scope
> > ./call.h: In member function ?void Call::setState(int)?:
> > ./call.h:69: error: ?_state? was not declared in this scope
> > ./call.h: In member function ?void Call::setCause(int)?:
> > ./call.h:70: error: ?_cause? was not declared in this scope
> > ./call.h:70: error: ?qDebug? was not declared in this scope
> > ./call.h: In member function ?void Call::setAnsweredEpoch(int)?:
> > ./call.h:77: error: ?_answeredEpoch? was not declared in this scope
> > ./account.h: At global scope:
> > ./account.h:7: error: expected constructor, destructor, or type
> conversion
> > before ?class?
> > make: *** [main.o] Error 1
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Fri Oct 22 05:28:31 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 22 Oct 2010 13:28:31 +0100
Subject: [Freeswitch-users] extension in use
In-Reply-To: <AANLkTinMgEbdvmU05yB1Fg2X4ijaPrFmJuQ1RXYWBZRA@mail.gmail.com>
References: <25918.1287734411@ccs.covici.com>
	<AANLkTinMgEbdvmU05yB1Fg2X4ijaPrFmJuQ1RXYWBZRA@mail.gmail.com>
Message-ID: <AANLkTinCaOc+PtcfqBAcJT1g6iuvaQUTVWXQQYB55f23@mail.gmail.com>

On 22 October 2010 12:36, Yehavi Bourvine <yehavi.bourvine at gmail.com> wrote:
> Hi,
>
> ? We've implemented it using mod_limit. However, there are two issues you
> should be aware of:
>
> - When the call is transfered to voicemail the extension is still thought as
> busy until the caller hangs up.
> ? We have been suggested to set limit_ignore_transfer? to false, but it
> doesn't help (at least on 1.0.6).

It does. You need to transfer to a different extension if the number
is busy which sends the call to voicemail though - it removes the
extra limit during that transfer.

> ? We are waiting fir the next release to try it again (production system
> which we have un-tolerant users...).

I'd suggest you set up a test system in a virtual machine to test it out on.

> - If another user picks the call (using intercept application) the original
> callee is still thought as busy unless released
> ? manually. This causes problems when someone does a directed pickup of the
> call and then wants to transfer it
> ? back to the orignal callee.
>
> Besides that it works fine. We use it with a database field to decide what
> to do when there is a call to a busy user:
> reject with busy tone, play a "call waiting" tone or transfer to voicemail.
> ?????????????????? Regards, __Yehavi:
>
> 2010/10/22 <covici at ccs.covici.com>
>>
>> Is there any variable or function so I can see if an extension is in use
>> before dialing? ?What I want to do is implement call waiting control or
>> maybe other things.
>>
>> Thanks.
>>
>> --
>> Your life is like a penny. ?You're going to lose it. ?The question is:
>> How do
>> you spend it?
>>
>> ? ? ? ? John Covici
>> ? ? ? ? covici at ccs.covici.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From covici at ccs.covici.com  Fri Oct 22 05:39:52 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 22 Oct 2010 08:39:52 -0400
Subject: [Freeswitch-users] extension in use
In-Reply-To: <AANLkTinMgEbdvmU05yB1Fg2X4ijaPrFmJuQ1RXYWBZRA@mail.gmail.com>
References: <25918.1287734411@ccs.covici.com>
	<AANLkTinMgEbdvmU05yB1Fg2X4ijaPrFmJuQ1RXYWBZRA@mail.gmail.com>
Message-ID: <28441.1287751192@ccs.covici.com>

I have latest git, so if you send me your code, I can see if it works.

I would like to see it anyway, it may give me some ideas.

Thanks for your suggestions.

Yehavi Bourvine <yehavi.bourvine at gmail.com> wrote:

> Hi,
> 
>   We've implemented it using mod_limit. However, there are two issues you
> should be aware of:
> 
> - When the call is transfered to voicemail the extension is still thought as
> busy until the caller hangs up.
>   We have been suggested to set limit_ignore_transfer  to false, but it
> doesn't help (at least on 1.0.6).
>   We are waiting fir the next release to try it again (production system
> which we have un-tolerant users...).
> - If another user picks the call (using intercept application) the original
> callee is still thought as busy unless released
>   manually. This causes problems when someone does a directed pickup of the
> call and then wants to transfer it
>   back to the orignal callee.
> 
> Besides that it works fine. We use it with a database field to decide what
> to do when there is a call to a busy user:
> reject with busy tone, play a "call waiting" tone or transfer to voicemail.
>                    Regards, __Yehavi:
> 
> 2010/10/22 <covici at ccs.covici.com>
> 
> > Is there any variable or function so I can see if an extension is in use
> > before dialing?  What I want to do is implement call waiting control or
> > maybe other things.
> >
> > Thanks.
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From Russell.Mosemann at cune.org  Fri Oct 22 06:05:02 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Fri, 22 Oct 2010 13:05:02 -0000
Subject: [Freeswitch-users] FreeTDM error
In-Reply-To: <AANLkTim8wLD2Jj+_LQ-tv7xKfp-h7Cx7HYok9YPc9+oT@mail.gmail.com>
Message-ID: <20101022130502.D96A23F3240@cuneorg-email.cune.pri>

Everything you have provided so far looks correct. Is FS running chroot?
If capabilities are not turned off, temporarily turn them off and see if
FS will run. Are there possibly any ACLs attached to the
files/directories that would prevent root from accessing the library? Are
there any helpful error messages in any of the system logs?

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From jonas.gauffin at gmail.com  Fri Oct 22 06:26:59 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Fri, 22 Oct 2010 15:26:59 +0200
Subject: [Freeswitch-users] FS disk thrashing after 1440 Auto-Adding
 Alias + segmentation fault
In-Reply-To: <AANLkTin0XbC2iLb=-YWvQgG_citJGbKKEGVRi5m9s7Py@mail.gmail.com>
References: <4CAB667E.4060405@mgtech.com>
	<AANLkTin0XbC2iLb=-YWvQgG_citJGbKKEGVRi5m9s7Py@mail.gmail.com>
Message-ID: <AANLkTimD15j=wQrAc8ZkWy0Spa1j5NN9Ym=zK1RoNJCU@mail.gmail.com>

Can fs detect register or invite ddos and generate an event with the
offending ip? Could be used through the event socket to add fw rules
automatically
Den 2010 10 5 20:53 skrev "Anthony Minessale" <anthony.minessale at gmail.com>:
> Are you perhaps getting ddos'd by some sip scanner?
>
>
> On Tue, Oct 5, 2010 at 12:55 PM, Mario <mario_fs at mgtech.com> wrote:
>> Starting last night at 2am (it had been up for 10 hours) FS issued:
>> 1440 Auto-Adding Alias [208.2.3.4] for profile [internal].
>> Now, when I start FS I get the message right away and heavy disk
>> thrashing. For the last couple of weeks while working on FS this
>> messages never came up (that I noticed). The machine goes to 100 percent
>> processor and the disk thrashes. I removed all external SIP accounts and
>> the ip-v6 profile to test and it still happens. This came out of the
>> blue and stopped everything I was working on. Looking into this all I
>> could find was it may be related to UPnP which the router has. Had been
>> working fine until now. Any help greatly appreciated. Mario
>>
>> When I issue the shutdown command FS end with an error:
>>
>> 2010-10-05 10:22:44.889143 [CONSOLE] switch_loadable_module.c:1401
>> mod_siren has no shutdown routine
>> 2010-10-05 10:22:44.889167 [NOTICE] switch_loadable_module.c:518
>> Deleting Codec SPEEX 99 Speex 32000hz 20ms
>> 2010-10-05 10:22:44.889213 [NOTICE] switch_loadable_module.c:518
>> Deleting Codec SPEEX 99 Speex 16000hz 20ms
>> 2010-10-05 10:22:44.889257 [NOTICE] switch_loadable_module.c:518
>> Deleting Codec SPEEX 99 Speex 8000hz 20ms
>> 2010-10-05 10:22:44.889330 [CONSOLE] switch_loadable_module.c:1401
>> mod_speex has no shutdown routine
>> Segmentation fault (core dumped)
>>
>>                     Name          Type
>>      Data      State
>>
=================================================================================================
>>          210.17.234.127         alias
>>  internal      ALIASED
>>                 internal       profile
>> sip:mod_sofia at 100.24.1.37:5060      RUNNING (0)
>>                 external       profile
>> sip:mod_sofia at 100.24.1.37:5080      RUNNING (0)
>>              100.24.1.37         alias
>>  internal      ALIASED
>>
=================================================================================================
>> 2 profiles 2 aliases
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mustafa.pk at gmail.com  Fri Oct 22 06:39:32 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Fri, 22 Oct 2010 18:39:32 +0500
Subject: [Freeswitch-users] FS disk thrashing after 1440 Auto-Adding
 Alias + segmentation fault
In-Reply-To: <AANLkTimD15j=wQrAc8ZkWy0Spa1j5NN9Ym=zK1RoNJCU@mail.gmail.com>
References: <4CAB667E.4060405@mgtech.com>
	<AANLkTin0XbC2iLb=-YWvQgG_citJGbKKEGVRi5m9s7Py@mail.gmail.com>
	<AANLkTimD15j=wQrAc8ZkWy0Spa1j5NN9Ym=zK1RoNJCU@mail.gmail.com>
Message-ID: <AANLkTinmCJv4ZztGpnT5cP3r2+mRKryvfbEXPG6wH-W3@mail.gmail.com>

perhaps u need to read this http://www.fail2ban.org/wiki/index.php/Main_Page

On Fri, Oct 22, 2010 at 6:26 PM, Jonas Gauffin <jonas.gauffin at gmail.com> wrote:
> Can fs detect register or invite ddos and generate an event with the
> offending ip? Could be used through the event socket to add fw rules
> automatically
>
> Den 2010 10 5 20:53 skrev "Anthony Minessale" <anthony.minessale at gmail.com>:
>> Are you perhaps getting ddos'd by some sip scanner?
>>
>>
>> On Tue, Oct 5, 2010 at 12:55 PM, Mario <mario_fs at mgtech.com> wrote:
>>> Starting last night at 2am (it had been up for 10 hours) FS issued:
>>> 1440 Auto-Adding Alias [208.2.3.4] for profile [internal].
>>> Now, when I start FS I get the message right away and heavy disk
>>> thrashing. For the last couple of weeks while working on FS this
>>> messages never came up (that I noticed). The machine goes to 100 percent
>>> processor and the disk thrashes. I removed all external SIP accounts and
>>> the ip-v6 profile to test and it still happens. This came out of the
>>> blue and stopped everything I was working on. Looking into this all I
>>> could find was it may be related to UPnP which the router has. Had been
>>> working fine until now. Any help greatly appreciated. Mario
>>>
>>> When I issue the shutdown command FS end with an error:
>>>
>>> 2010-10-05 10:22:44.889143 [CONSOLE] switch_loadable_module.c:1401
>>> mod_siren has no shutdown routine
>>> 2010-10-05 10:22:44.889167 [NOTICE] switch_loadable_module.c:518
>>> Deleting Codec SPEEX 99 Speex 32000hz 20ms
>>> 2010-10-05 10:22:44.889213 [NOTICE] switch_loadable_module.c:518
>>> Deleting Codec SPEEX 99 Speex 16000hz 20ms
>>> 2010-10-05 10:22:44.889257 [NOTICE] switch_loadable_module.c:518
>>> Deleting Codec SPEEX 99 Speex 8000hz 20ms
>>> 2010-10-05 10:22:44.889330 [CONSOLE] switch_loadable_module.c:1401
>>> mod_speex has no shutdown routine
>>> Segmentation fault (core dumped)
>>>
>>> ? ? ? ? ? ? ? ? ? ? Name ? ? ? ? ?Type
>>> ? ? ?Data ? ? ?State
>>>
>>> =================================================================================================
>>> ? ? ? ? ?210.17.234.127 ? ? ? ? alias
>>> ?internal ? ? ?ALIASED
>>> ? ? ? ? ? ? ? ? internal ? ? ? profile
>>> sip:mod_sofia at 100.24.1.37:5060 ? ? ?RUNNING (0)
>>> ? ? ? ? ? ? ? ? external ? ? ? profile
>>> sip:mod_sofia at 100.24.1.37:5080 ? ? ?RUNNING (0)
>>> ? ? ? ? ? ? ?100.24.1.37 ? ? ? ? alias
>>> ?internal ? ? ?ALIASED
>>>
>>> =================================================================================================
>>> 2 profiles 2 aliases
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com


From curriegrad2004 at gmail.com  Fri Oct 22 06:42:29 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Fri, 22 Oct 2010 06:42:29 -0700
Subject: [Freeswitch-users] FS disk thrashing after 1440 Auto-Adding
 Alias + segmentation fault
In-Reply-To: <AANLkTimD15j=wQrAc8ZkWy0Spa1j5NN9Ym=zK1RoNJCU@mail.gmail.com>
References: <4CAB667E.4060405@mgtech.com>
	<AANLkTin0XbC2iLb=-YWvQgG_citJGbKKEGVRi5m9s7Py@mail.gmail.com>
	<AANLkTimD15j=wQrAc8ZkWy0Spa1j5NN9Ym=zK1RoNJCU@mail.gmail.com>
Message-ID: <AANLkTi=7gA98cziPYTn=AuH2CoAOPDmkL5Gs4v0-6hQE@mail.gmail.com>

Fail2ban protects against this. Somebody mentioned it here on the
mailing list a few days ago and it's also on the wiki.

On Fri, Oct 22, 2010 at 6:26 AM, Jonas Gauffin <jonas.gauffin at gmail.com> wrote:
> Can fs detect register or invite ddos and generate an event with the
> offending ip? Could be used through the event socket to add fw rules
> automatically
>
> Den 2010 10 5 20:53 skrev "Anthony Minessale" <anthony.minessale at gmail.com>:
>> Are you perhaps getting ddos'd by some sip scanner?
>>
>>
>> On Tue, Oct 5, 2010 at 12:55 PM, Mario <mario_fs at mgtech.com> wrote:
>>> Starting last night at 2am (it had been up for 10 hours) FS issued:
>>> 1440 Auto-Adding Alias [208.2.3.4] for profile [internal].
>>> Now, when I start FS I get the message right away and heavy disk
>>> thrashing. For the last couple of weeks while working on FS this
>>> messages never came up (that I noticed). The machine goes to 100 percent
>>> processor and the disk thrashes. I removed all external SIP accounts and
>>> the ip-v6 profile to test and it still happens. This came out of the
>>> blue and stopped everything I was working on. Looking into this all I
>>> could find was it may be related to UPnP which the router has. Had been
>>> working fine until now. Any help greatly appreciated. Mario
>>>
>>> When I issue the shutdown command FS end with an error:
>>>
>>> 2010-10-05 10:22:44.889143 [CONSOLE] switch_loadable_module.c:1401
>>> mod_siren has no shutdown routine
>>> 2010-10-05 10:22:44.889167 [NOTICE] switch_loadable_module.c:518
>>> Deleting Codec SPEEX 99 Speex 32000hz 20ms
>>> 2010-10-05 10:22:44.889213 [NOTICE] switch_loadable_module.c:518
>>> Deleting Codec SPEEX 99 Speex 16000hz 20ms
>>> 2010-10-05 10:22:44.889257 [NOTICE] switch_loadable_module.c:518
>>> Deleting Codec SPEEX 99 Speex 8000hz 20ms
>>> 2010-10-05 10:22:44.889330 [CONSOLE] switch_loadable_module.c:1401
>>> mod_speex has no shutdown routine
>>> Segmentation fault (core dumped)
>>>
>>> ? ? ? ? ? ? ? ? ? ? Name ? ? ? ? ?Type
>>> ? ? ?Data ? ? ?State
>>>
>>> =================================================================================================
>>> ? ? ? ? ?210.17.234.127 ? ? ? ? alias
>>> ?internal ? ? ?ALIASED
>>> ? ? ? ? ? ? ? ? internal ? ? ? profile
>>> sip:mod_sofia at 100.24.1.37:5060 ? ? ?RUNNING (0)
>>> ? ? ? ? ? ? ? ? external ? ? ? profile
>>> sip:mod_sofia at 100.24.1.37:5080 ? ? ?RUNNING (0)
>>> ? ? ? ? ? ? ?100.24.1.37 ? ? ? ? alias
>>> ?internal ? ? ?ALIASED
>>>
>>> =================================================================================================
>>> 2 profiles 2 aliases
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From freeswitch-list at puzzled.xs4all.nl  Fri Oct 22 06:47:22 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Fri, 22 Oct 2010 15:47:22 +0200
Subject: [Freeswitch-users] FreeTDM error
In-Reply-To: <20101022130502.D96A23F3240@cuneorg-email.cune.pri>
References: <20101022130502.D96A23F3240@cuneorg-email.cune.pri>
Message-ID: <4CC195EA.90404@puzzled.xs4all.nl>

On 10/22/2010 03:05 PM, Russell.Mosemann at cune.org wrote:
> Everything you have provided so far looks correct. Is FS running chroot?
> If capabilities are not turned off, temporarily turn them off and see if
> FS will run. Are there possibly any ACLs attached to the
> files/directories that would prevent root from accessing the library? Are
> there any helpful error messages in any of the system logs?

Another suggestion: how about selinux? Is it active? If yes, check with 
ls -Z /usr/lib64/libsng_isdn.so.* if it has the proper security context. 
Should be something like: system_u:object_r:lib_t:s0
To set/restore the proper security context read man restorecon.

Regards,
Patrick



From codecomplete at free.fr  Fri Oct 22 06:49:59 2010
From: codecomplete at free.fr (GillesToo)
Date: Fri, 22 Oct 2010 06:49:59 -0700 (PDT)
Subject: [Freeswitch-users] [Linksys 3102] Couple of issues using FS and
	XLite
Message-ID: <1287755399019-5662612.post@n2.nabble.com>


Hello

I followed this tutorial to configure a Linksys 3102 to connect a Freeswitch
server to an analog phone line:

http://robsmart.co.uk/2009/06/02/freeswitch_linksys3102/

I could successfully call my cellphone, but I noticed a couple of issues:

1. While the call is still in progress, ie. I haven't answered the cellphone
yet, XLite says "Call established"
2. After I hang up the cellphone, XLite is still off-hook, and I need to
hang up manually

Is this due to some setting in XLite, in Freeswitch, or in the 3102?

Thank you.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Linksys-3102-Couple-of-issues-using-FS-and-XLite-tp5662612p5662612.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jason at jasonjgw.net  Thu Oct 21 19:23:22 2010
From: jason at jasonjgw.net (Jason White)
Date: Fri, 22 Oct 2010 13:23:22 +1100
Subject: [Freeswitch-users] Binding to external IP address during system
	boot
References: <87wrpj0wdu.fsf@jdc.jasonjgw.net>
	<7B07DBCC-65FA-4046-AC7A-E00D5F03C72E@freeswitch.org>
Message-ID: <87eibjc6ed.fsf@jdc.jasonjgw.net>

Brian West <brian at freeswitch.org> writes:

> Are you on the latest FS GIT?

It's several weeks old, but I'll upgrade and continue the thread if the
problem persists.

Restarting the SIP profile didn't help, and (at least in my version)
there's no other sofia reload command besides reloadxml, which obviously
isn't relevant here.



From teddywu72 at gmail.com  Fri Oct 22 00:19:15 2010
From: teddywu72 at gmail.com (Chia-Yen Wu)
Date: Fri, 22 Oct 2010 15:19:15 +0800
Subject: [Freeswitch-users] Music On Hold
Message-ID: <AANLkTin_qHEUnoFUo1jkX1xyWyyRiYe7sTE1P4czayUh@mail.gmail.com>

Hello,

I am using freeswitch as an IVR server.
I would like to be able to put the call on hold and play music while
performing some other tasks in the lua.
when the task is complete , the music stop and the IVR continue

but i cant find a way to do that, is there any function can do "music on
hold" in dialplan ?

Thank you
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From yivzhenko at mksat.net  Fri Oct 22 02:16:11 2010
From: yivzhenko at mksat.net (Yuriy Ivzhenko)
Date: Fri, 22 Oct 2010 12:16:11 +0300
Subject: [Freeswitch-users] record_session with fifo stops after transfer
Message-ID: <201010221216.11857.yivzhenko@mksat.net>

I route incoming calls to fifo.
And i want to record all conversation of incoming caller.

I call record_session just before fifo.
All works, but if the fifo agent transfer the call to another extension, record 
stops in a transfer point.
uuid_record give the same results.

If i call record_session after transfer (on A-leg) - nothing recordings, 
If i call record_session after transfer (on B-leg) - recording works, but this 
is not-so-good decision.
I want to record all A-leg conversation. Is this possible, or i do not 
understand something?


<extension name="ffCallCenter">
          <condition field="destination_number" expression="^ffCallCenter$"/>
          <condition field="${fifo(list)}" expression='<fifo 
name="fifo_CallCenter_WP at default[\s\S]*<member[\s\S]
*/fifo>[\s\S]*<fifo name="manual_calls'>
        <action application="answer"/>
	<action application="record_session" 
data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination
_number}_${caller_id_number}.wav"/> 

<!--    <action application="set" data="api_result=${uuid_record(${uuid} start 
$${base_dir}/recordings/${strftime(%Y-%m-%d-
%H-%M-%S)}_${destination_number}_${caller_id_number}_${uuid}.wav)}"/> -->

            <action application="set" data="fifo_orbit_exten=ffGirls:17"/>
            <action application="set" data="fifo_orbit_dialplan=XML"/>
            <action application="set" data="fifo_orbit_context=default"/>
            <action application="set" data="fifo_caller_exit_key=#"/>
            <action application="set" data="fifo_caller_exit_to_orbit=true"/>
            <action application="fifo" data="fifo_CallCenter_WP at default in undef 
local_stream://moh"/>

          </condition>
</extension>


From mustafa.pk at gmail.com  Fri Oct 22 07:08:35 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Fri, 22 Oct 2010 19:08:35 +0500
Subject: [Freeswitch-users] [Linksys 3102] Couple of issues using FS and
	XLite
In-Reply-To: <1287755399019-5662612.post@n2.nabble.com>
References: <1287755399019-5662612.post@n2.nabble.com>
Message-ID: <AANLkTin2q7o1DHp0tUguHiVMr1aJxLo4ryoedAHgW18_@mail.gmail.com>

Please post your dialplan's entries dealing with xlite and linksys.

regards.


On Fri, Oct 22, 2010 at 6:49 PM, GillesToo <codecomplete at free.fr> wrote:
>
> Hello
>
> I followed this tutorial to configure a Linksys 3102 to connect a Freeswitch
> server to an analog phone line:
>
> http://robsmart.co.uk/2009/06/02/freeswitch_linksys3102/
>
> I could successfully call my cellphone, but I noticed a couple of issues:
>
> 1. While the call is still in progress, ie. I haven't answered the cellphone
> yet, XLite says "Call established"
> 2. After I hang up the cellphone, XLite is still off-hook, and I need to
> hang up manually
>
> Is this due to some setting in XLite, in Freeswitch, or in the 3102?
>
> Thank you.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Linksys-3102-Couple-of-issues-using-FS-and-XLite-tp5662612p5662612.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com


From asilva at wirelessmundi.com  Fri Oct 22 07:18:25 2010
From: asilva at wirelessmundi.com (Antonio)
Date: Fri, 22 Oct 2010 16:18:25 +0200
Subject: [Freeswitch-users] Music On Hold
In-Reply-To: <AANLkTin_qHEUnoFUo1jkX1xyWyyRiYe7sTE1P4czayUh@mail.gmail.com>
References: <AANLkTin_qHEUnoFUo1jkX1xyWyyRiYe7sTE1P4czayUh@mail.gmail.com>
Message-ID: <1287757105.32635.28.camel@marces.tc.commsmundi.com>

You can try to put in your dialplan 

<action application="set" data="hold_music=/path/to/your/music.mp3"/>



On Fri, 2010-10-22 at 15:19 +0800, Chia-Yen Wu wrote:
> Hello, 
> 
> 
> I am using freeswitch as an IVR server. 
> I would like to be able to put the call on hold and play music while
> performing some other tasks in the lua. 
> when the task is complete , the music stop and the IVR continue
> 
> 
> but i cant find a way to do that, is there any function can do "music
> on hold" in dialplan ?
> 
> 
> Thank you
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 

Un cordial saludo / Best regards, 

 _________________________

Ant?nio Silva

E-mail:asilva at wirelessmundi.com



From Peter.Hinman at ParcelPool.com  Fri Oct 22 07:52:37 2010
From: Peter.Hinman at ParcelPool.com (Peter Hinman)
Date: Fri, 22 Oct 2010 08:52:37 -0600
Subject: [Freeswitch-users] ODBC and MSSQL
In-Reply-To: <1286459189732-5611425.post@n2.nabble.com>
References: <4CAB9F5A.2090503@ParcelPool.com>	<1286331931285-5605569.post@n2.nabble.com>	<4CACE7C5.6060208@ParcelPool.com>
	<1286459189732-5611425.post@n2.nabble.com>
Message-ID: <4CC1A535.9010706@ParcelPool.com>

Hi Jeff -

The initial test looks good.  Patched, compiled, and started on two 
different FreeBSD boxes.  Tables appear to have been created on the 
database correctly.

I'll do some further testing today and then try it on a live system 
after that.  Thanks for putting this together!

Peter


On 10/7/2010 7:46 AM, Jeff Lenk wrote:
> Hi Peter,
>
> If your so inclined please test the patch here:
> http://jira.freeswitch.org/browse/FS-2050
>
> and please let me know how it works for you.
>
> After patching and building
> remove commented ->  <!--  -->
>
>
> -Jeff




From codecomplete at free.fr  Fri Oct 22 07:55:34 2010
From: codecomplete at free.fr (GillesToo)
Date: Fri, 22 Oct 2010 07:55:34 -0700 (PDT)
Subject: [Freeswitch-users] [Linksys 3102] Couple of issues using FS and
	XLite
In-Reply-To: <AANLkTin2q7o1DHp0tUguHiVMr1aJxLo4ryoedAHgW18_@mail.gmail.com>
References: <1287755399019-5662612.post@n2.nabble.com>
	<AANLkTin2q7o1DHp0tUguHiVMr1aJxLo4ryoedAHgW18_@mail.gmail.com>
Message-ID: <1287759334654-5662810.post@n2.nabble.com>


Thanks for the help.

I use the default settings as provided in the binary for Windows (1.0.4).
XLite uses extension 1001, and the Linksys uses extension 1000.

The only changes I made were those in the tutorial above, in the "FreeSwitch
Configuration" section:

1. Added conf/dialplan/default/00_spa3102.xml:

<include>
<extension name=?To PSTN?>
<condition field=?destination_number? expression=?(.*)?>
<action application=?bridge? data=?sofia/internal/${destination_number}@[IP
address of your SPA3102]:5061? />
</condition>
</extension>
</include>

2. Edited conf/directory/default/1000.xml to remove the two CID-related
lines:

<variable name=?effective_caller_id_name? value=?Extension 1001?/>
<variable name=?effective_caller_id_number? value=?1001?/>

I just called my cellphone using a handset connected to the Linksys FXS
port, and when I hang up the cellphone, I can hear the busy signal so it
looks like the Linksys and/or Freeswitch does detect the hangup signal
correctly, and the issue is specific to Freeswitch/XLite.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Linksys-3102-Couple-of-issues-using-FS-and-XLite-tp5662612p5662810.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jeff at jefflenk.com  Fri Oct 22 08:13:09 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 22 Oct 2010 08:13:09 -0700 (PDT)
Subject: [Freeswitch-users] ODBC and MSSQL
In-Reply-To: <4CC1A535.9010706@ParcelPool.com>
References: <4CAB9F5A.2090503@ParcelPool.com>
	<1286331931285-5605569.post@n2.nabble.com>
	<4CACE7C5.6060208@ParcelPool.com>
	<1286459189732-5611425.post@n2.nabble.com>
	<4CC1A535.9010706@ParcelPool.com>
Message-ID: <1287760389322-5662887.post@n2.nabble.com>


Thanks Peter,

btw that patch was committed yesterday to git with some minor modifications
which do not change any functionality(MSSQL). It also preserves previous
table compatibility for non MSSQL based instances of fs.

-Jeff
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ODBC-and-MSSQL-tp5605076p5662887.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mario_fs at mgtech.com  Fri Oct 22 09:04:22 2010
From: mario_fs at mgtech.com (Mario)
Date: Fri, 22 Oct 2010 09:04:22 -0700
Subject: [Freeswitch-users] Can change email wav encoding?
Message-ID: <4CC1B606.6030701@mgtech.com>

Is there a way to change the bitrate of the wav email message? iPhone is
picky about wav files. I got mp3 to work from Linux but it's impossible
to build mp3 support on osX due to libvorbis problems. Thanks.
Mario


From ovvenkatesan at gmail.com  Fri Oct 22 09:08:21 2010
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Fri, 22 Oct 2010 21:38:21 +0530
Subject: [Freeswitch-users] dialogic FXO card will support freeSWITCH
Message-ID: <AANLkTimfFw0pLcivCh+PLkY6kEBEZ25_KRK4-OmXacyT@mail.gmail.com>

Hi.

Can you anyone plz tell me, whether Dialogic FXO card will support
freeswitch?


thanks,

Regards,
Venkat.

-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From david.ponzone at ipeva.fr  Fri Oct 22 09:15:12 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 22 Oct 2010 18:15:12 +0200
Subject: [Freeswitch-users] Can change email wav encoding?
In-Reply-To: <4CC1B606.6030701@mgtech.com>
References: <4CC1B606.6030701@mgtech.com>
Message-ID: <4EA769AD-0583-499E-9A1E-C00F4E491429@ipeva.fr>

Perhaps you could include some conversion with sox somewhere in your  
scripts ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 22/10/2010 ? 18:04, Mario a ?crit :

> Is there a way to change the bitrate of the wav email message?  
> iPhone is
> picky about wav files. I got mp3 to work from Linux but it's  
> impossible
> to build mp3 support on osX due to libvorbis problems. Thanks.
> Mario
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mario_fs at mgtech.com  Fri Oct 22 09:28:11 2010
From: mario_fs at mgtech.com (Mario)
Date: Fri, 22 Oct 2010 09:28:11 -0700
Subject: [Freeswitch-users] Can change email wav encoding?
In-Reply-To: <4EA769AD-0583-499E-9A1E-C00F4E491429@ipeva.fr>
References: <4CC1B606.6030701@mgtech.com>
	<4EA769AD-0583-499E-9A1E-C00F4E491429@ipeva.fr>
Message-ID: <4CC1BB9B.7030406@mgtech.com>

I am trying to avoid having to add more tools to osx (spent 50 times
longer on libvorbis install than FS and it still wont work) so it would
be much simpler to have the wav come out of FS in the format needed for
iPhone email.

On 10/22/10 09:15, David Ponzone wrote:
> Perhaps you could include some conversion with sox somewhere in your
> scripts ?
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com
> <BLOCKED::http://www.ipeva-studio.com/>
> 
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion non autoris?e est interdite. Tout message ?lectronique est
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous
> n'?tes pas destinataire de ce message, merci de le d?truire
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
> 
> 
> 
> Le 22/10/2010 ? 18:04, Mario a ?crit :
> 
>> Is there a way to change the bitrate of the wav email message? iPhone is
>> picky about wav files. I got mp3 to work from Linux but it's impossible
>> to build mp3 support on osX due to libvorbis problems. Thanks.
>> Mario
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From anthony.minessale at gmail.com  Fri Oct 22 09:36:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 22 Oct 2010 11:36:51 -0500
Subject: [Freeswitch-users] Can change email wav encoding?
In-Reply-To: <4CC1BB9B.7030406@mgtech.com>
References: <4CC1B606.6030701@mgtech.com>
	<4EA769AD-0583-499E-9A1E-C00F4E491429@ipeva.fr>
	<4CC1BB9B.7030406@mgtech.com>
Message-ID: <AANLkTimOk0rsTa10gVsw0uqsNAtenU0dnB7JtRVFXrxM@mail.gmail.com>

what format does it want?


On Fri, Oct 22, 2010 at 11:28 AM, Mario <mario_fs at mgtech.com> wrote:
> I am trying to avoid having to add more tools to osx (spent 50 times
> longer on libvorbis install than FS and it still wont work) so it would
> be much simpler to have the wav come out of FS in the format needed for
> iPhone email.
>
> On 10/22/10 09:15, David Ponzone wrote:
>> Perhaps you could include some conversion with sox somewhere in your
>> scripts ?
>>
>> David Ponzone ?Direction Technique
>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>> tel: ? ? ?01 74 03 18 97
>> gsm: ? 06 66 98 76 34
>>
>> Service Client IPeva
>> tel: ? ? ?0811 46 26 26
>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/> ?- ? www.ipeva-studio.com
>> <BLOCKED::http://www.ipeva-studio.com/>
>>
>> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou
>> diffusion non autoris?e est interdite. Tout message ?lectronique est
>> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au
>> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous
>> n'?tes pas destinataire de ce message, merci de le d?truire
>> imm?diatement et d'avertir l'exp?diteur./
>> /
>> /
>>
>>
>>
>> Le 22/10/2010 ? 18:04, Mario a ?crit :
>>
>>> Is there a way to change the bitrate of the wav email message? iPhone is
>>> picky about wav files. I got mp3 to work from Linux but it's impossible
>>> to build mp3 support on osX due to libvorbis problems. Thanks.
>>> Mario
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From msc at freeswitch.org  Fri Oct 22 09:42:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 22 Oct 2010 09:42:07 -0700
Subject: [Freeswitch-users] Accessing fax variables/status after
	rxfax/txfax
In-Reply-To: <AANLkTi=DUnXPcFHLJfmAawfzahdM4S_A=3muy-me7wys@mail.gmail.com>
References: <AANLkTimSPnc9qR3X+WxGaDKPKTDdLn_Y=MiS29x+L12G@mail.gmail.com>
	<AANLkTi=vXgXV7E4GAKEPUTYEEqYaCTJPks8=SOsikoaB@mail.gmail.com>
	<AANLkTi=DUnXPcFHLJfmAawfzahdM4S_A=3muy-me7wys@mail.gmail.com>
Message-ID: <AANLkTimBQ1P1L_oBmSbhC25w_1LtK-cLG-BT3-exSRai@mail.gmail.com>

On Thu, Oct 21, 2010 at 8:57 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Dang, that's a big page! Did I really make that many vars =0
>
> Yeah, and we we're still finding vars we haven't documented yet. :P
-MC
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From mario_fs at mgtech.com  Fri Oct 22 09:59:33 2010
From: mario_fs at mgtech.com (Mario)
Date: Fri, 22 Oct 2010 09:59:33 -0700
Subject: [Freeswitch-users] Can change email wav encoding?
In-Reply-To: <AANLkTimOk0rsTa10gVsw0uqsNAtenU0dnB7JtRVFXrxM@mail.gmail.com>
References: <4CC1B606.6030701@mgtech.com>	<4EA769AD-0583-499E-9A1E-C00F4E491429@ipeva.fr>	<4CC1BB9B.7030406@mgtech.com>
	<AANLkTimOk0rsTa10gVsw0uqsNAtenU0dnB7JtRVFXrxM@mail.gmail.com>
Message-ID: <4CC1C2F5.4060500@mgtech.com>

I found doc on mod_sndfile and changed it to aiff and it works! I will
post this in the wiki (I am documenting everthing for others). Thanks
for asking,. BTW, iPhone supports WAV but apparently not all bitrates.
AIFF works fine though. Thanks for the response!
Mario

On 10/22/10 09:36, Anthony Minessale wrote:
> what format does it want?
> 
> 
> On Fri, Oct 22, 2010 at 11:28 AM, Mario <mario_fs at mgtech.com> wrote:
>> I am trying to avoid having to add more tools to osx (spent 50 times
>> longer on libvorbis install than FS and it still wont work) so it would
>> be much simpler to have the wav come out of FS in the format needed for
>> iPhone email.
>>
>> On 10/22/10 09:15, David Ponzone wrote:
>>> Perhaps you could include some conversion with sox somewhere in your
>>> scripts ?
>>>
>>> David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com
>>> <BLOCKED::http://www.ipeva-studio.com/>
>>>
>>> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou
>>> diffusion non autoris?e est interdite. Tout message ?lectronique est
>>> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au
>>> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous
>>> n'?tes pas destinataire de ce message, merci de le d?truire
>>> imm?diatement et d'avertir l'exp?diteur./
>>> /
>>> /
>>>
>>>
>>>
>>> Le 22/10/2010 ? 18:04, Mario a ?crit :
>>>
>>>> Is there a way to change the bitrate of the wav email message? iPhone is
>>>> picky about wav files. I got mp3 to work from Linux but it's impossible
>>>> to build mp3 support on osX due to libvorbis problems. Thanks.
>>>> Mario
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 


From msc at freeswitch.org  Fri Oct 22 10:39:29 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 22 Oct 2010 10:39:29 -0700
Subject: [Freeswitch-users] Music On Hold
In-Reply-To: <AANLkTin_qHEUnoFUo1jkX1xyWyyRiYe7sTE1P4czayUh@mail.gmail.com>
References: <AANLkTin_qHEUnoFUo1jkX1xyWyyRiYe7sTE1P4czayUh@mail.gmail.com>
Message-ID: <AANLkTikeUuj8XHz2d9w3niXSO=wpptcQ-KSpdCqBefg1@mail.gmail.com>

Just x-fer the call to dp ext 9664. When you are done with the Lua tasks use
uuid_transfer (or uuid_bridge) to grab the call out of 9664 and send it
wherever you need it to go.

-MC

On Fri, Oct 22, 2010 at 12:19 AM, Chia-Yen Wu <teddywu72 at gmail.com> wrote:

> Hello,
>
> I am using freeswitch as an IVR server.
> I would like to be able to put the call on hold and play music while
> performing some other tasks in the lua.
> when the task is complete , the music stop and the IVR continue
>
> but i cant find a way to do that, is there any function can do "music on
> hold" in dialplan ?
>
> Thank you
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Fri Oct 22 10:40:35 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 22 Oct 2010 12:40:35 -0500
Subject: [Freeswitch-users] Can change email wav encoding?
In-Reply-To: <4CC1C2F5.4060500@mgtech.com>
References: <4CC1B606.6030701@mgtech.com>
	<4EA769AD-0583-499E-9A1E-C00F4E491429@ipeva.fr>
	<4CC1BB9B.7030406@mgtech.com>
	<AANLkTimOk0rsTa10gVsw0uqsNAtenU0dnB7JtRVFXrxM@mail.gmail.com>
	<4CC1C2F5.4060500@mgtech.com>
Message-ID: <AANLkTikOn68QhNjjOb2n2W=KX1_N9T8HjN1rqwULyfPx@mail.gmail.com>

I think you mean sample rate.

This is probably what you wanted in voicemail.conf.xml but aiff is fine too.
<param name="record-sample-rate" value="11025"/>


Also for what it's worth, I added a patch to the build process for
mod_shout to hack out ogg completely from existence so if you do this
from the build root:

rm -fr libs/libshout-2.2.2
git pull
make mod_shout-install

that should work on mac and anywhere ells without ogg/vorbis now


On Fri, Oct 22, 2010 at 11:59 AM, Mario <mario_fs at mgtech.com> wrote:
> I found doc on mod_sndfile and changed it to aiff and it works! I will
> post this in the wiki (I am documenting everthing for others). Thanks
> for asking,. BTW, iPhone supports WAV but apparently not all bitrates.
> AIFF works fine though. Thanks for the response!
> Mario
>
> On 10/22/10 09:36, Anthony Minessale wrote:
>> what format does it want?
>>
>>
>> On Fri, Oct 22, 2010 at 11:28 AM, Mario <mario_fs at mgtech.com> wrote:
>>> I am trying to avoid having to add more tools to osx (spent 50 times
>>> longer on libvorbis install than FS and it still wont work) so it would
>>> be much simpler to have the wav come out of FS in the format needed for
>>> iPhone email.
>>>
>>> On 10/22/10 09:15, David Ponzone wrote:
>>>> Perhaps you could include some conversion with sox somewhere in your
>>>> scripts ?
>>>>
>>>> David Ponzone ?Direction Technique
>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>> tel: ? ? ?01 74 03 18 97
>>>> gsm: ? 06 66 98 76 34
>>>>
>>>> Service Client IPeva
>>>> tel: ? ? ?0811 46 26 26
>>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/> ?- ? www.ipeva-studio.com
>>>> <BLOCKED::http://www.ipeva-studio.com/>
>>>>
>>>> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>> l'intention exclusive de ses destinataires. Toute utilisation ou
>>>> diffusion non autoris?e est interdite. Tout message ?lectronique est
>>>> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au
>>>> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous
>>>> n'?tes pas destinataire de ce message, merci de le d?truire
>>>> imm?diatement et d'avertir l'exp?diteur./
>>>> /
>>>> /
>>>>
>>>>
>>>>
>>>> Le 22/10/2010 ? 18:04, Mario a ?crit :
>>>>
>>>>> Is there a way to change the bitrate of the wav email message? iPhone is
>>>>> picky about wav files. I got mp3 to work from Linux but it's impossible
>>>>> to build mp3 support on osX due to libvorbis problems. Thanks.
>>>>> Mario
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From msc at freeswitch.org  Fri Oct 22 10:43:06 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 22 Oct 2010 10:43:06 -0700
Subject: [Freeswitch-users] [Linksys 3102] Couple of issues using FS and
	XLite
In-Reply-To: <1287759334654-5662810.post@n2.nabble.com>
References: <1287755399019-5662612.post@n2.nabble.com>
	<AANLkTin2q7o1DHp0tUguHiVMr1aJxLo4ryoedAHgW18_@mail.gmail.com>
	<1287759334654-5662810.post@n2.nabble.com>
Message-ID: <AANLkTi=R6MHHUWHpx64MM4K3waTZjMF3Cfe+4goo5qnH@mail.gmail.com>

On Fri, Oct 22, 2010 at 7:55 AM, GillesToo <codecomplete at free.fr> wrote:

>
> Thanks for the help.
>
> I use the default settings as provided in the binary for Windows (1.0.4).
> XLite uses extension 1001, and the Linksys uses extension 1000.
>
1.0.4 is old and crusty. Literally hundreds of bugs have been fixed since
then. I would update to latest. You have a few choices: get MSVCEE (or MS
Visual Studio) and compile it yourself or wait for Jeff Lenk (irc: jlenk) to
get the weekly Windows binaries uploaded to the files site. (This is
happening within a week or two I think.)

1.0.4 had some serious issues IIRC so don't spend much more time banging
your head.
-MC
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From renjian at gmail.com  Fri Oct 22 08:33:02 2010
From: renjian at gmail.com (Jian Ren)
Date: Fri, 22 Oct 2010 11:33:02 -0400
Subject: [Freeswitch-users] Help needed about building snd_dummy with
 the dummy.c in skypopen
In-Reply-To: <AANLkTik5BEvEi=P_+cEL2kwjfrTQYLGh7MNtF14-8ROE@mail.gmail.com>
References: <AANLkTikQ_vj3Ue1MYs7x1pVj8kiCgFbeKBTL=97PddX=@mail.gmail.com>
	<AANLkTik5BEvEi=P_+cEL2kwjfrTQYLGh7MNtF14-8ROE@mail.gmail.com>
Message-ID: <AANLkTikkXGL7H=1Sfhnzub454ob4MeCe7asoxXJi9myT@mail.gmail.com>

Tried under VMPlayer, got the same problem, will try real machine later.
Attached is the screen shot.

Thanks!
Jian

On Thu, Oct 21, 2010 at 4:45 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> Maybe is a problem with virtualbox. Also, is known that
> fs+mod_skypopen+skype clients does not works in virtualbox.
>
> Try it on a real (hardware) machine.
>
> Or (but is less popular) in a xen like virtual machine.
>
> -giovanni
>
> On Thu, Oct 21, 2010 at 4:21 PM, Jian Ren <renjian at gmail.com> wrote:
> > Hi,
> > I am trying to follow this wiki:
> > http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
> > to setup FS and Skypopen on Ubuntu 8.04(64bit server inside virualbox).
> >
> > There is one step asks me to build snd_dummy:
> >
> http://wiki.freeswitch.org/wiki/Skypopen_Skype_Endpoint_and_Trunk#ALSA_and_the_custom_snd-dummy
> > The section is like below:
> >
> > Ubuntu, Debian
> >
> > Note, Ubuntu may have alsa sound drivers installed at /lib/modules/<your
> > kernel version>/ubuntu/sound/..., you may need to remove the dir to allow
> > modprobe search from the default place: /lib/modules/<your kernel
> > verision>/kernel/sound/ (don't forget to rerun /sbin/depmod after
> removing
> > the old sound directory. Eg: /sbin/depmod -a 2.6.24-24-server).
> >
> > ./configure --with-redhat=no \
> > --with-cards=dummy,usb-audio,hda-intel,hrtimer,rtctimer \
> > --with-card-options=all
> > make && make install
> >
> > After the first make && make install, copy
> mod_skypopen/configs/alsa/dummy.c
> > to alsa-driver-1.0.20sound/drivers/dummy.c .
> >
> > make && make install #again :)
> >
> > The problem is after I did all of them, when I entered modprobe
> snd_dummy,
> > it always crashed and returned as "killed" on my terminal, I could see it
> > dumped out a huge block or data(numbers, etc) on the host but don't
> > understand the meaning. While if I used the original dummy.c inside
> > alsa-driver-1.0.20, it worked fine.
> > Besides, the default ubuntu installation doesn't include kernal dev and
> > source, so I did one more step(or it cannot build alsa).
> >
> > Did anyone try the same and get it working?
> >
> > Thanks!
> > Jian
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mario_fs at mgtech.com  Fri Oct 22 11:19:53 2010
From: mario_fs at mgtech.com (Mario)
Date: Fri, 22 Oct 2010 11:19:53 -0700
Subject: [Freeswitch-users] Can change email wav encoding?
In-Reply-To: <AANLkTikOn68QhNjjOb2n2W=KX1_N9T8HjN1rqwULyfPx@mail.gmail.com>
References: <4CC1B606.6030701@mgtech.com>	<4EA769AD-0583-499E-9A1E-C00F4E491429@ipeva.fr>	<4CC1BB9B.7030406@mgtech.com>	<AANLkTimOk0rsTa10gVsw0uqsNAtenU0dnB7JtRVFXrxM@mail.gmail.com>	<4CC1C2F5.4060500@mgtech.com>
	<AANLkTikOn68QhNjjOb2n2W=KX1_N9T8HjN1rqwULyfPx@mail.gmail.com>
Message-ID: <4CC1D5C9.5010707@mgtech.com>

Thanks, great!

On 10/22/10 10:40, Anthony Minessale wrote:
> I think you mean sample rate.
> 
> This is probably what you wanted in voicemail.conf.xml but aiff is fine too.
> <param name="record-sample-rate" value="11025"/>
> 
> 
> Also for what it's worth, I added a patch to the build process for
> mod_shout to hack out ogg completely from existence so if you do this
> from the build root:
> 
> rm -fr libs/libshout-2.2.2
> git pull
> make mod_shout-install
> 
> that should work on mac and anywhere ells without ogg/vorbis now
> 
> 
> On Fri, Oct 22, 2010 at 11:59 AM, Mario <mario_fs at mgtech.com> wrote:
>> I found doc on mod_sndfile and changed it to aiff and it works! I will
>> post this in the wiki (I am documenting everthing for others). Thanks
>> for asking,. BTW, iPhone supports WAV but apparently not all bitrates.
>> AIFF works fine though. Thanks for the response!
>> Mario
>>
>> On 10/22/10 09:36, Anthony Minessale wrote:
>>> what format does it want?
>>>
>>>
>>> On Fri, Oct 22, 2010 at 11:28 AM, Mario <mario_fs at mgtech.com> wrote:
>>>> I am trying to avoid having to add more tools to osx (spent 50 times
>>>> longer on libvorbis install than FS and it still wont work) so it would
>>>> be much simpler to have the wav come out of FS in the format needed for
>>>> iPhone email.
>>>>
>>>> On 10/22/10 09:15, David Ponzone wrote:
>>>>> Perhaps you could include some conversion with sox somewhere in your
>>>>> scripts ?
>>>>>
>>>>> David Ponzone  Direction Technique
>>>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>>> tel:      01 74 03 18 97
>>>>> gsm:   06 66 98 76 34
>>>>>
>>>>> Service Client IPeva
>>>>> tel:      0811 46 26 26
>>>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com
>>>>> <BLOCKED::http://www.ipeva-studio.com/>
>>>>>
>>>>> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>>> l'intention exclusive de ses destinataires. Toute utilisation ou
>>>>> diffusion non autoris?e est interdite. Tout message ?lectronique est
>>>>> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au
>>>>> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous
>>>>> n'?tes pas destinataire de ce message, merci de le d?truire
>>>>> imm?diatement et d'avertir l'exp?diteur./
>>>>> /
>>>>> /
>>>>>
>>>>>
>>>>>
>>>>> Le 22/10/2010 ? 18:04, Mario a ?crit :
>>>>>
>>>>>> Is there a way to change the bitrate of the wav email message? iPhone is
>>>>>> picky about wav files. I got mp3 to work from Linux but it's impossible
>>>>>> to build mp3 support on osX due to libvorbis problems. Thanks.
>>>>>> Mario
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 


From victor.chukalovskiy at utoronto.ca  Fri Oct 22 11:27:22 2010
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Fri, 22 Oct 2010 14:27:22 -0400
Subject: [Freeswitch-users] FS does not honour sip-force-contact. Advice?
In-Reply-To: <AANLkTikqL-+_S5+rcgMTvNbLkq_v23O5pjgZpPKoY0T9@mail.gmail.com>
References: <4CC05DC3.2080109@utoronto.ca>
	<AANLkTikqL-+_S5+rcgMTvNbLkq_v23O5pjgZpPKoY0T9@mail.gmail.com>
Message-ID: <4CC1D78A.4010800@utoronto.ca>

  Anthony,
Thank you. Disabling timers worked out.
Since session timers is a per-profile setting, what do we loose by 
disabling them?
Thank you,
Victor

On 21/10/10 03:56 PM, Anthony Minessale wrote:
> disable session timers on both sides of the call, they are interfering
> with the dialog once it's established.
>
>
> On Thu, Oct 21, 2010 at 10:35 AM, Victor Chukalovskiy
> <victor.chukalovskiy at utoronto.ca>  wrote:
>> Dear Gurus,
>>
>> We have a user behind NAT.
>> Currently we make him work by using
>>        <variable name="sip-force-contact"
>> value="sip:1234 at 171.134.62.106:5060"/>
>> It works since users IP is static and there is a port forwarding setup for
>> port 5060 on the users side.
>>
>> The problem is that after 15 minutes of a call FreeSWITCH attempts re-INVITE
>> (RFC 4028 I guess) and does not honour port specified above.
>> Instead, it uses some 40,000-ish UDP destination port.  Obviously, this
>> never reaches users phone and FS hangups with "expiry on timer" cause.
>>
>> Is the syntax of sip-force-contact above valid?
>>
>> Is it normal for re-INVITEs no to honour sip-force-contact?
>>
>> I'd try NDLB-tls-connectile-dysfunction but I need clarification:
>> -will it be honoured by re-INVITEs?
>> -will it default to port 5060 or should I specify the port?
>>
>> Thank you,
>> Victor
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>



From anthony.minessale at gmail.com  Fri Oct 22 12:10:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 22 Oct 2010 14:10:49 -0500
Subject: [Freeswitch-users] FS does not honour sip-force-contact. Advice?
In-Reply-To: <4CC1D78A.4010800@utoronto.ca>
References: <4CC05DC3.2080109@utoronto.ca>
	<AANLkTikqL-+_S5+rcgMTvNbLkq_v23O5pjgZpPKoY0T9@mail.gmail.com>
	<4CC1D78A.4010800@utoronto.ca>
Message-ID: <AANLkTinMr9qxo_x6YVaMK0LvjqfY6yY249KqjNHk8CAr@mail.gmail.com>

the session timers are used to make sure a call is still alive by
sending a periodic re-invite to make sure the server is still there.
If the devices you are interacting with can't do it properly there is
not much to lose from disabling it.


On Fri, Oct 22, 2010 at 1:27 PM, Victor Chukalovskiy
<victor.chukalovskiy at utoronto.ca> wrote:
> ?Anthony,
> Thank you. Disabling timers worked out.
> Since session timers is a per-profile setting, what do we loose by
> disabling them?
> Thank you,
> Victor
>
> On 21/10/10 03:56 PM, Anthony Minessale wrote:
>> disable session timers on both sides of the call, they are interfering
>> with the dialog once it's established.
>>
>>
>> On Thu, Oct 21, 2010 at 10:35 AM, Victor Chukalovskiy
>> <victor.chukalovskiy at utoronto.ca> ?wrote:
>>> Dear Gurus,
>>>
>>> We have a user behind NAT.
>>> Currently we make him work by using
>>> ? ? ? ?<variable name="sip-force-contact"
>>> value="sip:1234 at 171.134.62.106:5060"/>
>>> It works since users IP is static and there is a port forwarding setup for
>>> port 5060 on the users side.
>>>
>>> The problem is that after 15 minutes of a call FreeSWITCH attempts re-INVITE
>>> (RFC 4028 I guess) and does not honour port specified above.
>>> Instead, it uses some 40,000-ish UDP destination port. ?Obviously, this
>>> never reaches users phone and FS hangups with "expiry on timer" cause.
>>>
>>> Is the syntax of sip-force-contact above valid?
>>>
>>> Is it normal for re-INVITEs no to honour sip-force-contact?
>>>
>>> I'd try NDLB-tls-connectile-dysfunction but I need clarification:
>>> -will it be honoured by re-INVITEs?
>>> -will it default to port 5060 or should I specify the port?
>>>
>>> Thank you,
>>> Victor
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From juanito1982 at gmail.com  Fri Oct 22 12:42:00 2010
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Fri, 22 Oct 2010 21:42:00 +0200
Subject: [Freeswitch-users] FreeTDM error
In-Reply-To: <4CC195EA.90404@puzzled.xs4all.nl>
References: <20101022130502.D96A23F3240@cuneorg-email.cune.pri>
	<4CC195EA.90404@puzzled.xs4all.nl>
Message-ID: <AANLkTim_GzryUfxX2eBXTgTLeZVBSsgZScuym__x4=99@mail.gmail.com>

It was solver by one Sangoma support techs using a newer version that I
downloade from Sangoma wiki.
It seems to work ok but when I set up one call I always get a fack ringtone
before real one although the callee is busy. Is there any way to disable it?

Regards

2010/10/22 Patrick Lists <freeswitch-list at puzzled.xs4all.nl>

> On 10/22/2010 03:05 PM, Russell.Mosemann at cune.org wrote:
> > Everything you have provided so far looks correct. Is FS running chroot?
> > If capabilities are not turned off, temporarily turn them off and see if
> > FS will run. Are there possibly any ACLs attached to the
> > files/directories that would prevent root from accessing the library? Are
> > there any helpful error messages in any of the system logs?
>
> Another suggestion: how about selinux? Is it active? If yes, check with
> ls -Z /usr/lib64/libsng_isdn.so.* if it has the proper security context.
> Should be something like: system_u:object_r:lib_t:s0
> To set/restore the proper security context read man restorecon.
>
> Regards,
> Patrick
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From pjintheusa at gmail.com  Fri Oct 22 12:58:14 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 22 Oct 2010 15:58:14 -0400
Subject: [Freeswitch-users] Using the bridge | operator for Implementing
	Failover
In-Reply-To: <AANLkTimoE7t0LQwK3Nmv8Km-KSHHAtk+n-Teeiv7zhoQ@mail.gmail.com>
References: <AANLkTimZonOAnpKJKyy24ZnyJD9HnDPYMju40MYL6CvH@mail.gmail.com>
	<AANLkTimoE7t0LQwK3Nmv8Km-KSHHAtk+n-Teeiv7zhoQ@mail.gmail.com>
Message-ID: <AANLkTimfXA70zUQKK0cBoKyvjE2OhfaaxckTPMgpYwHu@mail.gmail.com>

Anthony - thanks for the response.

<include>
  <extension name="public_did">
    <condition field="destination_number" expression="^(3478309992)$">
      <action application="answer"/>
      <action application="sleep" data="2000"/>
      <action application="set"
data="fail_on_single_reject=NO_ANSWER,NORMAL_CLEARING,ALLOTTED_TIMEOUT"/>
      <action application="bridge"
data="{ignore_early_media=true}[leg_timeout=10]sofia/gateway/broadvox1/2158824374|[leg_timeout=25]sofia/gateway/broadvox2/2158824374"/>
    </condition>
  </extension>
</include>


I get ALLOTTED_TIMEOUT as expected from the first attempt at
calling 2158824374, but it is just called again through the 2nd broadvox
gateway. It does not fail as expected.

I am on todays GIT.

I do note that in the WIKI and code comments - it only mentions
fail_on_single_reject in connection the ',' AND operator not '|' pipe
operator.

Can you see what I am doing wrong here?

Thanks,


Pj


On Thu, Oct 21, 2010 at 11:55 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> put this before the bridge call
>
> <!-- add as many , sep cause codes which qualify to stop trying the
> pipe sep list -->
> <action application="set" data="fail_on_single_reject=no_answer"/>
>
>
> On Thu, Oct 21, 2010 at 8:23 PM, Phillip Jones <pjintheusa at gmail.com>
> wrote:
> > Hi there,
> > From the WIKI
> > Implementing Failover
> >
> > Failover for your outbound gateway is easy to implement at bridge time
> using
> > the | separator:
> >
> > <action application="bridge"
> >
> data="sofia/gateway/primary/dialstring|sofia/gateway/secondary/dialstring"/>
> >
> >
> > In the above example though, if the user does not pick the call, that
> bridge
> > will try again using the secondary gateway, which is not desired. The
> second
> > attempt should only be made in there is a problem with the gateway.
> >
> > How do you control that? .
> >
> >
> > Thanks
> >
> >
> > Pj
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From covici at ccs.covici.com  Fri Oct 22 13:20:34 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 22 Oct 2010 16:20:34 -0400
Subject: [Freeswitch-users] Can change email wav encoding?
In-Reply-To: <4CC1B606.6030701@mgtech.com>
References: <4CC1B606.6030701@mgtech.com>
Message-ID: <2540.1287778834@ccs.covici.com>

I have lame compiled on 10.6.4, if that helps any.

Mario <mario_fs at mgtech.com> wrote:

> Is there a way to change the bitrate of the wav email message? iPhone is
> picky about wav files. I got mp3 to work from Linux but it's impossible
> to build mp3 support on osX due to libvorbis problems. Thanks.
> Mario
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From Peter.Hinman at ParcelPool.com  Fri Oct 22 13:26:00 2010
From: Peter.Hinman at ParcelPool.com (Peter Hinman)
Date: Fri, 22 Oct 2010 14:26:00 -0600
Subject: [Freeswitch-users] ODBC and MSSQL
In-Reply-To: <1287760389322-5662887.post@n2.nabble.com>
References: <4CAB9F5A.2090503@ParcelPool.com>	<1286331931285-5605569.post@n2.nabble.com>	<4CACE7C5.6060208@ParcelPool.com>	<1286459189732-5611425.post@n2.nabble.com>	<4CC1A535.9010706@ParcelPool.com>
	<1287760389322-5662887.post@n2.nabble.com>
Message-ID: <4CC1F358.8080708@ParcelPool.com>

:) I must have looked at the Jira yesterday before you made the commit.  
I'll try the current git.

Thanks again.

Peter


On 10/22/2010 9:13 AM, Jeff Lenk wrote:
> Thanks Peter,
>
> btw that patch was committed yesterday to git with some minor modifications
> which do not change any functionality(MSSQL). It also preserves previous
> table compatibility for non MSSQL based instances of fs.
>
> -Jeff




From msc at freeswitch.org  Fri Oct 22 13:40:44 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 22 Oct 2010 13:40:44 -0700
Subject: [Freeswitch-users] dialogic FXO card will support freeSWITCH
In-Reply-To: <AANLkTimfFw0pLcivCh+PLkY6kEBEZ25_KRK4-OmXacyT@mail.gmail.com>
References: <AANLkTimfFw0pLcivCh+PLkY6kEBEZ25_KRK4-OmXacyT@mail.gmail.com>
Message-ID: <AANLkTi=zao8buWwvv3TDAPaP5sOPpc30_ik8VSG9ygyz@mail.gmail.com>

As far as I know, only zaptel/dahdi compatible cards and Sangoma cards are
compatible. (Khomp is coming soon I believe.) I've never seen or heard of a
Dialogic card working with FreeSWITCH.

-MC

On Fri, Oct 22, 2010 at 9:08 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:

> Hi.
>
> Can you anyone plz tell me, whether Dialogic FXO card will support
> freeswitch?
>
>
> thanks,
>
> Regards,
> Venkat.
>
> --
>
> If you have come to help me, you are wasting your time.
> If you have come to because your liberation is bound up in mine, we can
> work together.
>
>
> Regards
> Venkatesan OV.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Fri Oct 22 13:42:22 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 22 Oct 2010 15:42:22 -0500
Subject: [Freeswitch-users] Using the bridge | operator for Implementing
	Failover
In-Reply-To: <AANLkTimfXA70zUQKK0cBoKyvjE2OhfaaxckTPMgpYwHu@mail.gmail.com>
References: <AANLkTimZonOAnpKJKyy24ZnyJD9HnDPYMju40MYL6CvH@mail.gmail.com>
	<AANLkTimoE7t0LQwK3Nmv8Km-KSHHAtk+n-Teeiv7zhoQ@mail.gmail.com>
	<AANLkTimfXA70zUQKK0cBoKyvjE2OhfaaxckTPMgpYwHu@mail.gmail.com>
Message-ID: <AANLkTimT0QTVcJKkczFnuwQJMBmgdfjuG7vr6UxxAx0k@mail.gmail.com>

also add {originate_continue_on_timeout=false}


On Fri, Oct 22, 2010 at 2:58 PM, Phillip Jones <pjintheusa at gmail.com> wrote:
> Anthony - thanks for the response.
>
> <include>
> ? <extension name="public_did">
> ??? <condition field="destination_number" expression="^(3478309992)$">
> ????? <action application="answer"/>
> ????? <action application="sleep" data="2000"/>
> ?? ?? <action application="set"
> data="fail_on_single_reject=NO_ANSWER,NORMAL_CLEARING,ALLOTTED_TIMEOUT"/>
> ???? ?<action application="bridge"
> data="{ignore_early_media=true}[leg_timeout=10]sofia/gateway/broadvox1/2158824374|[leg_timeout=25]sofia/gateway/broadvox2/2158824374"/>
> ??? </condition>
> ? </extension>
> </include>
>
> I get ALLOTTED_TIMEOUT as expected?from the first attempt?at
> calling?2158824374, but it?is just called again through the 2nd broadvox
> gateway. It does not fail as expected.
>
> I am on todays GIT.
>
> I do note that in the WIKI and code comments - it only mentions
> fail_on_single_reject in connection the ',' AND operator not '|' pipe
> operator.
>
> Can you see what I am doing wrong here?
>
> Thanks,
>
>
> Pj
>
>
> On Thu, Oct 21, 2010 at 11:55 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> put this before the bridge call
>>
>> <!-- add as many , sep cause codes which qualify to stop trying the
>> pipe sep list -->
>> <action application="set" data="fail_on_single_reject=no_answer"/>
>>
>>
>> On Thu, Oct 21, 2010 at 8:23 PM, Phillip Jones <pjintheusa at gmail.com>
>> wrote:
>> > Hi there,
>> > From the WIKI
>> > Implementing Failover
>> >
>> > Failover for your outbound gateway is easy to implement at bridge time
>> > using
>> > the | separator:
>> >
>> > <action application="bridge"
>> >
>> > data="sofia/gateway/primary/dialstring|sofia/gateway/secondary/dialstring"/>
>> >
>> >
>> > In the above example though, if the user does not pick the call, that
>> > bridge
>> > will try again using?the secondary gateway, which is not desired. The
>> > second
>> > attempt should only be made in there is a problem with the gateway.
>> >
>> > How do you control that? .
>> >
>> >
>> > Thanks
>> >
>> >
>> > Pj
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From pjintheusa at gmail.com  Fri Oct 22 14:43:01 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 22 Oct 2010 17:43:01 -0400
Subject: [Freeswitch-users] Using the bridge | operator for Implementing
	Failover
In-Reply-To: <AANLkTimT0QTVcJKkczFnuwQJMBmgdfjuG7vr6UxxAx0k@mail.gmail.com>
References: <AANLkTimZonOAnpKJKyy24ZnyJD9HnDPYMju40MYL6CvH@mail.gmail.com>
	<AANLkTimoE7t0LQwK3Nmv8Km-KSHHAtk+n-Teeiv7zhoQ@mail.gmail.com>
	<AANLkTimfXA70zUQKK0cBoKyvjE2OhfaaxckTPMgpYwHu@mail.gmail.com>
	<AANLkTimT0QTVcJKkczFnuwQJMBmgdfjuG7vr6UxxAx0k@mail.gmail.com>
Message-ID: <AANLkTikomdMk_s3GovN_gkxm6u2U-_p3Y67Jw6bHLLrt@mail.gmail.com>

Thanks.

That didn't seem to work however.

<extension name="public_did">
    <condition field="destination_number" expression="^(3478309992)$">
      <action application="answer"/>
      <action application="sleep" data="2000"/>
      <action application="set"
data="fail_on_single_reject=NO_ANSWER,NORMAL_CLEARING,ALLOTTED_TIMEOUT"/>
      <action application="bridge"
data="{ignore_early_media=true,originate_continue_on_timeout=false}[leg_timeout=10]sofia/gateway/broadvox1/2158824374|[leg_timeout=25]sofia/gateway/broadvox2/2158824374"/>
    </condition>
 </extension>

2158824374 is called from each gateway.

I also tried, originate_continue_on_timeout=true since the WIKI says the
default is false.


Any other ideas?


Thanks!

Pj




On Fri, Oct 22, 2010 at 4:42 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> also add {originate_continue_on_timeout=false}
>
>
> On Fri, Oct 22, 2010 at 2:58 PM, Phillip Jones <pjintheusa at gmail.com>
> wrote:
> > Anthony - thanks for the response.
> >
> > <include>
> >   <extension name="public_did">
> >     <condition field="destination_number" expression="^(3478309992)$">
> >       <action application="answer"/>
> >       <action application="sleep" data="2000"/>
> >       <action application="set"
> > data="fail_on_single_reject=NO_ANSWER,NORMAL_CLEARING,ALLOTTED_TIMEOUT"/>
> >       <action application="bridge"
> >
> data="{ignore_early_media=true}[leg_timeout=10]sofia/gateway/broadvox1/2158824374|[leg_timeout=25]sofia/gateway/broadvox2/2158824374"/>
> >     </condition>
> >   </extension>
> > </include>
> >
> > I get ALLOTTED_TIMEOUT as expected from the first attempt at
> > calling 2158824374, but it is just called again through the 2nd broadvox
> > gateway. It does not fail as expected.
> >
> > I am on todays GIT.
> >
> > I do note that in the WIKI and code comments - it only mentions
> > fail_on_single_reject in connection the ',' AND operator not '|' pipe
> > operator.
> >
> > Can you see what I am doing wrong here?
> >
> > Thanks,
> >
> >
> > Pj
> >
> >
> > On Thu, Oct 21, 2010 at 11:55 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> put this before the bridge call
> >>
> >> <!-- add as many , sep cause codes which qualify to stop trying the
> >> pipe sep list -->
> >> <action application="set" data="fail_on_single_reject=no_answer"/>
> >>
> >>
> >> On Thu, Oct 21, 2010 at 8:23 PM, Phillip Jones <pjintheusa at gmail.com>
> >> wrote:
> >> > Hi there,
> >> > From the WIKI
> >> > Implementing Failover
> >> >
> >> > Failover for your outbound gateway is easy to implement at bridge time
> >> > using
> >> > the | separator:
> >> >
> >> > <action application="bridge"
> >> >
> >> >
> data="sofia/gateway/primary/dialstring|sofia/gateway/secondary/dialstring"/>
> >> >
> >> >
> >> > In the above example though, if the user does not pick the call, that
> >> > bridge
> >> > will try again using the secondary gateway, which is not desired. The
> >> > second
> >> > attempt should only be made in there is a problem with the gateway.
> >> >
> >> > How do you control that? .
> >> >
> >> >
> >> > Thanks
> >> >
> >> >
> >> > Pj
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From curriegrad2004 at gmail.com  Fri Oct 22 16:11:27 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Fri, 22 Oct 2010 16:11:27 -0700
Subject: [Freeswitch-users] [Linksys 3102] Couple of issues using FS and
	XLite
In-Reply-To: <AANLkTi=R6MHHUWHpx64MM4K3waTZjMF3Cfe+4goo5qnH@mail.gmail.com>
References: <1287755399019-5662612.post@n2.nabble.com>
	<AANLkTin2q7o1DHp0tUguHiVMr1aJxLo4ryoedAHgW18_@mail.gmail.com>
	<1287759334654-5662810.post@n2.nabble.com>
	<AANLkTi=R6MHHUWHpx64MM4K3waTZjMF3Cfe+4goo5qnH@mail.gmail.com>
Message-ID: <AANLkTimkLpGj7i3GcsYP8LZ8FnABTvz+Bmjb_JOqmkwB@mail.gmail.com>

For anyone that's interested, I just built the FreeSwitch binaries for
Windows using the latest git tree (as of now that is).

Happy VoIPing!

http://www.megaupload.com/?d=TW9O0HOD

On Fri, Oct 22, 2010 at 10:43 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Fri, Oct 22, 2010 at 7:55 AM, GillesToo <codecomplete at free.fr> wrote:
>>
>> Thanks for the help.
>>
>> I use the default settings as provided in the binary for Windows (1.0.4).
>> XLite uses extension 1001, and the Linksys uses extension 1000.
>
> 1.0.4 is old and crusty. Literally hundreds of bugs have been fixed since
> then. I would update to latest. You have a few choices: get MSVCEE (or MS
> Visual Studio) and compile it yourself or wait for Jeff Lenk (irc: jlenk) to
> get the weekly Windows binaries uploaded to the files site. (This is
> happening within a week or two I think.)
>
> 1.0.4 had some serious issues IIRC so don't spend much more time banging
> your head.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From moises.silva at gmail.com  Fri Oct 22 16:52:43 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 22 Oct 2010 19:52:43 -0400
Subject: [Freeswitch-users] Call for wiki help: cleanup OpenZAP pages on
	wiki
In-Reply-To: <AANLkTiks-kt_wqopLnwbYkX0qf2_btPAJj=6LkCuFPG1@mail.gmail.com>
References: <AANLkTiks-kt_wqopLnwbYkX0qf2_btPAJj=6LkCuFPG1@mail.gmail.com>
Message-ID: <AANLkTinqb2Nkgq594n_6A+ZGuieCsxHb-_58ap9JGgPq@mail.gmail.com>

On Wed, Oct 20, 2010 at 2:50 PM, Michael Collins <msc at freeswitch.org> wrote:
> We need some assistance with the migration from OpenZAP to FreeTDM. In
> particular we have some legacy OpenZAP pages that need to be translated into
> FreeTDM pages on the wiki. In some cases it may be as simple as swapping
> "OpenZAP" with "FreeTDM" and doing a page redirect. In other cases it may be
> a bit more involved. I am inviting anyone with OpenZAP and FreeTDM
> experience to work on these wiki pages:
>
> http://wiki.freeswitch.org/wiki/Openzap.conf_Examples
> http://wiki.freeswitch.org/wiki/Openzap.conf.xml_Examples
> http://wiki.freeswitch.org/wiki/Configuration_OpenZAP-_DigiumTDM400P_Example
> http://wiki.freeswitch.org/wiki/Openzap.sangoma.libpri
> http://wiki.freeswitch.org/wiki/OpenZap_Dahdi
> http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2
> http://wiki.freeswitch.org/wiki/OpenZAP_Rhino
>

The openr2 page was already ported and some example configs pages too.
 The other ones still need the porting though.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From msc at freeswitch.org  Fri Oct 22 17:06:14 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 22 Oct 2010 17:06:14 -0700
Subject: [Freeswitch-users] Call for wiki help: cleanup OpenZAP pages on
	wiki
In-Reply-To: <AANLkTinqb2Nkgq594n_6A+ZGuieCsxHb-_58ap9JGgPq@mail.gmail.com>
References: <AANLkTiks-kt_wqopLnwbYkX0qf2_btPAJj=6LkCuFPG1@mail.gmail.com>
	<AANLkTinqb2Nkgq594n_6A+ZGuieCsxHb-_58ap9JGgPq@mail.gmail.com>
Message-ID: <AANLkTi=RgGHTc9jtAHHjtcoNmRhMR5ic3ougmcy8eKhX@mail.gmail.com>

Moy,

Thanks for adding the redirect message!
-MC

On Fri, Oct 22, 2010 at 4:52 PM, Moises Silva <moises.silva at gmail.com>wrote:

> On Wed, Oct 20, 2010 at 2:50 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> > We need some assistance with the migration from OpenZAP to FreeTDM. In
> > particular we have some legacy OpenZAP pages that need to be translated
> into
> > FreeTDM pages on the wiki. In some cases it may be as simple as swapping
> > "OpenZAP" with "FreeTDM" and doing a page redirect. In other cases it may
> be
> > a bit more involved. I am inviting anyone with OpenZAP and FreeTDM
> > experience to work on these wiki pages:
> >
> > http://wiki.freeswitch.org/wiki/Openzap.conf_Examples
> > http://wiki.freeswitch.org/wiki/Openzap.conf.xml_Examples
> >
> http://wiki.freeswitch.org/wiki/Configuration_OpenZAP-_DigiumTDM400P_Example
> > http://wiki.freeswitch.org/wiki/Openzap.sangoma.libpri
> > http://wiki.freeswitch.org/wiki/OpenZap_Dahdi
> > http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2
> > http://wiki.freeswitch.org/wiki/OpenZAP_Rhino
> >
>
> The openr2 page was already ported and some example configs pages too.
>  The other ones still need the porting though.
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
> L3R 9R6 Canada
> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From covici at ccs.covici.com  Fri Oct 22 19:29:17 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 22 Oct 2010 22:29:17 -0400
Subject: [Freeswitch-users] problem with session timers (I think)
Message-ID: <13102.1287800957@ccs.covici.com>

Hi.  When I call a certain number, using fs and portaudio, fs hangs up
in just about a half hour.  This is suspiciously like a session timer or
something, so I added the parameter <param name="session-timeout"
value="12600"/>  but it is still happening.  I did not change the
internal profile, but I did change the external one in the client which
registers with the internal profile -- fs is the client as well as the
pbx.

Any assistance on these would be appreciated.
-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From dome at tel.co.th  Fri Oct 22 21:43:20 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Sat, 23 Oct 2010 11:43:20 +0700
Subject: [Freeswitch-users] problem with session timers (I think)
In-Reply-To: <13102.1287800957@ccs.covici.com>
References: <13102.1287800957@ccs.covici.com>
Message-ID: <AANLkTi=A4kqcE09q5UJdNUaR7PcZnv=FT+9Lmo7DGkyA@mail.gmail.com>

Please try
<action application="sched_hangup" data="+12600 alloted_timeout"/>

Dome C.

2010/10/23  <covici at ccs.covici.com>:
> Hi. ?When I call a certain number, using fs and portaudio, fs hangs up
> in just about a half hour. ?This is suspiciously like a session timer or
> something, so I added the parameter <param name="session-timeout"
> value="12600"/> ?but it is still happening. ?I did not change the
> internal profile, but I did change the external one in the client which
> registers with the internal profile -- fs is the client as well as the
> pbx.
>
> Any assistance on these would be appreciated.
> --
> Your life is like a penny. ?You're going to lose it. ?The question is:
> How do
> you spend it?
>
> ? ? ? ? John Covici
> ? ? ? ? covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From covici at ccs.covici.com  Sat Oct 23 01:38:27 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sat, 23 Oct 2010 04:38:27 -0400
Subject: [Freeswitch-users] problem with session timers (I think)
In-Reply-To: <AANLkTi=A4kqcE09q5UJdNUaR7PcZnv=FT+9Lmo7DGkyA@mail.gmail.com>
References: <13102.1287800957@ccs.covici.com>
	<AANLkTi=A4kqcE09q5UJdNUaR7PcZnv=FT+9Lmo7DGkyA@mail.gmail.com>
Message-ID: <16672.1287823107@ccs.covici.com>

I don't want the call to hangup at all, I just put that number in for a
very large number.  Sorry if I was unclear.

Dome Charoenyost <dome at tel.co.th> wrote:

> Please try
> <action application="sched_hangup" data="+12600 alloted_timeout"/>
> 
> Dome C.
> 
> 2010/10/23  <covici at ccs.covici.com>:
> > Hi. ?When I call a certain number, using fs and portaudio, fs hangs up
> > in just about a half hour. ?This is suspiciously like a session timer or
> > something, so I added the parameter <param name="session-timeout"
> > value="12600"/> ?but it is still happening. ?I did not change the
> > internal profile, but I did change the external one in the client which
> > registers with the internal profile -- fs is the client as well as the
> > pbx.
> >
> > Any assistance on these would be appreciated.
> > --
> > Your life is like a penny. ?You're going to lose it. ?The question is:
> > How do
> > you spend it?
> >
> > ? ? ? ? John Covici
> > ? ? ? ? covici at ccs.covici.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From steveayre at gmail.com  Sat Oct 23 02:56:44 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 23 Oct 2010 10:56:44 +0100
Subject: [Freeswitch-users] problem with session timers (I think)
In-Reply-To: <13102.1287800957@ccs.covici.com>
References: <13102.1287800957@ccs.covici.com>
Message-ID: <AANLkTinz9b6Ep4YHOOqfe41Q2BqeRS7buXCcOvRD_cfu@mail.gmail.com>

Do you have a debug-level log of when FS hangs the call up? It should
indicate why the call's hung up.

There's several things it could be - hangup from the other end, a
requested hangup on FS, session timers, (lack of) RTP timer...

-Steve


On 23 October 2010 03:29,  <covici at ccs.covici.com> wrote:
> Hi. ?When I call a certain number, using fs and portaudio, fs hangs up
> in just about a half hour. ?This is suspiciously like a session timer or
> something, so I added the parameter <param name="session-timeout"
> value="12600"/> ?but it is still happening. ?I did not change the
> internal profile, but I did change the external one in the client which
> registers with the internal profile -- fs is the client as well as the
> pbx.
>
> Any assistance on these would be appreciated.
> --
> Your life is like a penny. ?You're going to lose it. ?The question is:
> How do
> you spend it?
>
> ? ? ? ? John Covici
> ? ? ? ? covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From covici at ccs.covici.com  Sat Oct 23 03:37:33 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sat, 23 Oct 2010 06:37:33 -0400
Subject: [Freeswitch-users] problem with session timers (I think)
In-Reply-To: <AANLkTinz9b6Ep4YHOOqfe41Q2BqeRS7buXCcOvRD_cfu@mail.gmail.com>
References: <13102.1287800957@ccs.covici.com>
	<AANLkTinz9b6Ep4YHOOqfe41Q2BqeRS7buXCcOvRD_cfu@mail.gmail.com>
Message-ID: <17763.1287830253@ccs.covici.com>

It just says NORMAL_CLEARiNG but its always after a half hour, so I
thought it was session timers.  Could there be some other reason?

Steven Ayre <steveayre at gmail.com> wrote:

> Do you have a debug-level log of when FS hangs the call up? It should
> indicate why the call's hung up.
> 
> There's several things it could be - hangup from the other end, a
> requested hangup on FS, session timers, (lack of) RTP timer...
> 
> -Steve
> 
> 
> On 23 October 2010 03:29,  <covici at ccs.covici.com> wrote:
> > Hi. ?When I call a certain number, using fs and portaudio, fs hangs up
> > in just about a half hour. ?This is suspiciously like a session timer or
> > something, so I added the parameter <param name="session-timeout"
> > value="12600"/> ?but it is still happening. ?I did not change the
> > internal profile, but I did change the external one in the client which
> > registers with the internal profile -- fs is the client as well as the
> > pbx.
> >
> > Any assistance on these would be appreciated.
> > --
> > Your life is like a penny. ?You're going to lose it. ?The question is:
> > How do
> > you spend it?
> >
> > ? ? ? ? John Covici
> > ? ? ? ? covici at ccs.covici.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From codecomplete at free.fr  Sat Oct 23 05:49:27 2010
From: codecomplete at free.fr (GillesToo)
Date: Sat, 23 Oct 2010 05:49:27 -0700 (PDT)
Subject: [Freeswitch-users] [Linksys 3102] Couple of issues using FS and
	XLite
In-Reply-To: <AANLkTimkLpGj7i3GcsYP8LZ8FnABTvz+Bmjb_JOqmkwB@mail.gmail.com>
References: <1287755399019-5662612.post@n2.nabble.com>
	<AANLkTin2q7o1DHp0tUguHiVMr1aJxLo4ryoedAHgW18_@mail.gmail.com>
	<1287759334654-5662810.post@n2.nabble.com>
	<AANLkTi=R6MHHUWHpx64MM4K3waTZjMF3Cfe+4goo5qnH@mail.gmail.com>
	<AANLkTimkLpGj7i3GcsYP8LZ8FnABTvz+Bmjb_JOqmkwB@mail.gmail.com>
Message-ID: <1287838167405-5665332.post@n2.nabble.com>


Thanks Jeffrey, but the download from MegaUpload got stuck halfway through,
and when I tried downloading it again, it said "The file you are trying to
access is temporarily unavailable."

Would it be possible to upload it to a non-limited, temporary server, or in
the Freeswitch server to replace the old 1.0.4 version there?

http://files.freeswitch.org/windows_installer/

Thank you.

-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Linksys-3102-Couple-of-issues-using-FS-and-XLite-tp5662612p5665332.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From codecomplete at free.fr  Sat Oct 23 06:06:25 2010
From: codecomplete at free.fr (GillesToo)
Date: Sat, 23 Oct 2010 06:06:25 -0700 (PDT)
Subject: [Freeswitch-users] [Linksys 3102] Couple of issues using FS and
	XLite
In-Reply-To: <1287838167405-5665332.post@n2.nabble.com>
References: <1287755399019-5662612.post@n2.nabble.com>
	<AANLkTin2q7o1DHp0tUguHiVMr1aJxLo4ryoedAHgW18_@mail.gmail.com>
	<1287759334654-5662810.post@n2.nabble.com>
	<AANLkTi=R6MHHUWHpx64MM4K3waTZjMF3Cfe+4goo5qnH@mail.gmail.com>
	<AANLkTimkLpGj7i3GcsYP8LZ8FnABTvz+Bmjb_JOqmkwB@mail.gmail.com>
	<1287838167405-5665332.post@n2.nabble.com>
Message-ID: <1287839185519-5665357.post@n2.nabble.com>


After waiting a few minutes and giving it another shot, the download went OK
and I could install this latest version on an XP test host.

However, after connecting to it from a remote XP host with XLite and dialing
9999 to check the music on hold feature, I get the error message "The person
you are trying to call is unavailable".

Here's what the console says:
=========
freeswitch at msi-test> 2010-10-23 16:02:15.125000 [NOTICE]
switch_channel.c:784 Ne
w Channel sofia/internal/1001 at 192.168.0.7
[d4882f69-f096-4626-aa44-d2d7e63657bd]

2010-10-23 16:02:15.281250 [INFO] mod_dialplan_xml.c:331 Processing Frees
witch <1001>->9999 in context default
2010-10-23 16:02:15.312500 [NOTICE] switch_ivr.c:1507 Transfer
sofia/internal/10
01 at 192.168.0.7 to enum[9999 at default]
2010-10-23 16:02:15.390625 [INFO] switch_core_state_machine.c:142 No Route,
Abor
ting
2010-10-23 16:02:15.390625 [NOTICE] switch_core_state_machine.c:143 Hangup
sofia
/internal/1001 at 192.168.0.7 [CS_ROUTING] [NO_ROUTE_DESTINATION]
2010-10-23 16:02:15.500000 [NOTICE] switch_core_session.c:1242 Session 6
(sofia/
internal/1001 at 192.168.0.7) Ended
2010-10-23 16:02:15.500000 [NOTICE] switch_core_session.c:1244 Close Channel
sof
ia/internal/1001 at 192.168.0.7 [CS_DESTROY]
=========

If this binary was built from the latest SVN code, could the source code be
in an unstable state? If that's the case, would it be possible to recompile
the code from a stable 1.0.6 source, so I can give it another shot and see
if the original XLite problem went away?

Thank you.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Linksys-3102-Couple-of-issues-using-FS-and-XLite-tp5662612p5665357.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mcampbellsmith at gmail.com  Sat Oct 23 05:04:52 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sat, 23 Oct 2010 23:04:52 +1100
Subject: [Freeswitch-users] FS not detecting public IP address change
Message-ID: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>

HI!

I know this question must have been answered 100's of times....

My adsl is a bit dodgy at the moment and tends to go up and down; which
means I get assigned a new ip address from my ISP.  FS is nat'd behind a
upnp capable router.

The problem is that FS does not seem to be detecting the change.   For both
the internal and external profile, I have auto-nat setup, and I have tried
stun and host settings in vars.conf.  When I issue a nat_map status I see my
old IP address, and if I then issue a nat_map reinit, I see the new public
IP address.  How can I get this to be automatic?

Also the public IP address shown in the nat_map status is not the same as
shown in the sofia profile internal/external printouts.  Why is this?

I'm sure this is configuration, just not sure what to change.

Thanks!
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From testeador01 at gmail.com  Sat Oct 23 06:25:09 2010
From: testeador01 at gmail.com (Milena)
Date: Sat, 23 Oct 2010 08:25:09 -0500
Subject: [Freeswitch-users] [Linksys 3102] Couple of issues using FS and
	XLite
In-Reply-To: <1287839185519-5665357.post@n2.nabble.com>
References: <1287755399019-5662612.post@n2.nabble.com>
	<AANLkTin2q7o1DHp0tUguHiVMr1aJxLo4ryoedAHgW18_@mail.gmail.com>
	<1287759334654-5662810.post@n2.nabble.com>
	<AANLkTi=R6MHHUWHpx64MM4K3waTZjMF3Cfe+4goo5qnH@mail.gmail.com>
	<AANLkTimkLpGj7i3GcsYP8LZ8FnABTvz+Bmjb_JOqmkwB@mail.gmail.com>
	<1287838167405-5665332.post@n2.nabble.com>
	<1287839185519-5665357.post@n2.nabble.com>
Message-ID: <AANLkTikX0QnxAQhrp=3qJoEKbZibb8J7NqxmhWDY21Hr@mail.gmail.com>

You should take the wiki and read it all,

echo extension is not 9999 anymore, get started here:
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Some_common_extensions_for_testing

Have fun :)

On Sat, Oct 23, 2010 at 8:06 AM, GillesToo <codecomplete at free.fr> wrote:

>
> After waiting a few minutes and giving it another shot, the download went
> OK
> and I could install this latest version on an XP test host.
>
> However, after connecting to it from a remote XP host with XLite and
> dialing
> 9999 to check the music on hold feature, I get the error message "The
> person
> you are trying to call is unavailable".
>
> Here's what the console says:
> =========
> freeswitch at msi-test> 2010-10-23 16:02:15.125000 [NOTICE]
> switch_channel.c:784 Ne
> w Channel sofia/internal/1001 at 192.168.0.7
> [d4882f69-f096-4626-aa44-d2d7e63657bd]
>
> 2010-10-23 16:02:15.281250 [INFO] mod_dialplan_xml.c:331 Processing Frees
> witch <1001>->9999 in context default
> 2010-10-23 16:02:15.312500 [NOTICE] switch_ivr.c:1507 Transfer
> sofia/internal/10
> 01 at 192.168.0.7 to enum[9999 at default]
> 2010-10-23 16:02:15.390625 [INFO] switch_core_state_machine.c:142 No Route,
> Abor
> ting
> 2010-10-23 16:02:15.390625 [NOTICE] switch_core_state_machine.c:143 Hangup
> sofia
> /internal/1001 at 192.168.0.7 [CS_ROUTING] [NO_ROUTE_DESTINATION]
> 2010-10-23 16:02:15.500000 [NOTICE] switch_core_session.c:1242 Session 6
> (sofia/
> internal/1001 at 192.168.0.7) Ended
> 2010-10-23 16:02:15.500000 [NOTICE] switch_core_session.c:1244 Close
> Channel
> sof
> ia/internal/1001 at 192.168.0.7 [CS_DESTROY]
> =========
>
> If this binary was built from the latest SVN code, could the source code be
> in an unstable state? If that's the case, would it be possible to recompile
> the code from a stable 1.0.6 source, so I can give it another shot and see
> if the original XLite problem went away?
>
> Thank you.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Linksys-3102-Couple-of-issues-using-FS-and-XLite-tp5662612p5665357.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch-list at puzzled.xs4all.nl  Sat Oct 23 06:27:08 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Sat, 23 Oct 2010 15:27:08 +0200
Subject: [Freeswitch-users] [Linksys 3102] Couple of issues using FS and
 XLite
In-Reply-To: <1287839185519-5665357.post@n2.nabble.com>
References: <1287755399019-5662612.post@n2.nabble.com>	<AANLkTin2q7o1DHp0tUguHiVMr1aJxLo4ryoedAHgW18_@mail.gmail.com>	<1287759334654-5662810.post@n2.nabble.com>	<AANLkTi=R6MHHUWHpx64MM4K3waTZjMF3Cfe+4goo5qnH@mail.gmail.com>	<AANLkTimkLpGj7i3GcsYP8LZ8FnABTvz+Bmjb_JOqmkwB@mail.gmail.com>	<1287838167405-5665332.post@n2.nabble.com>
	<1287839185519-5665357.post@n2.nabble.com>
Message-ID: <4CC2E2AC.7090301@puzzled.xs4all.nl>

On 10/23/2010 03:06 PM, GillesToo wrote:
[snip]
> If this binary was built from the latest SVN code, could the source code be
> in an unstable state? If that's the case, would it be possible to recompile
> the code from a stable 1.0.6 source

Latest git (not svn) has better stability and zillions of bugfixes 
compared to 1.0.6. The general recommendation is to use latest git.

I'm not an expert but your problem seems related to "2010-10-23 
16:02:15.390625 [INFO] switch_core_state_machine.c:142 No Route,
Aborting" and going through enum. Seems something can not be reached.

Regards,
Patrick


From curriegrad2004 at gmail.com  Sat Oct 23 06:44:02 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Sat, 23 Oct 2010 06:44:02 -0700
Subject: [Freeswitch-users] [Linksys 3102] Couple of issues using FS and
	XLite
In-Reply-To: <4CC2E2AC.7090301@puzzled.xs4all.nl>
References: <1287755399019-5662612.post@n2.nabble.com>
	<AANLkTin2q7o1DHp0tUguHiVMr1aJxLo4ryoedAHgW18_@mail.gmail.com>
	<1287759334654-5662810.post@n2.nabble.com>
	<AANLkTi=R6MHHUWHpx64MM4K3waTZjMF3Cfe+4goo5qnH@mail.gmail.com>
	<AANLkTimkLpGj7i3GcsYP8LZ8FnABTvz+Bmjb_JOqmkwB@mail.gmail.com>
	<1287838167405-5665332.post@n2.nabble.com>
	<1287839185519-5665357.post@n2.nabble.com>
	<4CC2E2AC.7090301@puzzled.xs4all.nl>
Message-ID: <AANLkTi=ZH0vjD_sJeWoJ5WC7ZbRSs_rKgQhtZ7cAELqO@mail.gmail.com>

git code definitely has more bug fixes that the 1.0.6 code. However
the only feature I never managed to get working is mod_dingaling with
TLS support - apparently that's broken under Windows for some reason.
dingaling *with* TLS support is needed to interface FS with Google
Voice.


On Sat, Oct 23, 2010 at 6:27 AM, Patrick Lists
<freeswitch-list at puzzled.xs4all.nl> wrote:
> On 10/23/2010 03:06 PM, GillesToo wrote:
> [snip]
>> If this binary was built from the latest SVN code, could the source code be
>> in an unstable state? If that's the case, would it be possible to recompile
>> the code from a stable 1.0.6 source
>
> Latest git (not svn) has better stability and zillions of bugfixes
> compared to 1.0.6. The general recommendation is to use latest git.
>
> I'm not an expert but your problem seems related to "2010-10-23
> 16:02:15.390625 [INFO] switch_core_state_machine.c:142 No Route,
> Aborting" and going through enum. Seems something can not be reached.
>
> Regards,
> Patrick
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From christian.knoblauch at astylos.de  Sat Oct 23 06:30:11 2010
From: christian.knoblauch at astylos.de (Christian Knoblauch)
Date: Sat, 23 Oct 2010 16:30:11 +0300
Subject: [Freeswitch-users] FreeSwitch and Cisco 7941 IP Phone (
	INVITE	port number )
In-Reply-To: <001201cb712c$696dbbd0$3c493370$@astylos.de>
References: <057bc2928b30a59af5a20d08c3fb5141@astylos.de>	<AANLkTik5BkOMcDvaQseJocsp2276_yAa31-PEkpK1kMG@mail.gmail.com>
	<001201cb712c$696dbbd0$3c493370$@astylos.de>
Message-ID: <000901cb72b6$6d90eda0$48b2c8e0$@astylos.de>

OK,

 

By using

 

<variable name="sip-force-contact" value="sip:1011 at 10.0.0.2:5060"/>

 

for each "Cisco IP Phone 7941" directory entry it works fine (forcing the
UDP port number)

 

Regards

Christian

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Christian Knoblauch
Sent: Thursday, October 21, 2010 5:30 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] FreeSwitch and Cisco 7941 IP Phone ( INVITE
port number )

 

Hi,

 

After commenting-out the mentioned code block, the INVITE is send to port
5060 so that the call gets established.

 

But still:

 

After hanging up one of the 7941 (call was established between 2 of them),
the BYE message towards the other 7941 is still send to the higher port
number ( so the corresponding 7941 does not end the line )

 

Thanks and regards

Christian 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ognjen
Seslija
Sent: Wednesday, October 20, 2010 12:23 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FreeSwitch and Cisco 7941 IP Phone ( INVITE
port number )

 

Please see http://jira.freeswitch.org/browse/FS-2773 I reported.

The mentioned code is now commented in FS, so nat is no longer assumed.

Regards.

On Mon, Oct 18, 2010 at 5:15 PM, Christian Knoblauch
<christian.knoblauch at astylos.de> wrote:


Hello,
 
 
This is about FreeSwitch 1.0.6 (Windows Build) and Cisco 7941 IP Phone, and
the same issue was recently reported by another subscriber of this list
(Michael W. Lucas).
 
Anyhow, I attach traces so that maybe someone can look into it.
 
 
CONTEXT:
 
The 7941 successfull REGISTER in FreeSwitch, and is able to call another
softphone.
The softphone is not able to call the 7941, and this is because the INVITE
from FreeSwitch towards 7941 goes to the UDP port that the 7941 was using
during REGISTER (high port number, instead of 5060)
 
I compared this with traces for 3CX where all works fine. 3CX sends the
INVITE towards 7941 to UDP port 5060, and the 7941 seems to like this :-)
 
 
Please find attached the REGISTER / INVITE traces for booth 3CX and
FreeSwitch, and also the FreeSwitch console-output from"sofia profile
internal siptrace on"
 
 
Thanks for review !
 
Regards
Christian
 


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

 

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From abid_freeswitch at live.com  Sat Oct 23 07:01:28 2010
From: abid_freeswitch at live.com (Abid Saleem)
Date: Sat, 23 Oct 2010 20:01:28 +0600
Subject: [Freeswitch-users] Radius AAA
In-Reply-To: <AANLkTi=Cgi4LjDTfZAzvJOSU0u1+OWMsewa=-KOU1JeZ@mail.gmail.com>
References: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>,
	<AANLkTi=GiQaO-48BP=T8c-A+09sJMKzQH7BX=ewBdB8R@mail.gmail.com>,
	<AANLkTikw8LGDqQki16D30b8JL0eJid2UOTs-bEaZSc2k@mail.gmail.com>,
	<4CADC2E4.90608@gmail.com>,
	<COL109-W528C7DF8FD0E4D203FD866F4570@phx.gbl>,
	<AANLkTi=oXjajMBSn8tJCT8chm6mM8E4XVzXVynDda_+O@mail.gmail.com>,
	<AANLkTi=P1QH2L5B-HckXvqRgHsQEpEia2X7mA_DDT=DN@mail.gmail.com>,
	<COL109-W43A18E93B3B89A7D57AE61F45B0@phx.gbl>,
	<AANLkTi=Cgi4LjDTfZAzvJOSU0u1+OWMsewa=-KOU1JeZ@mail.gmail.com>
Message-ID: <COL109-W27AC0F47F5E5A5F4404B4DF45F0@phx.gbl>


Dear Tihomir,
Thanks a lot. It is loaded now but when I try to trigger auth in the dialplan, the module is not being called at all. I am using the example config as provided by you and Nazim.
Scenerio is that I call from 1000 to 2000. In dialplan for destination_number 2000, I call the auth_function Application as said by Nazim but it is not being called at all. I have no clue why. Please help.
Thanks.
Regards---------------Abid Saleem

Date: Tue, 19 Oct 2010 15:46:19 +0200
From: tculjaga at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Radius AAA


[root at terminus lib]# ln -sf /usr/local/lib/libfreeradius-client.so.2 libfreeradius-client.so


try running this this:
ln -sf /usr/local/lib/libfreeradius-client.so.2 libfreeradius-client.so.2






_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 		 	   		  
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From rupa at rupa.com  Sat Oct 23 07:06:53 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 23 Oct 2010 09:06:53 -0500
Subject: [Freeswitch-users] FS not detecting public IP address change
In-Reply-To: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>
References: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>
Message-ID: <AANLkTinEoTcLmUVcpras8-m8pdkfpE=_r0spz1S50iMF@mail.gmail.com>

In a upnp config, FS is depending on the router to notify it of the ip
address change.  It is possible the upnp message(s) aren't formatted as
expected.  If you turn on debug logging you'll get the upnp messages on the
console.  If you can pastebin the capture of those while dropping and
restarting the dsl connection I can maybe see if there is something obvious
going on.  The debug should also have messages related to the processing of
those upnp messages.

On Sat, Oct 23, 2010 at 7:04 AM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> HI!
>
> I know this question must have been answered 100's of times....
>
> My adsl is a bit dodgy at the moment and tends to go up and down; which
> means I get assigned a new ip address from my ISP.  FS is nat'd behind a
> upnp capable router.
>
> The problem is that FS does not seem to be detecting the change.   For both
> the internal and external profile, I have auto-nat setup, and I have tried
> stun and host settings in vars.conf.  When I issue a nat_map status I see my
> old IP address, and if I then issue a nat_map reinit, I see the new public
> IP address.  How can I get this to be automatic?
>
> Also the public IP address shown in the nat_map status is not the same as
> shown in the sofia profile internal/external printouts.  Why is this?
>
> I'm sure this is configuration, just not sure what to change.
>
> Thanks!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From rupa at rupa.com  Sat Oct 23 07:08:06 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 23 Oct 2010 09:08:06 -0500
Subject: [Freeswitch-users] FS not detecting public IP address change
In-Reply-To: <AANLkTinEoTcLmUVcpras8-m8pdkfpE=_r0spz1S50iMF@mail.gmail.com>
References: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>
	<AANLkTinEoTcLmUVcpras8-m8pdkfpE=_r0spz1S50iMF@mail.gmail.com>
Message-ID: <AANLkTi=MoV27FW0Ukd6K+mqmgnkyNgb9hjFx-vBYV1ak@mail.gmail.com>

btw: what router are you using?

On Sat, Oct 23, 2010 at 9:06 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> In a upnp config, FS is depending on the router to notify it of the ip
> address change.  It is possible the upnp message(s) aren't formatted as
> expected.  If you turn on debug logging you'll get the upnp messages on the
> console.  If you can pastebin the capture of those while dropping and
> restarting the dsl connection I can maybe see if there is something obvious
> going on.  The debug should also have messages related to the processing of
> those upnp messages.
>
> On Sat, Oct 23, 2010 at 7:04 AM, Mark Campbell-Smith <
> mcampbellsmith at gmail.com> wrote:
>
>> HI!
>>
>> I know this question must have been answered 100's of times....
>>
>> My adsl is a bit dodgy at the moment and tends to go up and down; which
>> means I get assigned a new ip address from my ISP.  FS is nat'd behind a
>> upnp capable router.
>>
>> The problem is that FS does not seem to be detecting the change.   For
>> both the internal and external profile, I have auto-nat setup, and I have
>> tried stun and host settings in vars.conf.  When I issue a nat_map status I
>> see my old IP address, and if I then issue a nat_map reinit, I see the new
>> public IP address.  How can I get this to be automatic?
>>
>> Also the public IP address shown in the nat_map status is not the same as
>> shown in the sofia profile internal/external printouts.  Why is this?
>>
>> I'm sure this is configuration, just not sure what to change.
>>
>> Thanks!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>



-- 
-Rupa
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From dftoro at yahoo.com  Sat Oct 23 07:15:13 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Sat, 23 Oct 2010 07:15:13 -0700 (PDT)
Subject: [Freeswitch-users] dialogic FXO card will support freeSWITCH
In-Reply-To: <AANLkTi=zao8buWwvv3TDAPaP5sOPpc30_ik8VSG9ygyz@mail.gmail.com>
Message-ID: <14973.86561.qm@web33501.mail.mud.yahoo.com>

Hi, check DiaStar project 

http://www.projectdiastar.org/ 
http://wiki.projectdiastar.org/index.php/FreeSWITCH_Woomera_Client_Install
http://wiki.projectdiastar.org/index.php/FreeSWITCH/Woomera_User_Guide

The hardware supported is Digital Network Interface (DNI) boards

Diego Toro 

http://voipensando.blogspot.com/

--- On Fri, 10/22/10, Michael Collins <msc at freeswitch.org> wrote:

From: Michael Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] dialogic FXO card will support freeSWITCH
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Friday, October 22, 2010, 3:40 PM

As far as I know, only zaptel/dahdi compatible cards and Sangoma cards are compatible. (Khomp is coming soon I believe.) I've never seen or heard of a Dialogic card working with FreeSWITCH.

-MC


On Fri, Oct 22, 2010 at 9:08 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:

Hi. 

Can you anyone plz tell me, whether Dialogic FXO card will support? freeswitch? 


thanks,

Regards,
Venkat. 

-- 

If you have come to help me, you are wasting your time. 


If you have come to because your liberation is bound up in mine, we can work together.

?
Regards
Venkatesan OV.



_______________________________________________

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

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_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From jeff at jefflenk.com  Sat Oct 23 09:00:23 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sat, 23 Oct 2010 09:00:23 -0700 (PDT)
Subject: [Freeswitch-users] [Linksys 3102] Couple of issues using FS and
	XLite
In-Reply-To: <AANLkTi=ZH0vjD_sJeWoJ5WC7ZbRSs_rKgQhtZ7cAELqO@mail.gmail.com>
References: <1287755399019-5662612.post@n2.nabble.com>
	<AANLkTin2q7o1DHp0tUguHiVMr1aJxLo4ryoedAHgW18_@mail.gmail.com>
	<1287759334654-5662810.post@n2.nabble.com>
	<AANLkTi=R6MHHUWHpx64MM4K3waTZjMF3Cfe+4goo5qnH@mail.gmail.com>
	<AANLkTimkLpGj7i3GcsYP8LZ8FnABTvz+Bmjb_JOqmkwB@mail.gmail.com>
	<1287838167405-5665332.post@n2.nabble.com>
	<1287839185519-5665357.post@n2.nabble.com>
	<4CC2E2AC.7090301@puzzled.xs4all.nl>
	<AANLkTi=ZH0vjD_sJeWoJ5WC7ZbRSs_rKgQhtZ7cAELqO@mail.gmail.com>
Message-ID: <1287849623835-5665744.post@n2.nabble.com>


mod_dingaling does not have tls support under windows at this time. This is
an artifact of not having support for gnutls under windows in the project -
patches welcome.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Linksys-3102-Couple-of-issues-using-FS-and-XLite-tp5662612p5665744.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fraserredmond at gmail.com  Sat Oct 23 09:44:04 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Sat, 23 Oct 2010 17:44:04 +0100
Subject: [Freeswitch-users] Loss of first second of media
In-Reply-To: <AANLkTinpaRivNHteDp4i6WJjmgsPAE6otVx5LjgJcvWr@mail.gmail.com>
References: <AANLkTimECwtVmnbFsBJRiN3YSaGyXbvh-sKtKYPey8xx@mail.gmail.com>
	<AANLkTik5H74MupY8iyx6Ao4=Rv=wCm8jv-w=xCtSnm0Z@mail.gmail.com>
	<AANLkTikpVH-s6AvJ7fQoZu2PE4dduTg+d-ygyBWSfr4R@mail.gmail.com>
	<AANLkTi=FZDqev9R1aTMpdvTk8t9NfxiyeisQSRBxfTEb@mail.gmail.com>
	<AANLkTikY68ovuiSO9xkbZZPSskxCNsjrpTSdb-42bwJB@mail.gmail.com>
	<AANLkTikiG3xPKDVkwU4GDk5t6KDe67AQmM-dkheMCSu7@mail.gmail.com>
	<AANLkTinJRcpRx_BQyKifECcurYb+vG2P8Nct1xT_X3XO@mail.gmail.com>
	<AANLkTinpaRivNHteDp4i6WJjmgsPAE6otVx5LjgJcvWr@mail.gmail.com>
Message-ID: <AANLkTikiVxFAOwftrmfqd3LtD4H__UUytsM117WBhZy0@mail.gmail.com>

Thanks Anthony,

Finally managed to get a sip trace - could you do me a favor and take a look
and/or give me some ideas of what to look for?

http://pastebin.freeswitch.org/14300

I've highlighted lines 168 and 193. In between these lines is where the
number is dialed and rings once, then picks up, then theres silence for a
second or two, and that second SIP message is when I start hearing audio.

Thanks,
Fraser




On Thu, Oct 21, 2010 at 6:44 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> its a blue message on cli
>
> It could also be the other side expecting us to send media first or
> something silly.
> try getting a sip trace and figure out when the rtp starts arriving.
>
>
> On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond
> <fraserredmond at gmail.com> wrote:
> > Sorry, yes, I am setting ignore_early_media=true in the first area. (Or
> are
> > you saying that should be off? I forget now why I needed it on, but there
> > was a reason I added it.)
> >
> > Yes, the bridge doesn't start until after the A-leg has answered.
> >
> > Thanks for the suggestion about nat/auto-changing port, I'll have a look
> > into that - would that be in the cli output or in a sip trace? I've
> already
> > looked and it's not appearing in the CLI output (with loglevel=debug),
> > haven't looked in the sip trace yet.
> >
> > Cheers,
> > Fraser
> >
> >
> >
> >
> > On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> are you setting ignore_early_media=true in the first vars=values area?
> >>
> >> This looks like you could be calling one leg who is still not answered
> >> and then bridging it to another dest.  The bridge app will wait for
> >> the first leg to answer before bridging.
> >>
> >> Also if you have any NAT anywhere, look for an "auto-changing port"
> >> type message which can also be attributed to this due to a detection
> >> period for incorrect ports.
> >>
> >>
> >>
> >> On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond
> >> <fraserredmond at gmail.com> wrote:
> >> > event_socket:
> >> > api originate {vars=values}user/$fromExtn at Domain
> >> > 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline
> >> >
> >> > then
> >> >
> >> > dialplan:
> >> > <action application="set"
> >> > data="effective_caller_id_number=+1800number"/>
> >> > <action application="set" data="effective_caller_id_name="/>
> >> > (set and/or export a bunch of other vars too)
> >> > <action application="set"
> >> >
> >> > data="dial_string=sofia/gateway/
> gatewayname.com/00${destination_number}<http://gatewayname.com/00$%7Bdestination_number%7D>
> "/>
> >> > <action application="bridge" data="${dial_string}"/>
> >> >
> >> > Cheers,
> >> > Fraser
> >> >
> >> >
> >> >
> >> >
> >> > On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> how are you accomplishing that? by which technique?
> >> >>
> >> >> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond
> >> >> <fraserredmond at gmail.com> wrote:
> >> >> > The call is originated from Freeswitch (via CLI) to a softphone,
> then
> >> >> > when
> >> >> > that is connected it bridges out to the gateway.
> >> >> >
> >> >> > Cheers,
> >> >> > Fraser
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale
> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >>
> >> >> >> Where is the other side of this call coming from?
> >> >> >>
> >> >> >> [ (   ) ] -> FS -> (PSTN via SIP)
> >> >> >>
> >> >> >> What goes in the empty space above?
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Sat Oct 23 09:45:08 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 23 Oct 2010 17:45:08 +0100
Subject: [Freeswitch-users] problem with session timers (I think)
In-Reply-To: <17763.1287830253@ccs.covici.com>
References: <13102.1287800957@ccs.covici.com>
	<AANLkTinz9b6Ep4YHOOqfe41Q2BqeRS7buXCcOvRD_cfu@mail.gmail.com>
	<17763.1287830253@ccs.covici.com>
Message-ID: <AANLkTikDy=BnTbuonjLASJ4xTvyzeRXEnZJaPkEqduRA@mail.gmail.com>

Are you able to post the log, at debug level, with sip trace on and
and with context (lines above the disconnect)?

Also, is there media flowing? Some SIP servers will disconnect a
connection if there's been no media for a certain amount of time. FS
has that option but it's off by default. In a bypass media situation
that could cause problems.

NAT is another possibility. Some routers will disconnect a connection
that hasn't had any activity for a while.

-Steve


On 23 October 2010 11:37,  <covici at ccs.covici.com> wrote:
> It just says NORMAL_CLEARiNG but its always after a half hour, so I
> thought it was session timers. ?Could there be some other reason?
>
> Steven Ayre <steveayre at gmail.com> wrote:
>
>> Do you have a debug-level log of when FS hangs the call up? It should
>> indicate why the call's hung up.
>>
>> There's several things it could be - hangup from the other end, a
>> requested hangup on FS, session timers, (lack of) RTP timer...
>>
>> -Steve
>>
>>
>> On 23 October 2010 03:29, ?<covici at ccs.covici.com> wrote:
>> > Hi. ?When I call a certain number, using fs and portaudio, fs hangs up
>> > in just about a half hour. ?This is suspiciously like a session timer or
>> > something, so I added the parameter <param name="session-timeout"
>> > value="12600"/> ?but it is still happening. ?I did not change the
>> > internal profile, but I did change the external one in the client which
>> > registers with the internal profile -- fs is the client as well as the
>> > pbx.
>> >
>> > Any assistance on these would be appreciated.
>> > --
>> > Your life is like a penny. ?You're going to lose it. ?The question is:
>> > How do
>> > you spend it?
>> >
>> > ? ? ? ? John Covici
>> > ? ? ? ? covici at ccs.covici.com
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> --
> Your life is like a penny. ?You're going to lose it. ?The question is:
> How do
> you spend it?
>
> ? ? ? ? John Covici
> ? ? ? ? covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mario_fs at mgtech.com  Sat Oct 23 11:14:52 2010
From: mario_fs at mgtech.com (Mario)
Date: Sat, 23 Oct 2010 11:14:52 -0700
Subject: [Freeswitch-users] Latest git missing files on osX?
Message-ID: <4CC3261C.50002@mgtech.com>

I download a fresh copy today to put in the final installation and I
find that modules.conf is missing as well as flite in the libs
directory. I just did this fine yesterday morning. I download twice
today with the same results. Did something change? I want to set flite
on before running make, etc. And why is flite missing?
Mario


From mario_fs at mgtech.com  Sat Oct 23 11:36:05 2010
From: mario_fs at mgtech.com (Mario)
Date: Sat, 23 Oct 2010 11:36:05 -0700
Subject: [Freeswitch-users] Latest git missing files on osX?
In-Reply-To: <4CC3261C.50002@mgtech.com>
References: <4CC3261C.50002@mgtech.com>
Message-ID: <4CC32B15.4060406@mgtech.com>

I needed to do a ./bootstrap.sh in order for modules.conf to show up and
./configure && make for flite to show up.

On 10/23/10 11:14, Mario wrote:
> I download a fresh copy today to put in the final installation and I
> find that modules.conf is missing as well as flite in the libs
> directory. I just did this fine yesterday morning. I download twice
> today with the same results. Did something change? I want to set flite
> on before running make, etc. And why is flite missing?
> Mario
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From steveayre at gmail.com  Sat Oct 23 12:40:35 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 23 Oct 2010 20:40:35 +0100
Subject: [Freeswitch-users] Latest git missing files on osX?
In-Reply-To: <4CC32B15.4060406@mgtech.com>
References: <4CC3261C.50002@mgtech.com>
	<4CC32B15.4060406@mgtech.com>
Message-ID: <AANLkTimZYC=tDQ0mkF=yH5LZdhiQuXXgdJNqeCVkQYYv@mail.gmail.com>

Yes, that's normal.

-Steve


On 23 October 2010 19:36, Mario <mario_fs at mgtech.com> wrote:
> I needed to do a ./bootstrap.sh in order for modules.conf to show up and
> ./configure && make for flite to show up.
>
> On 10/23/10 11:14, Mario wrote:
>> I download a fresh copy today to put in the final installation and I
>> find that modules.conf is missing as well as flite in the libs
>> directory. I just did this fine yesterday morning. I download twice
>> today with the same results. Did something change? I want to set flite
>> on before running make, etc. And why is flite missing?
>> Mario
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From garrison at codefix.net  Sat Oct 23 14:24:43 2010
From: garrison at codefix.net (Garrison Hoffman)
Date: Sat, 23 Oct 2010 17:24:43 -0400
Subject: [Freeswitch-users] Hangup loopback/voicemail-b [CS_EXECUTE]
	[INCOMPATIBLE_DESTINATION]
Message-ID: <4CC3529B.1010901@codefix.net>

Hi all,

I've been updating my FS build after having been occupied with other 
things a while; I think I've caught up with the module changes (I'm now 
able to compile & install from the latest GIT) but voicemail fails with 
INCOMPATIBLE_DESTINATION.

Looking at my logs ( http://pastebin.freeswitch.org/14302 ) I don't see 
a clear indication of where the problem lies ? what am I missing?

-gh


From jeff at jefflenk.com  Sat Oct 23 14:49:04 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sat, 23 Oct 2010 14:49:04 -0700 (PDT)
Subject: [Freeswitch-users] Hangup loopback/voicemail-b [CS_EXECUTE]
 [INCOMPATIBLE_DESTINATION]
In-Reply-To: <4CC3529B.1010901@codefix.net>
References: <4CC3529B.1010901@codefix.net>
Message-ID: <1287870544404-5666483.post@n2.nabble.com>


see http://jira.freeswitch.org/browse/FS-2795 for a temporary fix
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Hangup-loopback-voicemail-b-CS-EXECUTE-INCOMPATIBLE-DESTINATION-tp5666449p5666483.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jeff at jefflenk.com  Sat Oct 23 15:17:11 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sat, 23 Oct 2010 15:17:11 -0700 (PDT)
Subject: [Freeswitch-users] Using the bridge | operator for Implementing
	Failover
In-Reply-To: <AANLkTikomdMk_s3GovN_gkxm6u2U-_p3Y67Jw6bHLLrt@mail.gmail.com>
References: <AANLkTimZonOAnpKJKyy24ZnyJD9HnDPYMju40MYL6CvH@mail.gmail.com>
	<AANLkTimoE7t0LQwK3Nmv8Km-KSHHAtk+n-Teeiv7zhoQ@mail.gmail.com>
	<AANLkTimfXA70zUQKK0cBoKyvjE2OhfaaxckTPMgpYwHu@mail.gmail.com>
	<AANLkTimT0QTVcJKkczFnuwQJMBmgdfjuG7vr6UxxAx0k@mail.gmail.com>
	<AANLkTikomdMk_s3GovN_gkxm6u2U-_p3Y67Jw6bHLLrt@mail.gmail.com>
Message-ID: <1287872231993-5666526.post@n2.nabble.com>


what happens with  <action application="set"
data="fail_on_single_reject=true"/>

-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Using-the-bridge-operator-for-Implementing-Failover-tp5661010p5666526.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From covici at ccs.covici.com  Sat Oct 23 15:59:14 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sat, 23 Oct 2010 18:59:14 -0400
Subject: [Freeswitch-users] problem with session timers (I think)
In-Reply-To: <AANLkTikDy=BnTbuonjLASJ4xTvyzeRXEnZJaPkEqduRA@mail.gmail.com>
References: <13102.1287800957@ccs.covici.com>
	<AANLkTinz9b6Ep4YHOOqfe41Q2BqeRS7buXCcOvRD_cfu@mail.gmail.com>
	<17763.1287830253@ccs.covici.com>
	<AANLkTikDy=BnTbuonjLASJ4xTvyzeRXEnZJaPkEqduRA@mail.gmail.com>
Message-ID: <30176.1287874754@ccs.covici.com>

I think I fixed it by putting enable-timer value="false" -- I am not
sure why this works where making the session-timeout large does not.
Very strange.

Thanks for your suggestions, I will keep them in mind for the future.

Steven Ayre <steveayre at gmail.com> wrote:

> Are you able to post the log, at debug level, with sip trace on and
> and with context (lines above the disconnect)?
> 
> Also, is there media flowing? Some SIP servers will disconnect a
> connection if there's been no media for a certain amount of time. FS
> has that option but it's off by default. In a bypass media situation
> that could cause problems.
> 
> NAT is another possibility. Some routers will disconnect a connection
> that hasn't had any activity for a while.
> 
> -Steve
> 
> 
> On 23 October 2010 11:37,  <covici at ccs.covici.com> wrote:
> > It just says NORMAL_CLEARiNG but its always after a half hour, so I
> > thought it was session timers. ?Could there be some other reason?
> >
> > Steven Ayre <steveayre at gmail.com> wrote:
> >
> >> Do you have a debug-level log of when FS hangs the call up? It should
> >> indicate why the call's hung up.
> >>
> >> There's several things it could be - hangup from the other end, a
> >> requested hangup on FS, session timers, (lack of) RTP timer...
> >>
> >> -Steve
> >>
> >>
> >> On 23 October 2010 03:29, ?<covici at ccs.covici.com> wrote:
> >> > Hi. ?When I call a certain number, using fs and portaudio, fs hangs up
> >> > in just about a half hour. ?This is suspiciously like a session timer or
> >> > something, so I added the parameter <param name="session-timeout"
> >> > value="12600"/> ?but it is still happening. ?I did not change the
> >> > internal profile, but I did change the external one in the client which
> >> > registers with the internal profile -- fs is the client as well as the
> >> > pbx.
> >> >
> >> > Any assistance on these would be appreciated.
> >> > --
> >> > Your life is like a penny. ?You're going to lose it. ?The question is:
> >> > How do
> >> > you spend it?
> >> >
> >> > ? ? ? ? John Covici
> >> > ? ? ? ? covici at ccs.covici.com
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > --
> > Your life is like a penny. ?You're going to lose it. ?The question is:
> > How do
> > you spend it?
> >
> > ? ? ? ? John Covici
> > ? ? ? ? covici at ccs.covici.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From garrison at codefix.net  Sat Oct 23 19:12:07 2010
From: garrison at codefix.net (Garrison Hoffman)
Date: Sat, 23 Oct 2010 22:12:07 -0400
Subject: [Freeswitch-users] Hangup loopback/voicemail-b [CS_EXECUTE]
	[INCOMPATIBLE_DESTINATION]
In-Reply-To: <1287870544404-5666483.post@n2.nabble.com>
References: <4CC3529B.1010901@codefix.net>
	<1287870544404-5666483.post@n2.nabble.com>
Message-ID: <4CC395F7.3020204@codefix.net>

Thanks. I added a patch that gets vm to answer (really just commented 
out one line), but I've not yet poked around to see if it breaks 
anything vital.

Is anyone maintaining a list of (relatively) stable git revisions?

-gh


From jeff at jefflenk.com  Sat Oct 23 20:50:24 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sat, 23 Oct 2010 20:50:24 -0700 (PDT)
Subject: [Freeswitch-users] Hangup loopback/voicemail-b [CS_EXECUTE]
 [INCOMPATIBLE_DESTINATION]
In-Reply-To: <4CC395F7.3020204@codefix.net>
References: <4CC3529B.1010901@codefix.net>
	<1287870544404-5666483.post@n2.nabble.com>
	<4CC395F7.3020204@codefix.net>
Message-ID: <1287892224127-5666928.post@n2.nabble.com>


git head is almost always the best version due to the tireless efforts of the
fs team! Please have patience as the infrequent problems that due occur are
indentified and corrected very quickly.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Hangup-loopback-voicemail-b-CS-EXECUTE-INCOMPATIBLE-DESTINATION-tp5666449p5666928.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From curriegrad2004 at gmail.com  Sat Oct 23 23:30:35 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Sat, 23 Oct 2010 23:30:35 -0700
Subject: [Freeswitch-users] [Linksys 3102] Couple of issues using FS and
	XLite
In-Reply-To: <1287849623835-5665744.post@n2.nabble.com>
References: <1287755399019-5662612.post@n2.nabble.com>
	<AANLkTin2q7o1DHp0tUguHiVMr1aJxLo4ryoedAHgW18_@mail.gmail.com>
	<1287759334654-5662810.post@n2.nabble.com>
	<AANLkTi=R6MHHUWHpx64MM4K3waTZjMF3Cfe+4goo5qnH@mail.gmail.com>
	<AANLkTimkLpGj7i3GcsYP8LZ8FnABTvz+Bmjb_JOqmkwB@mail.gmail.com>
	<1287838167405-5665332.post@n2.nabble.com>
	<1287839185519-5665357.post@n2.nabble.com>
	<4CC2E2AC.7090301@puzzled.xs4all.nl>
	<AANLkTi=ZH0vjD_sJeWoJ5WC7ZbRSs_rKgQhtZ7cAELqO@mail.gmail.com>
	<1287849623835-5665744.post@n2.nabble.com>
Message-ID: <AANLkTinozzYj7Mskpp+ezn5z2aY0VcuMLB-GwWPAtEYE@mail.gmail.com>

For some people who are complaining about MU, I rebuilt the git
sources again today. This time it's in a 7-Zip SFX archive

http://mysites.bcit.ca/A00780375/FreeSwitch_git_20101023.exe

This file can disappear anytime soon. I'm giving this a month or until
my school gets angry at me for hosting this file.

On Sat, Oct 23, 2010 at 9:00 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
>
> mod_dingaling does not have tls support under windows at this time. This is
> an artifact of not having support for gnutls under windows in the project -
> patches welcome.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Linksys-3102-Couple-of-issues-using-FS-and-XLite-tp5662612p5665744.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From dujinfang at gmail.com  Sun Oct 24 04:44:06 2010
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 24 Oct 2010 19:44:06 +0800
Subject: [Freeswitch-users] encrypt gateway password in FSComm
Message-ID: <AANLkTim6eEZb89uWKh=nsGXLUUrhgAbY6wiU_EZoV+0r@mail.gmail.com>

Hi,

Current FSComm using XML to store gateway information, is there a way
to encrypt password?  It would be  great if it can do A1 hash like
user directory.

I remember there's a mod (mod_qsettings ? I forgot the name) in FSComm
which been removed, should we get it back to support encrypted
password?

Thanks.

-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From jan.berger at video24.no  Sun Oct 24 06:07:43 2010
From: jan.berger at video24.no (Jan Berger)
Date: Sun, 24 Oct 2010 15:07:43 +0200
Subject: [Freeswitch-users] FS Integrator needed
Message-ID: <2DE88A91A9A34BA8B046226C470486C5@dell9400>

Hi folks,

 

I am looking for a developer willing to take on a task to integrate FS with
the CCXML/VXML engine on www.video24.no <http://www.video24.no/>  

 

Your first task will be to map the 2 API's - this is more documentation than
actual work - I will suggest you just write a small module and start testing
one even at the time. CCXML/VXML requires a list of telecom-events that must
be supported in FS. A majority of the events exist, but some do not.

 

So - this is in "theory" a small task of mapping events between 2 API's -
with some extras

 

Jan

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From mnhassan at usa.net  Sun Oct 24 06:31:00 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Sun, 24 Oct 2010 19:31:00 +0600
Subject: [Freeswitch-users] [Freeswitch-dev] FS Integrator needed
In-Reply-To: <2DE88A91A9A34BA8B046226C470486C5@dell9400>
References: <2DE88A91A9A34BA8B046226C470486C5@dell9400>
Message-ID: <AANLkTi=ed1=pnBVpk0_GFtCTtw60qSYw7VOgNQapao7B@mail.gmail.com>

Best help would probably be to contact consulting at freeswitch.org

Regards
HASSAN


On 2010-10-24, Jan Berger <jan.berger at video24.no> wrote:
> Hi folks,
>
>
>
> I am looking for a developer willing to take on a task to integrate FS with
> the CCXML/VXML engine on www.video24.no <http://www.video24.no/>
>
>
>
> Your first task will be to map the 2 API's - this is more documentation than
> actual work - I will suggest you just write a small module and start testing
> one even at the time. CCXML/VXML requires a list of telecom-events that must
> be supported in FS. A majority of the events exist, but some do not.
>
>
>
> So - this is in "theory" a small task of mapping events between 2 API's -
> with some extras
>
>
>
> Jan
>
>

-- 
Sent from my mobile device


From pjintheusa at gmail.com  Sun Oct 24 08:09:43 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sun, 24 Oct 2010 11:09:43 -0400
Subject: [Freeswitch-users] Using the bridge | operator for Implementing
	Failover
In-Reply-To: <1287872231993-5666526.post@n2.nabble.com>
References: <AANLkTimZonOAnpKJKyy24ZnyJD9HnDPYMju40MYL6CvH@mail.gmail.com>
	<AANLkTimoE7t0LQwK3Nmv8Km-KSHHAtk+n-Teeiv7zhoQ@mail.gmail.com>
	<AANLkTimfXA70zUQKK0cBoKyvjE2OhfaaxckTPMgpYwHu@mail.gmail.com>
	<AANLkTimT0QTVcJKkczFnuwQJMBmgdfjuG7vr6UxxAx0k@mail.gmail.com>
	<AANLkTikomdMk_s3GovN_gkxm6u2U-_p3Y67Jw6bHLLrt@mail.gmail.com>
	<1287872231993-5666526.post@n2.nabble.com>
Message-ID: <AANLkTi=JfTPXxAwyPPJ1CuKdmJ1pcys=18EE3apW9D1b@mail.gmail.com>

Jeff - Thanks for the response.

<condition field="destination_number" expression="^(3478309992)$">
      <action application="answer"/>
      <action application="sleep" data="2000"/>
       <!--action application="set"
data="fail_on_single_reject=NO_ANSWER,NORMAL_CLEARING,ALLOTTED_TIMEOUT"/-->
       <action application="set" data="fail_on_single_reject=true"/>
       <action application="bridge"
data="{ignore_early_media=true,originate_continue_on_timeout=false}[leg_timeout=10]sofia/gateway/broadvox1/2158824374|[leg_timeout=25]sofia/gateway/broadvox2/2158824374"/>
 </condition>
This gives exactly the same result. The 2158824374 is called by each gateway
specified.

Pj
On Sat, Oct 23, 2010 at 6:17 PM, Jeff Lenk <jeff at jefflenk.com> wrote:

>
> what happens with  <action application="set"
> data="fail_on_single_reject=true"/>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Using-the-bridge-operator-for-Implementing-Failover-tp5661010p5666526.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From codecomplete at free.fr  Sun Oct 24 09:25:27 2010
From: codecomplete at free.fr (GillesToo)
Date: Sun, 24 Oct 2010 09:25:27 -0700 (PDT)
Subject: [Freeswitch-users] [Linksys 3102] Couple of issues using FS and
	XLite
In-Reply-To: <AANLkTikX0QnxAQhrp=3qJoEKbZibb8J7NqxmhWDY21Hr@mail.gmail.com>
References: <1287755399019-5662612.post@n2.nabble.com>
	<AANLkTin2q7o1DHp0tUguHiVMr1aJxLo4ryoedAHgW18_@mail.gmail.com>
	<1287759334654-5662810.post@n2.nabble.com>
	<AANLkTi=R6MHHUWHpx64MM4K3waTZjMF3Cfe+4goo5qnH@mail.gmail.com>
	<AANLkTimkLpGj7i3GcsYP8LZ8FnABTvz+Bmjb_JOqmkwB@mail.gmail.com>
	<1287838167405-5665332.post@n2.nabble.com>
	<1287839185519-5665357.post@n2.nabble.com>
	<AANLkTikX0QnxAQhrp=3qJoEKbZibb8J7NqxmhWDY21Hr@mail.gmail.com>
Message-ID: <1287937527437-5668016.post@n2.nabble.com>



Milena-2-2 wrote:
> echo extension is not 9999 anymore

Thanks, problem solved. I didn't know the extensions changed in 1.0.6.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Linksys-3102-Couple-of-issues-using-FS-and-XLite-tp5662612p5668016.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From codecomplete at free.fr  Sun Oct 24 09:45:26 2010
From: codecomplete at free.fr (GillesToo)
Date: Sun, 24 Oct 2010 09:45:26 -0700 (PDT)
Subject: [Freeswitch-users] [Linksys 3102] Couple of issues using FS and
	XLite
In-Reply-To: <1287937527437-5668016.post@n2.nabble.com>
References: <1287755399019-5662612.post@n2.nabble.com>
	<AANLkTin2q7o1DHp0tUguHiVMr1aJxLo4ryoedAHgW18_@mail.gmail.com>
	<1287759334654-5662810.post@n2.nabble.com>
	<AANLkTi=R6MHHUWHpx64MM4K3waTZjMF3Cfe+4goo5qnH@mail.gmail.com>
	<AANLkTimkLpGj7i3GcsYP8LZ8FnABTvz+Bmjb_JOqmkwB@mail.gmail.com>
	<1287838167405-5665332.post@n2.nabble.com>
	<1287839185519-5665357.post@n2.nabble.com>
	<AANLkTikX0QnxAQhrp=3qJoEKbZibb8J7NqxmhWDY21Hr@mail.gmail.com>
	<1287937527437-5668016.post@n2.nabble.com>
Message-ID: <1287938726241-5668067.post@n2.nabble.com>


Well, actually, while the extension issue is solved... the original issue
isn't: After configuring the 3102 to act as gateway to the POTS, when
running XLite to call my cellphone, XLite says "Call established" although
the 3102 is still dialing and I haven't picked up the call yet.

Does someone know if this is an issue in XLite proper, or some setting in
Freeswitch?

Thank you.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Linksys-3102-Couple-of-issues-using-FS-and-XLite-tp5662612p5668067.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rupa at rupa.com  Sun Oct 24 10:38:16 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 24 Oct 2010 12:38:16 -0500
Subject: [Freeswitch-users] [Linksys 3102] Couple of issues using FS and
	XLite
In-Reply-To: <1287938726241-5668067.post@n2.nabble.com>
References: <1287755399019-5662612.post@n2.nabble.com>
	<AANLkTin2q7o1DHp0tUguHiVMr1aJxLo4ryoedAHgW18_@mail.gmail.com>
	<1287759334654-5662810.post@n2.nabble.com>
	<AANLkTi=R6MHHUWHpx64MM4K3waTZjMF3Cfe+4goo5qnH@mail.gmail.com>
	<AANLkTimkLpGj7i3GcsYP8LZ8FnABTvz+Bmjb_JOqmkwB@mail.gmail.com>
	<1287838167405-5665332.post@n2.nabble.com>
	<1287839185519-5665357.post@n2.nabble.com>
	<AANLkTikX0QnxAQhrp=3qJoEKbZibb8J7NqxmhWDY21Hr@mail.gmail.com>
	<1287937527437-5668016.post@n2.nabble.com>
	<1287938726241-5668067.post@n2.nabble.com>
Message-ID: <AANLkTinHUTDvixgjBQ6uw_RViK+f4nRcp_2MhRC000Rn@mail.gmail.com>

This is a "problem" with analog gateways in general.  The 3102 has no way to
distinguish between early media (ringing) and call established.  So it just
skips the whole early media stage.

On Sun, Oct 24, 2010 at 11:45 AM, GillesToo <codecomplete at free.fr> wrote:

>
> Well, actually, while the extension issue is solved... the original issue
> isn't: After configuring the 3102 to act as gateway to the POTS, when
> running XLite to call my cellphone, XLite says "Call established" although
> the 3102 is still dialing and I haven't picked up the call yet.
>
> Does someone know if this is an issue in XLite proper, or some setting in
> Freeswitch?
>
> Thank you.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Linksys-3102-Couple-of-issues-using-FS-and-XLite-tp5662612p5668067.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From nazim.aghabayov at gmail.com  Sun Oct 24 13:44:51 2010
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Mon, 25 Oct 2010 01:44:51 +0500
Subject: [Freeswitch-users] Radius AAA
In-Reply-To: <COL109-W27AC0F47F5E5A5F4404B4DF45F0@phx.gbl>
References: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>,
	<AANLkTi=GiQaO-48BP=T8c-A+09sJMKzQH7BX=ewBdB8R@mail.gmail.com>,
	<AANLkTikw8LGDqQki16D30b8JL0eJid2UOTs-bEaZSc2k@mail.gmail.com>,
	<4CADC2E4.90608@gmail.com>,
	<COL109-W528C7DF8FD0E4D203FD866F4570@phx.gbl>,
	<AANLkTi=oXjajMBSn8tJCT8chm6mM8E4XVzXVynDda_+O@mail.gmail.com>,
	<AANLkTi=P1QH2L5B-HckXvqRgHsQEpEia2X7mA_DDT=DN@mail.gmail.com>,
	<COL109-W43A18E93B3B89A7D57AE61F45B0@phx.gbl>,
	<AANLkTi=Cgi4LjDTfZAzvJOSU0u1+OWMsewa=-KOU1JeZ@mail.gmail.com>
	<COL109-W27AC0F47F5E5A5F4404B4DF45F0@phx.gbl>
Message-ID: <4CC49AC3.3060504@gmail.com>

Hi! Could yu please check if module is actually loaded?

freeswitch at internal> module_exists mod_rad_auth
true



On 10/23/2010 07:01 PM, Abid Saleem wrote:
> Dear Tihomir,
> Thanks a lot. It is loaded now but when I try to trigger auth in the dialplan, the module is not being called at all. I am using the example config as provided by you and Nazim.
> Scenerio is that I call from 1000 to 2000. In dialplan for destination_number 2000, I call the auth_function Application as said by Nazim but it is not being called at all. I have no clue why. Please help.
> Thanks.
> Regards---------------Abid Saleem
>
> Date: Tue, 19 Oct 2010 15:46:19 +0200
> From: tculjaga at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Radius AAA
>
>
> [root at terminus lib]# ln -sf /usr/local/lib/libfreeradius-client.so.2 libfreeradius-client.so
>
>
> try running this this:
> ln -sf /usr/local/lib/libfreeradius-client.so.2 libfreeradius-client.so.2
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 		 	   		
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From Nabble at slickdeals.endjunk.com  Sun Oct 24 14:09:56 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 24 Oct 2010 14:09:56 -0700 (PDT)
Subject: [Freeswitch-users] Need help on regex
Message-ID: <1287954596503-5668666.post@n2.nabble.com>


I am not much of a regex person but am looking for some help on crafting a
simple regex filter that can do the following criterias:

1. The dialed number is always a 10-digits number, i.e. an area code +
7-digits number (without any leading digit(s) of +1 and/or 1).

2. If the 4-th digit of the dialed number (after the area code) is a
non-zero number, do nothing. Otherwise, the regex filter will strip the
leading 3 digits (area code) as well as the last 2 digits number and this
will leave only 1-st 5-digits number (no area code). For instance, if the
10-digits dialed number is 0120123456, then the regex filter will return
01234. More examples followed:

10-digits input number     output number
---------------------      -------------
    4120123456                01234
    0121234567                0121234567
    1310156434                01564
    8231234567                8231234567
    0126543210                0126543210
Anyone? Thanks.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Need-help-on-regex-tp5668666p5668666.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From gmaruzz at gmail.com  Sun Oct 24 14:32:24 2010
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sun, 24 Oct 2010 23:32:24 +0200
Subject: [Freeswitch-users] Help needed about building snd_dummy with
 the dummy.c in skypopen
In-Reply-To: <AANLkTik6aj0dRqL4BqkRgT9vvp+7OgQtWT=abJ0F2EZ5@mail.gmail.com>
References: <AANLkTikQ_vj3Ue1MYs7x1pVj8kiCgFbeKBTL=97PddX=@mail.gmail.com>
	<AANLkTik5BEvEi=P_+cEL2kwjfrTQYLGh7MNtF14-8ROE@mail.gmail.com>
	<AANLkTikkXGL7H=1Sfhnzub454ob4MeCe7asoxXJi9myT@mail.gmail.com>
	<AANLkTiku3YBi004AZrC_-vW_Kc01x4NQbR3rtf_yxeNT@mail.gmail.com>
	<AANLkTik6aj0dRqL4BqkRgT9vvp+7OgQtWT=abJ0F2EZ5@mail.gmail.com>
Message-ID: <AANLkTi=L1ro64h=CBqQ_EUgw8exukt7yno31Fc_O2FTW@mail.gmail.com>

On Sun, Oct 24, 2010 at 11:28 PM, Jian Ren <renjian at gmail.com> wrote:
> Ok, I found out the new error was caused by missing of?32bit compatibility
> libraries, after I ran this:
>
> apt-get -y install ia32-libs lib32asound2 libc6-i386 lib32gcc1 \
> lib32stdc++6 lib32ncurses5 lib32z1

Btw, in the wiki is written about the 32bit libraries.

If you follow the wiki, it works. I've done that many times.

After you have installed ALSA, you need to reboot the machine, or to
rmmod all the snd* modules from the kernel, before to modprobe
snd-dummy

-giovanni


>
> Now I am getting exactly the same problem as if under virtualBox or
> VMPlayer. So it's not a VM problem.
> Please help!
> Thanks!
> Jian
>
> On Sat, Oct 23, 2010 at 10:19 PM, Jian Ren <renjian at gmail.com> wrote:
>>
>> Hi, I tried real machine. Got different error:
>> FATAL: Error inserting
>> snd_dummy(/lib/modules/2.6.24-26-server/kernel/sound/drivers/snd-dummy.ko):
>> Unknown symbol in module, or unknown parameter (see demsg).
>> Then if I type demsg, got the attached screen.
>> Please help.
>> Thanks!
>>
>> On Fri, Oct 22, 2010 at 11:33 AM, Jian Ren <renjian at gmail.com> wrote:
>>>
>>> Tried under VMPlayer, got the same problem, will try real machine later.
>>> Attached is the screen shot.
>>>
>>> Thanks!
>>> Jian
>>>
>>> On Thu, Oct 21, 2010 at 4:45 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>>> wrote:
>>>>
>>>> Maybe is a problem with virtualbox. Also, is known that
>>>> fs+mod_skypopen+skype clients does not works in virtualbox.
>>>>
>>>> Try it on a real (hardware) machine.
>>>>
>>>> Or (but is less popular) in a xen like virtual machine.
>>>>
>>>> -giovanni
>>>>
>>>> On Thu, Oct 21, 2010 at 4:21 PM, Jian Ren <renjian at gmail.com> wrote:
>>>> > Hi,
>>>> > I am trying to follow this wiki:
>>>> > http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>> > to setup FS and Skypopen on Ubuntu 8.04(64bit server inside
>>>> > virualbox).
>>>> >
>>>> > There is one step asks me to build snd_dummy:
>>>> >
>>>> > http://wiki.freeswitch.org/wiki/Skypopen_Skype_Endpoint_and_Trunk#ALSA_and_the_custom_snd-dummy
>>>> > The section is like below:
>>>> >
>>>> > Ubuntu, Debian
>>>> >
>>>> > Note, Ubuntu may have alsa sound drivers installed at
>>>> > /lib/modules/<your
>>>> > kernel version>/ubuntu/sound/..., you may need to remove the dir to
>>>> > allow
>>>> > modprobe search from the default place: /lib/modules/<your kernel
>>>> > verision>/kernel/sound/ (don't forget to rerun /sbin/depmod after
>>>> > removing
>>>> > the old sound directory. Eg: /sbin/depmod -a 2.6.24-24-server).
>>>> >
>>>> > ./configure --with-redhat=no \
>>>> > --with-cards=dummy,usb-audio,hda-intel,hrtimer,rtctimer \
>>>> > --with-card-options=all
>>>> > make && make install
>>>> >
>>>> > After the first make && make install, copy
>>>> > mod_skypopen/configs/alsa/dummy.c
>>>> > to alsa-driver-1.0.20sound/drivers/dummy.c .
>>>> >
>>>> > make && make install #again?:)
>>>> >
>>>> > The problem is after I did all of them, when I entered modprobe
>>>> > snd_dummy,
>>>> > it always crashed and returned as "killed" on my terminal, I could see
>>>> > it
>>>> > dumped out a huge block or data(numbers, etc) on the host but don't
>>>> > understand the meaning. While if I used the original dummy.c inside
>>>> > alsa-driver-1.0.20, it worked fine.
>>>> > Besides, the default ubuntu installation doesn't include kernal dev
>>>> > and
>>>> > source, so I did one more step(or it cannot build alsa).
>>>> >
>>>> > Did anyone try the same and get it working?
>>>> >
>>>> > Thanks!
>>>> > Jian
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >
>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>>
>>>>
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> Cell : +39-347-2665618
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From steveayre at gmail.com  Sun Oct 24 15:51:37 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 24 Oct 2010 23:51:37 +0100
Subject: [Freeswitch-users] Need help on regex
In-Reply-To: <1287954596503-5668666.post@n2.nabble.com>
References: <1287954596503-5668666.post@n2.nabble.com>
Message-ID: <37A7ED80-F6C5-40CD-93F0-C728282E1628@gmail.com>

<extension name="zero">
  <condition field="destination_number" expression="^\d\d\d(0\d\d\d\d)\d\d$">
    <!-- $1 holds the number -->
  </condition>
</extension>
<extension name="nonzero">
  <condition field="destination_number" expression="^(\d\d\d[123456789]\d\d\d\d\d\d)$">
    <!-- $1 holds the number -->
  </condition>
</extension>

Steve on iPhone

On 24 Oct 2010, at 22:09, mazilo <Nabble at slickdeals.endjunk.com> wrote:

> 
> I am not much of a regex person but am looking for some help on crafting a
> simple regex filter that can do the following criterias:
> 
> 1. The dialed number is always a 10-digits number, i.e. an area code +
> 7-digits number (without any leading digit(s) of +1 and/or 1).
> 
> 2. If the 4-th digit of the dialed number (after the area code) is a
> non-zero number, do nothing. Otherwise, the regex filter will strip the
> leading 3 digits (area code) as well as the last 2 digits number and this
> will leave only 1-st 5-digits number (no area code). For instance, if the
> 10-digits dialed number is 0120123456, then the regex filter will return
> 01234. More examples followed:
> 
> 10-digits input number     output number
> ---------------------      -------------
>    4120123456                01234
>    0121234567                0121234567
>    1310156434                01564
>    8231234567                8231234567
>    0126543210                0126543210
> Anyone? Thanks.
> 
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> -- 
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Need-help-on-regex-tp5668666p5668666.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From mnhassan at usa.net  Sun Oct 24 16:04:58 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Mon, 25 Oct 2010 05:04:58 +0600
Subject: [Freeswitch-users] Need help on regex
In-Reply-To: <37A7ED80-F6C5-40CD-93F0-C728282E1628@gmail.com>
References: <1287954596503-5668666.post@n2.nabble.com>
	<37A7ED80-F6C5-40CD-93F0-C728282E1628@gmail.com>
Message-ID: <AANLkTikDk9uX49G8umGsWMen7i6xu-3s97gQpDeAOxjt@mail.gmail.com>

Apart from the numerous regex help on the net, I have found a very
good helper to the open source text editor Notepad++, which got me out
of my regex woes a couple of months ago.

Regards
HASSAN


On 2010-10-25, Steven Ayre <steveayre at gmail.com> wrote:
> <extension name="zero">
>   <condition field="destination_number"
> expression="^\d\d\d(0\d\d\d\d)\d\d$">
>     <!-- $1 holds the number -->
>   </condition>
> </extension>
> <extension name="nonzero">
>   <condition field="destination_number"
> expression="^(\d\d\d[123456789]\d\d\d\d\d\d)$">
>     <!-- $1 holds the number -->
>   </condition>
> </extension>
>
> Steve on iPhone
>
> On 24 Oct 2010, at 22:09, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
>>
>> I am not much of a regex person but am looking for some help on crafting a
>> simple regex filter that can do the following criterias:
>>
>> 1. The dialed number is always a 10-digits number, i.e. an area code +
>> 7-digits number (without any leading digit(s) of +1 and/or 1).
>>
>> 2. If the 4-th digit of the dialed number (after the area code) is a
>> non-zero number, do nothing. Otherwise, the regex filter will strip the
>> leading 3 digits (area code) as well as the last 2 digits number and this
>> will leave only 1-st 5-digits number (no area code). For instance, if the
>> 10-digits dialed number is 0120123456, then the regex filter will return
>> 01234. More examples followed:
>>
>> 10-digits input number     output number
>> ---------------------      -------------
>>    4120123456                01234
>>    0121234567                0121234567
>>    1310156434                01564
>>    8231234567                8231234567
>>    0126543210                0126543210
>> Anyone? Thanks.
>>
>> -----
>> don't and stop are the ONLY two 4-letter words considered offensive to
>> men,
>> but not when used together.
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Need-help-on-regex-tp5668666p5668666.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device


From Nabble at slickdeals.endjunk.com  Sun Oct 24 18:39:07 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 24 Oct 2010 18:39:07 -0700 (PDT)
Subject: [Freeswitch-users] Need help on regex
In-Reply-To: <37A7ED80-F6C5-40CD-93F0-C728282E1628@gmail.com>
References: <1287954596503-5668666.post@n2.nabble.com>
	<37A7ED80-F6C5-40CD-93F0-C728282E1628@gmail.com>
Message-ID: <1287970747309-5669081.post@n2.nabble.com>



Steven Ayre wrote:
> 
> <extension name="zero">
>   <condition field="destination_number"
> expression="^\d\d\d(0\d\d\d\d)\d\d$">
>     <!-- $1 holds the number -->
>   </condition>
> </extension>
> <extension name="nonzero">
>   <condition field="destination_number"
> expression="^(\d\d\d[123456789]\d\d\d\d\d\d)$">
>     <!-- $1 holds the number -->
>   </condition>
> </extension>
> 
> Steve on iPhone
Steven, thank you for your quick response. I reckon the \d\d\d is the same
as \d{3} such that above expressions can be rewritten as
expression="^\d{3}(0\d{4})\d{2}$" and expression="^(\d{3}[1-9]\d{6})$",
respectively. If so, is it possible to combine the two regex expressions
into a single expression using pipe within one dialplan? I tried
expression="^\d{3}(0\d{4})\d{2}$|^(\d{3}[1-9]\d{6})$" and it only works for
the 1st regex while the 2nd regex gives a null.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Need-help-on-regex-tp5668666p5669081.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Russell.Mosemann at cune.org  Sun Oct 24 18:59:54 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sun, 24 Oct 2010 20:59:54 -0500
Subject: [Freeswitch-users] Need help on regex
In-Reply-To: <1287970747309-5669081.post@n2.nabble.com>
References: <1287954596503-5668666.post@n2.nabble.com><37A7ED80-F6C5-40CD-93F0-C728282E1628@gmail.com>
	<1287970747309-5669081.post@n2.nabble.com>
Message-ID: <98C6BDA8E2674AF7BF957C02803F0A15@cune.pri>

mazilo wrote:

> is it possible to combine the two regex expressions
> into a single expression using pipe within one dialplan? I tried
> expression="^\d{3}(0\d{4})\d{2}$|^(\d{3}[1-9]\d{6})$" and it only works
> for the 1st regex while the 2nd regex gives a null.

"^(?:\d{3}(0\d{4})\d{2}|(\d{3}[1-9]\d{6}))$"

Note that if the first value is captured, it will be in $1. The second captured value will be in $2.

This is described at http://wiki.freeswitch.org/wiki/Regular_Expression


mose



From dujinfang at gmail.com  Sun Oct 24 20:15:02 2010
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 25 Oct 2010 11:15:02 +0800
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on
	loopback/app=voicemail, default dialplan
Message-ID: <AANLkTim66YgK4oZZc5gw+d7RM9XDhW=FsnJv_bi2rSq+@mail.gmail.com>

Hi,

I'm on git code few days ago, and noticed this problem when I called
from 1000 to 1001 with default config.

http://pastebin.freeswitch.org/14306

Can someone help to take a look.

Also, why use loopback/app=voicemail instead of just the voicemail
app? advantage or just an example to use loopback ? I think the
loopback/app= is not documented on wiki, is it like inline dialplan?
Will document when get familiar.

thanks.



-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From codecomplete at free.fr  Sun Oct 24 21:41:12 2010
From: codecomplete at free.fr (GillesToo)
Date: Sun, 24 Oct 2010 21:41:12 -0700 (PDT)
Subject: [Freeswitch-users] [Linksys 3102] Couple of issues using FS and
	XLite
In-Reply-To: <AANLkTinHUTDvixgjBQ6uw_RViK+f4nRcp_2MhRC000Rn@mail.gmail.com>
References: <AANLkTin2q7o1DHp0tUguHiVMr1aJxLo4ryoedAHgW18_@mail.gmail.com>
	<1287759334654-5662810.post@n2.nabble.com>
	<AANLkTi=R6MHHUWHpx64MM4K3waTZjMF3Cfe+4goo5qnH@mail.gmail.com>
	<AANLkTimkLpGj7i3GcsYP8LZ8FnABTvz+Bmjb_JOqmkwB@mail.gmail.com>
	<1287838167405-5665332.post@n2.nabble.com>
	<1287839185519-5665357.post@n2.nabble.com>
	<AANLkTikX0QnxAQhrp=3qJoEKbZibb8J7NqxmhWDY21Hr@mail.gmail.com>
	<1287937527437-5668016.post@n2.nabble.com>
	<1287938726241-5668067.post@n2.nabble.com>
	<AANLkTinHUTDvixgjBQ6uw_RViK+f4nRcp_2MhRC000Rn@mail.gmail.com>
Message-ID: <1287981672573-5669365.post@n2.nabble.com>



Rupa Schomaker wrote:
> This is a "problem" with analog gateways in general.  The 3102 has no way
> to distinguish between early media (ringing) and call established.  So it
> just skips the whole early media stage.

Thanks Rupa. I'll see if XLite can be tweaked into displaying a different
string while it's waiting for a call to be actually connected.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Linksys-3102-Couple-of-issues-using-FS-and-XLite-tp5662612p5669365.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Mon Oct 25 01:00:04 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 25 Oct 2010 09:00:04 +0100
Subject: [Freeswitch-users] Need help on regex
In-Reply-To: <1287970747309-5669081.post@n2.nabble.com>
References: <1287954596503-5668666.post@n2.nabble.com>
	<37A7ED80-F6C5-40CD-93F0-C728282E1628@gmail.com>
	<1287970747309-5669081.post@n2.nabble.com>
Message-ID: <AANLkTimWM-Vfrpuf4ZwfbkwM2vh_2GN9yv4859s2TCb0@mail.gmail.com>

Yes, \d\d\d and \d{3} are equivalent... use whichever you prefer.

You can have both in a single regex, but since they won't be matched
within the same brackets they'll not be in the same variable.

^\d{3}(0\d{4})\d{2}$|^(\d{3}[1-9]\d{6})$
with a zero would be in $1, $2 will be null
and without a zero would be in $2, $1 will be null

So I can't think of a way to handle both in the same extension... You
could have both transfer to a third extension where the call is
handled though if you want to avoid duplicated actions in the
dialplan.

Unless FS supports named captures:
^\d{3}(?<num>0\d{4})\d{2}$|^(?<num>\d{3}[1-9]\d{6})$

In which case it's be stored in $num (?) in both cases, but I don't
know whether FS's implementation supports named capturing...

-Steve


On 25 October 2010 02:39, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
>
> Steven Ayre wrote:
>>
>> <extension name="zero">
>> ? <condition field="destination_number"
>> expression="^\d\d\d(0\d\d\d\d)\d\d$">
>> ? ? <!-- $1 holds the number -->
>> ? </condition>
>> </extension>
>> <extension name="nonzero">
>> ? <condition field="destination_number"
>> expression="^(\d\d\d[123456789]\d\d\d\d\d\d)$">
>> ? ? <!-- $1 holds the number -->
>> ? </condition>
>> </extension>
>>
>> Steve on iPhone
> Steven, thank you for your quick response. I reckon the \d\d\d is the same
> as \d{3} such that above expressions can be rewritten as
> expression="^\d{3}(0\d{4})\d{2}$" and expression="^(\d{3}[1-9]\d{6})$",
> respectively. If so, is it possible to combine the two regex expressions
> into a single expression using pipe within one dialplan? I tried
> expression="^\d{3}(0\d{4})\d{2}$|^(\d{3}[1-9]\d{6})$" and it only works for
> the 1st regex while the 2nd regex gives a null.
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Need-help-on-regex-tp5668666p5669081.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From steveayre at gmail.com  Mon Oct 25 01:10:46 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 25 Oct 2010 09:10:46 +0100
Subject: [Freeswitch-users] Need help on regex
In-Reply-To: <AANLkTimWM-Vfrpuf4ZwfbkwM2vh_2GN9yv4859s2TCb0@mail.gmail.com>
References: <1287954596503-5668666.post@n2.nabble.com>
	<37A7ED80-F6C5-40CD-93F0-C728282E1628@gmail.com>
	<1287970747309-5669081.post@n2.nabble.com>
	<AANLkTimWM-Vfrpuf4ZwfbkwM2vh_2GN9yv4859s2TCb0@mail.gmail.com>
Message-ID: <AANLkTimdxQNGuLxYZSH8YukYoT1b2J0nJFGVOUv9_MCQ@mail.gmail.com>

Actually, I've just realised there is a way:

^(?|\d{3}(0\d{4})\d{2}$|^(\d{3}[1-9]\d{6}))$

The (?| ) bracket resets the numbering for each alternative, so the
first set of brackets in each alternative both point to $1.

-Steve


On 25 October 2010 09:00, Steven Ayre <steveayre at gmail.com> wrote:
> Yes, \d\d\d and \d{3} are equivalent... use whichever you prefer.
>
> You can have both in a single regex, but since they won't be matched
> within the same brackets they'll not be in the same variable.
>
> ^\d{3}(0\d{4})\d{2}$|^(\d{3}[1-9]\d{6})$
> with a zero would be in $1, $2 will be null
> and without a zero would be in $2, $1 will be null
>
> So I can't think of a way to handle both in the same extension... You
> could have both transfer to a third extension where the call is
> handled though if you want to avoid duplicated actions in the
> dialplan.
>
> Unless FS supports named captures:
> ^\d{3}(?<num>0\d{4})\d{2}$|^(?<num>\d{3}[1-9]\d{6})$
>
> In which case it's be stored in $num (?) in both cases, but I don't
> know whether FS's implementation supports named capturing...
>
> -Steve
>
>
> On 25 October 2010 02:39, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>>
>>
>> Steven Ayre wrote:
>>>
>>> <extension name="zero">
>>> ? <condition field="destination_number"
>>> expression="^\d\d\d(0\d\d\d\d)\d\d$">
>>> ? ? <!-- $1 holds the number -->
>>> ? </condition>
>>> </extension>
>>> <extension name="nonzero">
>>> ? <condition field="destination_number"
>>> expression="^(\d\d\d[123456789]\d\d\d\d\d\d)$">
>>> ? ? <!-- $1 holds the number -->
>>> ? </condition>
>>> </extension>
>>>
>>> Steve on iPhone
>> Steven, thank you for your quick response. I reckon the \d\d\d is the same
>> as \d{3} such that above expressions can be rewritten as
>> expression="^\d{3}(0\d{4})\d{2}$" and expression="^(\d{3}[1-9]\d{6})$",
>> respectively. If so, is it possible to combine the two regex expressions
>> into a single expression using pipe within one dialplan? I tried
>> expression="^\d{3}(0\d{4})\d{2}$|^(\d{3}[1-9]\d{6})$" and it only works for
>> the 1st regex while the 2nd regex gives a null.
>>
>> -----
>> don't and stop are the ONLY two 4-letter words considered offensive to men,
>> but not when used together.
>> --
>> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Need-help-on-regex-tp5668666p5669081.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From davidwaf at gmail.com  Mon Oct 25 01:58:36 2010
From: davidwaf at gmail.com (David Wafula)
Date: Mon, 25 Oct 2010 10:58:36 +0200
Subject: [Freeswitch-users] Invalid Application hash
Message-ID: <AANLkTimX=5mruJzK683hdqOrjZ_unwTmVyWo3oNa601+@mail.gmail.com>

Hi all,
Notice the last line in the log below when i dial 1001. This is from the
latest git as of this morning. Any pointers?
thanks.

Dialplan: sofia/internal/1000 at 146.141.76.153 Regex (PASS) [Local_Extension]
destination_number(1001) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/1000 at 146.141.76.153 Action
set(dialed_extension=1001)
Dialplan: sofia/internal/1000 at 146.141.76.153 Action
export(dialed_extension=1001)
Dialplan: sofia/internal/1000 at 146.141.76.153 Action bind_meta_app(1 b s
execute_extension::dx XML features)
Dialplan: sofia/internal/1000 at 146.141.76.153 Action bind_meta_app(2 b s
record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav)

Dialplan: sofia/internal/1000 at 146.141.76.153 Action bind_meta_app(3 b s
execute_extension::cf XML features)
Dialplan: sofia/internal/1000 at 146.141.76.153 Action set(ringback=${us-ring})

Dialplan: sofia/internal/1000 at 146.141.76.153 Action
set(transfer_ringback=local_stream://moh)
Dialplan: sofia/internal/1000 at 146.141.76.153 Action set(call_timeout=30)
Dialplan: sofia/internal/1000 at 146.141.76.153 Action
set(hangup_after_bridge=true)
Dialplan: sofia/internal/1000 at 146.141.76.153 Action
set(continue_on_fail=true)
Dialplan: sofia/internal/1000 at 146.141.76.153 Action
hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})

Dialplan: sofia/internal/1000 at 146.141.76.153 Action
hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
Dialplan: sofia/internal/1000 at 146.141.76.153 Action
set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)})
Dialplan: sofia/internal/1000 at 146.141.76.153 Action
hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
Dialplan: sofia/internal/1000 at 146.141.76.153 Action
bridge(user/${dialed_extension}@${domain_name})
Dialplan: sofia/internal/1000 at 146.141.76.153 Action answer()
Dialplan: sofia/internal/1000 at 146.141.76.153 Action sleep(1000)
Dialplan: sofia/internal/1000 at 146.141.76.153 Action voicemail(default
${domain_name} ${dialed_extension})
2010-10-25 10:47:47.193206 [DEBUG] switch_core_state_machine.c:119
(sofia/internal/1000 at 146.141.76.153) State Change CS_ROUTING -> CS_EXECUTE
2010-10-25 10:47:47.193206 [DEBUG] switch_core_session.c:1057 Send signal
sofia/internal/1000 at 146.141.76.153 [BREAK]
2010-10-25 10:47:47.193206 [DEBUG] switch_core_state_machine.c:345
(sofia/internal/1000 at 146.141.76.153) State ROUTING going to sleep
2010-10-25 10:47:47.193206 [DEBUG] switch_core_state_machine.c:318
(sofia/internal/1000 at 146.141.76.153) Running State Change CS_EXECUTE
2010-10-25 10:47:47.193206 [DEBUG] switch_core_state_machine.c:352
(sofia/internal/1000 at 146.141.76.153) State EXECUTE
2010-10-25 10:47:47.193206 [DEBUG] mod_sofia.c:239 sofia/internal/
1000 at 146.141.76.153 SOFIA EXECUTE
2010-10-25 10:47:47.193206 [DEBUG] switch_core_state_machine.c:157
sofia/internal/1000 at 146.141.76.153 Standard EXECUTE
EXECUTE sofia/internal/1000 at 146.141.76.153 set(open=true)
2010-10-25 10:47:47.193206 [DEBUG] mod_dptools.c:1028 sofia/internal/
1000 at 146.141.76.153 SET [open]=[true]
2010-10-25 10:47:47.193206 [ERR] switch_core_session.c:1807 Invalid
Application hash


-- 
David Wafula
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From tculjaga at gmail.com  Mon Oct 25 03:28:49 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 25 Oct 2010 12:28:49 +0200
Subject: [Freeswitch-users] Radius AAA
In-Reply-To: <4CC49AC3.3060504@gmail.com>
References: <COL109-W362E7EF22E650EEC002FDDF46E0@phx.gbl>
	<AANLkTi=GiQaO-48BP=T8c-A+09sJMKzQH7BX=ewBdB8R@mail.gmail.com>
	<AANLkTikw8LGDqQki16D30b8JL0eJid2UOTs-bEaZSc2k@mail.gmail.com>
	<4CADC2E4.90608@gmail.com>
	<COL109-W528C7DF8FD0E4D203FD866F4570@phx.gbl>
	<AANLkTi=oXjajMBSn8tJCT8chm6mM8E4XVzXVynDda_+O@mail.gmail.com>
	<AANLkTi=P1QH2L5B-HckXvqRgHsQEpEia2X7mA_DDT=DN@mail.gmail.com>
	<COL109-W43A18E93B3B89A7D57AE61F45B0@phx.gbl>
	<AANLkTi=Cgi4LjDTfZAzvJOSU0u1+OWMsewa=-KOU1JeZ@mail.gmail.com>
	<COL109-W27AC0F47F5E5A5F4404B4DF45F0@phx.gbl>
	<4CC49AC3.3060504@gmail.com>
Message-ID: <AANLkTi=PLaZ8y0DQ=KeRzd4=gfRw2B+=+Zn1d9XerPz6@mail.gmail.com>

On Sun, Oct 24, 2010 at 10:44 PM, Nazim Aghabayov <nazim.aghabayov at gmail.com
> wrote:

> Hi! Could yu please check if module is actually loaded?
>
> freeswitch at internal> module_exists mod_rad_auth
> true
>
>
>
What do you mean by: "it is not being called at all" ?
Does it throw an error or what?

also, please can you paste your config (dialplan/rad_auth.conf) and yur lvl7
debug here ?
if its too large use pastebin.


T.
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From steveayre at gmail.com  Mon Oct 25 03:43:31 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 25 Oct 2010 11:43:31 +0100
Subject: [Freeswitch-users] Invalid Application hash
In-Reply-To: <AANLkTimX=5mruJzK683hdqOrjZ_unwTmVyWo3oNa601+@mail.gmail.com>
References: <AANLkTimX=5mruJzK683hdqOrjZ_unwTmVyWo3oNa601+@mail.gmail.com>
Message-ID: <AANLkTimShuxrmaNA6MQDipFsZqD9pxiJ1UOGOC7LZJH6@mail.gmail.com>

This is on the FAQ...
http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_Why_do_I_get_the_error_.22Invalid_Application_.3Cname.3E.22.3F

Are you upgrading and using an older config?

Limit handling was changed (quite a while ago now). mod_limit no
longer exists, it's been moved into the core and modules now provide
storage backends for it. The mod_limit module that now exists is more
of a stub really, and will log an error on startup telling you that
you need to update your config.

If you were loading mod_limit before, you should now load mod_hash and
mod_db. They should be built by default with latest git and loaded by
the example config, so I assume you're using an old config file.

Check conf/autoload_configs/modules.conf.xml contains:
<load module="mod_hash"/>

Regards,
-Steve


On 25 October 2010 09:58, David Wafula <davidwaf at gmail.com> wrote:
> Hi all,
> Notice the last line in the log below when i dial 1001. This is from the
> latest git as of this morning. Any pointers?
> thanks.
>
> Dialplan: sofia/internal/1000 at 146.141.76.153 Regex (PASS) [Local_Extension]
> destination_number(1001) =~ /^(10[01][0-9])$/ break=on-false
> Dialplan: sofia/internal/1000 at 146.141.76.153 Action
> set(dialed_extension=1001)
> Dialplan: sofia/internal/1000 at 146.141.76.153 Action
> export(dialed_extension=1001)
> Dialplan: sofia/internal/1000 at 146.141.76.153 Action bind_meta_app(1 b s
> execute_extension::dx XML features)
> Dialplan: sofia/internal/1000 at 146.141.76.153 Action bind_meta_app(2 b s
> record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav)
> Dialplan: sofia/internal/1000 at 146.141.76.153 Action bind_meta_app(3 b s
> execute_extension::cf XML features)
> Dialplan: sofia/internal/1000 at 146.141.76.153 Action set(ringback=${us-ring})
> Dialplan: sofia/internal/1000 at 146.141.76.153 Action
> set(transfer_ringback=local_stream://moh)
> Dialplan: sofia/internal/1000 at 146.141.76.153 Action set(call_timeout=30)
> Dialplan: sofia/internal/1000 at 146.141.76.153 Action
> set(hangup_after_bridge=true)
> Dialplan: sofia/internal/1000 at 146.141.76.153 Action
> set(continue_on_fail=true)
> Dialplan: sofia/internal/1000 at 146.141.76.153 Action
> hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})
> Dialplan: sofia/internal/1000 at 146.141.76.153 Action
> hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
> Dialplan: sofia/internal/1000 at 146.141.76.153 Action
> set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
> var callgroup)})
> Dialplan: sofia/internal/1000 at 146.141.76.153 Action
> hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
> Dialplan: sofia/internal/1000 at 146.141.76.153 Action
> bridge(user/${dialed_extension}@${domain_name})
> Dialplan: sofia/internal/1000 at 146.141.76.153 Action answer()
> Dialplan: sofia/internal/1000 at 146.141.76.153 Action sleep(1000)
> Dialplan: sofia/internal/1000 at 146.141.76.153 Action voicemail(default
> ${domain_name} ${dialed_extension})
> 2010-10-25 10:47:47.193206 [DEBUG] switch_core_state_machine.c:119
> (sofia/internal/1000 at 146.141.76.153) State Change CS_ROUTING -> CS_EXECUTE
> 2010-10-25 10:47:47.193206 [DEBUG] switch_core_session.c:1057 Send signal
> sofia/internal/1000 at 146.141.76.153 [BREAK]
> 2010-10-25 10:47:47.193206 [DEBUG] switch_core_state_machine.c:345
> (sofia/internal/1000 at 146.141.76.153) State ROUTING going to sleep
> 2010-10-25 10:47:47.193206 [DEBUG] switch_core_state_machine.c:318
> (sofia/internal/1000 at 146.141.76.153) Running State Change CS_EXECUTE
> 2010-10-25 10:47:47.193206 [DEBUG] switch_core_state_machine.c:352
> (sofia/internal/1000 at 146.141.76.153) State EXECUTE
> 2010-10-25 10:47:47.193206 [DEBUG] mod_sofia.c:239
> sofia/internal/1000 at 146.141.76.153 SOFIA EXECUTE
> 2010-10-25 10:47:47.193206 [DEBUG] switch_core_state_machine.c:157
> sofia/internal/1000 at 146.141.76.153 Standard EXECUTE
> EXECUTE sofia/internal/1000 at 146.141.76.153 set(open=true)
> 2010-10-25 10:47:47.193206 [DEBUG] mod_dptools.c:1028
> sofia/internal/1000 at 146.141.76.153 SET [open]=[true]
> 2010-10-25 10:47:47.193206 [ERR] switch_core_session.c:1807 Invalid
> Application hash
>
>
> --
> David Wafula
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From covici at ccs.covici.com  Mon Oct 25 04:05:56 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 25 Oct 2010 07:05:56 -0400
Subject: [Freeswitch-users] transfer problem
In-Reply-To: <AANLkTikh67cGuMn3v4D3QZ+AuQ0vWrLE_fRhL5+g1fg7@mail.gmail.com>
References: <13464.1287506974@ccs.covici.com>
	<AANLkTikh67cGuMn3v4D3QZ+AuQ0vWrLE_fRhL5+g1fg7@mail.gmail.com>
Message-ID: <19713.1288004756@ccs.covici.com>

OK, the debug log is here http://pastebin.freeswitch.org/14307.  The
relevant part of the configs is below.
    <extension name="conference_ivr">
      <condition field="destination_number" expression="^7000$">
        <action application="answer"/>
<action application="sleep" data="500"/>
<action application="set" data="api_hangup_hook=system sayconf -a local  -a ${caller_id_number} -a hung -a up"/>
	<action application="ivr" data="conference_ivr"/>
      </condition>
    </extension>
and the ivr just asks for passwords and send the call to an extension in
context conferences.

Hope this helps.

Thanks.

Michael Collins <msc at freeswitch.org> wrote:

> Please supply the configs and a debug trace. My guess is that there's
> probably just a simple config element that is wrong or missing.
> 
> -MC
> 
> On Tue, Oct 19, 2010 at 9:49 AM, <covici at ccs.covici.com> wrote:
> 
> > Hi.  I have an ivr which connects to conferences which I have in their
> > own context rather than the default.  The problem is that if I call
> > someone else and have a 3-way between the other party and the
> > conference, its fine till I hang up and then fs tries to transfer to the
> > correct conference name, but uses default for the context instead.  Is
> > this expected behavior or should I file a bug?
> >
> > Thanks.
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From davidwaf at gmail.com  Mon Oct 25 04:07:49 2010
From: davidwaf at gmail.com (David Wafula)
Date: Mon, 25 Oct 2010 13:07:49 +0200
Subject: [Freeswitch-users] Invalid Application hash
In-Reply-To: <AANLkTimShuxrmaNA6MQDipFsZqD9pxiJ1UOGOC7LZJH6@mail.gmail.com>
References: <AANLkTimX=5mruJzK683hdqOrjZ_unwTmVyWo3oNa601+@mail.gmail.com>
	<AANLkTimShuxrmaNA6MQDipFsZqD9pxiJ1UOGOC7LZJH6@mail.gmail.com>
Message-ID: <AANLkTimyis5WmY0D2L+6zUKdJbirGMM0viAuHDZ-N_Ev@mail.gmail.com>

Thank you, i upgraded ....
<load module="mod_hash"/>
did it..


On Mon, Oct 25, 2010 at 12:43 PM, Steven Ayre <steveayre at gmail.com> wrote:

> This is on the FAQ...
>
> http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_Why_do_I_get_the_error_.22Invalid_Application_.3Cname.3E.22.3F
>
> Are you upgrading and using an older config?
>
> Limit handling was changed (quite a while ago now). mod_limit no
> longer exists, it's been moved into the core and modules now provide
> storage backends for it. The mod_limit module that now exists is more
> of a stub really, and will log an error on startup telling you that
> you need to update your config.
>
> If you were loading mod_limit before, you should now load mod_hash and
> mod_db. They should be built by default with latest git and loaded by
> the example config, so I assume you're using an old config file.
>
> Check conf/autoload_configs/modules.conf.xml contains:
> <load module="mod_hash"/>
>
> Regards,
> -Steve
>
>
> On 25 October 2010 09:58, David Wafula <davidwaf at gmail.com> wrote:
> > Hi all,
> > Notice the last line in the log below when i dial 1001. This is from the
> > latest git as of this morning. Any pointers?
> > thanks.
> >
> > Dialplan: sofia/internal/1000 at 146.141.76.153 Regex (PASS)
> [Local_Extension]
> > destination_number(1001) =~ /^(10[01][0-9])$/ break=on-false
> > Dialplan: sofia/internal/1000 at 146.141.76.153 Action
> > set(dialed_extension=1001)
> > Dialplan: sofia/internal/1000 at 146.141.76.153 Action
> > export(dialed_extension=1001)
> > Dialplan: sofia/internal/1000 at 146.141.76.153 Action bind_meta_app(1 b s
> > execute_extension::dx XML features)
> > Dialplan: sofia/internal/1000 at 146.141.76.153 Action bind_meta_app(2 b s
> >
> record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav)
> > Dialplan: sofia/internal/1000 at 146.141.76.153 Action bind_meta_app(3 b s
> > execute_extension::cf XML features)
> > Dialplan: sofia/internal/1000 at 146.141.76.153 Action
> set(ringback=${us-ring})
> > Dialplan: sofia/internal/1000 at 146.141.76.153 Action
> > set(transfer_ringback=local_stream://moh)
> > Dialplan: sofia/internal/1000 at 146.141.76.153 Action set(call_timeout=30)
> > Dialplan: sofia/internal/1000 at 146.141.76.153 Action
> > set(hangup_after_bridge=true)
> > Dialplan: sofia/internal/1000 at 146.141.76.153 Action
> > set(continue_on_fail=true)
> > Dialplan: sofia/internal/1000 at 146.141.76.153 Action
> >
> hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})
> > Dialplan: sofia/internal/1000 at 146.141.76.153 Action
> > hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
> > Dialplan: sofia/internal/1000 at 146.141.76.153 Action
> > set(called_party_callgroup=${user_data(${dialed_extension}@
> ${domain_name}
> > var callgroup)})
> > Dialplan: sofia/internal/1000 at 146.141.76.153 Action
> > hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
> > Dialplan: sofia/internal/1000 at 146.141.76.153 Action
> > bridge(user/${dialed_extension}@${domain_name})
> > Dialplan: sofia/internal/1000 at 146.141.76.153 Action answer()
> > Dialplan: sofia/internal/1000 at 146.141.76.153 Action sleep(1000)
> > Dialplan: sofia/internal/1000 at 146.141.76.153 Action voicemail(default
> > ${domain_name} ${dialed_extension})
> > 2010-10-25 10:47:47.193206 [DEBUG] switch_core_state_machine.c:119
> > (sofia/internal/1000 at 146.141.76.153) State Change CS_ROUTING ->
> CS_EXECUTE
> > 2010-10-25 10:47:47.193206 [DEBUG] switch_core_session.c:1057 Send signal
> > sofia/internal/1000 at 146.141.76.153 [BREAK]
> > 2010-10-25 10:47:47.193206 [DEBUG] switch_core_state_machine.c:345
> > (sofia/internal/1000 at 146.141.76.153) State ROUTING going to sleep
> > 2010-10-25 10:47:47.193206 [DEBUG] switch_core_state_machine.c:318
> > (sofia/internal/1000 at 146.141.76.153) Running State Change CS_EXECUTE
> > 2010-10-25 10:47:47.193206 [DEBUG] switch_core_state_machine.c:352
> > (sofia/internal/1000 at 146.141.76.153) State EXECUTE
> > 2010-10-25 10:47:47.193206 [DEBUG] mod_sofia.c:239
> > sofia/internal/1000 at 146.141.76.153 SOFIA EXECUTE
> > 2010-10-25 10:47:47.193206 [DEBUG] switch_core_state_machine.c:157
> > sofia/internal/1000 at 146.141.76.153 Standard EXECUTE
> > EXECUTE sofia/internal/1000 at 146.141.76.153 set(open=true)
> > 2010-10-25 10:47:47.193206 [DEBUG] mod_dptools.c:1028
> > sofia/internal/1000 at 146.141.76.153 SET [open]=[true]
> > 2010-10-25 10:47:47.193206 [ERR] switch_core_session.c:1807 Invalid
> > Application hash
> >
> >
> > --
> > David Wafula
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
David Wafula
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From tculjaga at gmail.com  Mon Oct 25 04:17:27 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 25 Oct 2010 13:17:27 +0200
Subject: [Freeswitch-users] Need help on regex
In-Reply-To: <1287954596503-5668666.post@n2.nabble.com>
References: <1287954596503-5668666.post@n2.nabble.com>
Message-ID: <AANLkTimFMreq2SOSrvCcBcLbNsSt-DuysR0Dj0TYwLfi@mail.gmail.com>

On Sun, Oct 24, 2010 at 11:09 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> I am not much of a regex person but am looking for some help on crafting a
> simple regex filter that can do the following criterias:
>
> 1. The dialed number is always a 10-digits number, i.e. an area code +
> 7-digits number (without any leading digit(s) of +1 and/or 1).
>
> 2. If the 4-th digit of the dialed number (after the area code) is a
> non-zero number, do nothing. Otherwise, the regex filter will strip the
> leading 3 digits (area code) as well as the last 2 digits number and this
> will leave only 1-st 5-digits number (no area code). For instance, if the
> 10-digits dialed number is 0120123456, then the regex filter will return
> 01234. More examples followed:
>
> 10-digits input number     output number
> ---------------------      -------------
>    4120123456                01234
>    0121234567                0121234567
>    1310156434                01564
>    8231234567                8231234567
>    0126543210                0126543210
> Anyone? Thanks.
>
>

i'd say, this one:


<action application="set" data="myNUM=${cond(${destination_number:4:1} == 0
? ${regex(${destination_number}|\d{3}(0\d{4}).*|%1)} :
${destination_number})}"/>


i wrote it by hearth .. so you might tune it a bit ... but the basic is here
:)
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From tculjaga at gmail.com  Mon Oct 25 04:42:56 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 25 Oct 2010 13:42:56 +0200
Subject: [Freeswitch-users] Need help on regex
In-Reply-To: <AANLkTimFMreq2SOSrvCcBcLbNsSt-DuysR0Dj0TYwLfi@mail.gmail.com>
References: <1287954596503-5668666.post@n2.nabble.com>
	<AANLkTimFMreq2SOSrvCcBcLbNsSt-DuysR0Dj0TYwLfi@mail.gmail.com>
Message-ID: <AANLkTikonie7GnCfmNtB+KJ1_CJkCt63Zf1RVhdXQzUv@mail.gmail.com>

On Mon, Oct 25, 2010 at 1:17 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

>
>
> On Sun, Oct 24, 2010 at 11:09 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:
>
>>
>> I am not much of a regex person but am looking for some help on crafting a
>> simple regex filter that can do the following criterias:
>>
>> 1. The dialed number is always a 10-digits number, i.e. an area code +
>> 7-digits number (without any leading digit(s) of +1 and/or 1).
>>
>> 2. If the 4-th digit of the dialed number (after the area code) is a
>> non-zero number, do nothing. Otherwise, the regex filter will strip the
>> leading 3 digits (area code) as well as the last 2 digits number and this
>> will leave only 1-st 5-digits number (no area code). For instance, if the
>> 10-digits dialed number is 0120123456, then the regex filter will return
>> 01234. More examples followed:
>>
>> 10-digits input number     output number
>> ---------------------      -------------
>>    4120123456                01234
>>    0121234567                0121234567
>>    1310156434                01564
>>    8231234567                8231234567
>>    0126543210                0126543210
>> Anyone? Thanks.
>>
>>
>
> i'd say, this one:
>
>
> <action application="set" data="myNUM=${cond(${destination_number:4:1} == 0
> ? ${regex(${destination_number}|\d{3}(0\d{4}).*|%1)} :
> ${destination_number})}"/>
>
>
> i wrote it by hearth .. so you might tune it a bit ... but the basic is
> here :)
>
>
>
actualy it should work simply by this as well:

<action application="set"
data="myNUM=${regex(${destination_number}|\d{3}(0\d{4}).*|%1)}"/>


if rexexp match fails, the api returns the source string without any
changes.




chers!
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From babstar99 at gmail.com  Sun Oct 24 16:27:35 2010
From: babstar99 at gmail.com (Babstar)
Date: Mon, 25 Oct 2010 10:27:35 +1100
Subject: [Freeswitch-users] Memory problems
Message-ID: <AANLkTina_J3XMLcrTweeKVw6AaVjJaCsvytFGSo+iPfv@mail.gmail.com>

I have been using Freeswitch (git updated regularly) for about 9 months on a
very low load ubuntu server.
I have noticed that the after a period of time (sometimes only a few days) I
am unable to make calls from attached SIP phones.  Logging into the server I
find freeswitch is taking upto 70% of the memory (measured by top), and I
can hear the disk being thrashed (swapping, I assume).  Restarting
Freeswitch fixes all the problems & it then uses only 5% of them memory.

How can I collect more debuging information to help fix the problem?
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From mark.duling at biola.edu  Sun Oct 24 01:23:40 2010
From: mark.duling at biola.edu (Mark Duling)
Date: Sun, 24 Oct 2010 01:23:40 -0700
Subject: [Freeswitch-users] OS X 10.6 - CORE_SOFTTIMER_MODULE.so &
	CORE_PCM_MODULE.so missing
Message-ID: <AANLkTim1KMusg1hUKJqs02tVC1NsGWejBm0gZD-3m1WF@mail.gmail.com>

I am a total freeswitch nubie.  I use macports for everything I downloaded a
snapshot tarball as as of yesterday and whipped up a freeswitch port to get
it installed (using gmake etc), and it installed fine even with jingle
support enabled and started up.  But I get this message when starting -

2010-10-23 23:33:54.168320 [CRIT] switch_loadable_module.c:926 Error Loading
module /CORE_SOFTTIMER_MODULE.so
**dlopen(/CORE_SOFTTIMER_MODULE.so, 6): image not found**
2010-10-23 23:33:54.168397 [CRIT] switch_loadable_module.c:926 Error Loading
module /CORE_PCM_MODULE.so
**dlopen(/CORE_PCM_MODULE.so, 6): image not found**

After looking through the archives just now I saw that the developers of
freeswitch only recommend Apple's development tools.  But I don't get any
compile errors, yet those modules aren't built or installed, nor I can find
any reference to them in the build output at all.  Could anyone point me to
what might be going on?  I assume I need these modules -I could not make a
test call with x-lite.

Mark
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From mark.duling at biola.edu  Sun Oct 24 20:46:02 2010
From: mark.duling at biola.edu (Mark Duling)
Date: Sun, 24 Oct 2010 20:46:02 -0700
Subject: [Freeswitch-users] OS X 10.6 - CORE_SOFTTIMER_MODULE.so &
	CORE_PCM_MODULE.so missing
In-Reply-To: <AANLkTim1KMusg1hUKJqs02tVC1NsGWejBm0gZD-3m1WF@mail.gmail.com>
References: <AANLkTim1KMusg1hUKJqs02tVC1NsGWejBm0gZD-3m1WF@mail.gmail.com>
Message-ID: <AANLkTimGi8hMYOPSKPS7bgUN6GX7nuh0s4BWq+jcdGJf@mail.gmail.com>

The problem turned out to be a path issue.  I used configure switches to set
the paths and freeswitch didn't like that.  I'm still not sure why, but I
went with only specifying PREFIX for now and everything works fine.


On Sun, Oct 24, 2010 at 1:23 AM, Mark Duling <mark.duling at biola.edu> wrote:

> I am a total freeswitch nubie.  I use macports for everything I downloaded
> a snapshot tarball as as of yesterday and whipped up a freeswitch port to
> get it installed (using gmake etc), and it installed fine even with jingle
> support enabled and started up.  But I get this message when starting -
>
> 2010-10-23 23:33:54.168320 [CRIT] switch_loadable_module.c:926 Error
> Loading module /CORE_SOFTTIMER_MODULE.so
> **dlopen(/CORE_SOFTTIMER_MODULE.so, 6): image not found**
> 2010-10-23 23:33:54.168397 [CRIT] switch_loadable_module.c:926 Error
> Loading module /CORE_PCM_MODULE.so
> **dlopen(/CORE_PCM_MODULE.so, 6): image not found**
>
> After looking through the archives just now I saw that the developers of
> freeswitch only recommend Apple's development tools.  But I don't get any
> compile errors, yet those modules aren't built or installed, nor I can find
> any reference to them in the build output at all.  Could anyone point me to
> what might be going on?  I assume I need these modules -I could not make a
> test call with x-lite.
>
> Mark
>
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From mcampbellsmith at gmail.com  Sat Oct 23 15:36:54 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 24 Oct 2010 09:36:54 +1100
Subject: [Freeswitch-users] FS not detecting public IP address change
In-Reply-To: <AANLkTi=MoV27FW0Ukd6K+mqmgnkyNgb9hjFx-vBYV1ak@mail.gmail.com>
References: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>
	<AANLkTinEoTcLmUVcpras8-m8pdkfpE=_r0spz1S50iMF@mail.gmail.com>
	<AANLkTi=MoV27FW0Ukd6K+mqmgnkyNgb9hjFx-vBYV1ak@mail.gmail.com>
Message-ID: <AANLkTimCRfKSWG+5OdQY0=HTt2kOqxwk+0vKvaBNdmAd@mail.gmail.com>

Hi!

The router is a ASUS router but it is running the tomato firmware which runs
miniupnd.

In the debug I see the message: 2010-10-24 08:02:59.178918 [DEBUG]
sofia.c:956 nua_i_outbound: unknown event 8: 101 NAT detected

This is what I see below. I hope I enabled all debug messages.  I thought I
saw some xml type messages earlier, but not sure how I enabled them.

nta_outgoing: RTT is 67.958 ms
outbound(0xb6e07c00): NAT binding changed: [210.xx.xxx.xx]:5080 !=
[58.xxx.xx.xx]:5080
nua(0xb6e07c00): event i_outbound 102 NAT binding changed
nua: nua_application_event: entering
2010-10-24 09:05:08.849525 [DEBUG] sofia.c:956 nua_i_outbound: unknown event
8: 102 NAT binding changed


On Sun, Oct 24, 2010 at 1:08 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> btw: what router are you using?
>
>
> On Sat, Oct 23, 2010 at 9:06 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> In a upnp config, FS is depending on the router to notify it of the ip
>> address change.  It is possible the upnp message(s) aren't formatted as
>> expected.  If you turn on debug logging you'll get the upnp messages on the
>> console.  If you can pastebin the capture of those while dropping and
>> restarting the dsl connection I can maybe see if there is something obvious
>> going on.  The debug should also have messages related to the processing of
>> those upnp messages.
>>
>> On Sat, Oct 23, 2010 at 7:04 AM, Mark Campbell-Smith <
>> mcampbellsmith at gmail.com> wrote:
>>
>>> HI!
>>>
>>> I know this question must have been answered 100's of times....
>>>
>>> My adsl is a bit dodgy at the moment and tends to go up and down; which
>>> means I get assigned a new ip address from my ISP.  FS is nat'd behind a
>>> upnp capable router.
>>>
>>> The problem is that FS does not seem to be detecting the change.   For
>>> both the internal and external profile, I have auto-nat setup, and I have
>>> tried stun and host settings in vars.conf.  When I issue a nat_map status I
>>> see my old IP address, and if I then issue a nat_map reinit, I see the new
>>> public IP address.  How can I get this to be automatic?
>>>
>>> Also the public IP address shown in the nat_map status is not the same as
>>> shown in the sofia profile internal/external printouts.  Why is this?
>>>
>>> I'm sure this is configuration, just not sure what to change.
>>>
>>> Thanks!
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Sun Oct 24 02:04:08 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sun, 24 Oct 2010 20:04:08 +1100
Subject: [Freeswitch-users] A few questions
Message-ID: <AANLkTikoFv4n9Naoq3RBzGJA8cGDwbS1P_aQ3WL9kMfB@mail.gmail.com>

HI!

I just updated to the latest git revision, and in my log file I see a few
errors at startup:

1.
2010-10-24 19:55:46.930256 [WARNING] mod_limit.c:47 Loading mod_limit - a
shim for backwards compatability with the new limit system.  This is
deprecated, remove mod_limit and instead load mod_hash and mod_db!
2010-10-24 19:55:46.934054 [WARNING] mod_limit.c:51 mod_hash not loaded,
trying to load...!
2010-10-24 19:55:46.936177 [CRIT] switch_loadable_module.c:928 Error Loading
module /usr/local/freeswitch/mod/mod_hash.so
**/usr/local/freeswitch/mod/mod_hash.so: cannot open shared object file: No
such file or directory**
2010-10-24 19:55:46.940086 [ERR] mod_limit.c:53 Unable to load mod_hash
(module load file routine returned an error)!

2. I don't use database backend... how do I disable this.
2010-10-24 19:55:46.941498 [WARNING] mod_limit.c:59 mod_db not loaded,
trying to load...!
2010-10-24 19:55:46.945519 [CRIT] switch_loadable_module.c:928 Error Loading
module /usr/local/freeswitch/mod/mod_db.so
**/usr/local/freeswitch/mod/mod_db.so: cannot open shared object file: No
such file or directory**
2010-10-24 19:55:46.949313 [ERR] mod_limit.c:61 Unable to load mod_db
(module load file routine returned an error)!

3. I have a G729 license.  Do I haver to update the codec file when FS is
updated?
2010-10-24 19:55:47.326592 [CRIT] switch_loadable_module.c:928 Error Loading
module /usr/local/freeswitch/mod/mod_com_g729.so
**Trying to load an out of date module, please rebuild the module.**

Thanks!
/Mark
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From renjian at gmail.com  Sat Oct 23 19:19:24 2010
From: renjian at gmail.com (Jian Ren)
Date: Sat, 23 Oct 2010 22:19:24 -0400
Subject: [Freeswitch-users] Help needed about building snd_dummy with
 the dummy.c in skypopen
In-Reply-To: <AANLkTikkXGL7H=1Sfhnzub454ob4MeCe7asoxXJi9myT@mail.gmail.com>
References: <AANLkTikQ_vj3Ue1MYs7x1pVj8kiCgFbeKBTL=97PddX=@mail.gmail.com>
	<AANLkTik5BEvEi=P_+cEL2kwjfrTQYLGh7MNtF14-8ROE@mail.gmail.com>
	<AANLkTikkXGL7H=1Sfhnzub454ob4MeCe7asoxXJi9myT@mail.gmail.com>
Message-ID: <AANLkTiku3YBi004AZrC_-vW_Kc01x4NQbR3rtf_yxeNT@mail.gmail.com>

Hi, I tried real machine. Got different error:
FATAL: Error inserting
snd_dummy(/lib/modules/2.6.24-26-server/kernel/sound/drivers/snd-dummy.ko):
Unknown symbol in module, or unknown parameter (see demsg).
Then if I type demsg, got the attached screen.
Please help.

Thanks!

On Fri, Oct 22, 2010 at 11:33 AM, Jian Ren <renjian at gmail.com> wrote:

> Tried under VMPlayer, got the same problem, will try real machine later.
> Attached is the screen shot.
>
> Thanks!
> Jian
>
>
> On Thu, Oct 21, 2010 at 4:45 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>
>> Maybe is a problem with virtualbox. Also, is known that
>> fs+mod_skypopen+skype clients does not works in virtualbox.
>>
>> Try it on a real (hardware) machine.
>>
>> Or (but is less popular) in a xen like virtual machine.
>>
>> -giovanni
>>
>> On Thu, Oct 21, 2010 at 4:21 PM, Jian Ren <renjian at gmail.com> wrote:
>> > Hi,
>> > I am trying to follow this wiki:
>> > http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>> > to setup FS and Skypopen on Ubuntu 8.04(64bit server inside virualbox).
>> >
>> > There is one step asks me to build snd_dummy:
>> >
>> http://wiki.freeswitch.org/wiki/Skypopen_Skype_Endpoint_and_Trunk#ALSA_and_the_custom_snd-dummy
>> > The section is like below:
>> >
>> > Ubuntu, Debian
>> >
>> > Note, Ubuntu may have alsa sound drivers installed at /lib/modules/<your
>> > kernel version>/ubuntu/sound/..., you may need to remove the dir to
>> allow
>> > modprobe search from the default place: /lib/modules/<your kernel
>> > verision>/kernel/sound/ (don't forget to rerun /sbin/depmod after
>> removing
>> > the old sound directory. Eg: /sbin/depmod -a 2.6.24-24-server).
>> >
>> > ./configure --with-redhat=no \
>> > --with-cards=dummy,usb-audio,hda-intel,hrtimer,rtctimer \
>> > --with-card-options=all
>> > make && make install
>> >
>> > After the first make && make install, copy
>> mod_skypopen/configs/alsa/dummy.c
>> > to alsa-driver-1.0.20sound/drivers/dummy.c .
>> >
>> > make && make install #again :)
>> >
>> > The problem is after I did all of them, when I entered modprobe
>> snd_dummy,
>> > it always crashed and returned as "killed" on my terminal, I could see
>> it
>> > dumped out a huge block or data(numbers, etc) on the host but don't
>> > understand the meaning. While if I used the original dummy.c inside
>> > alsa-driver-1.0.20, it worked fine.
>> > Besides, the default ubuntu installation doesn't include kernal dev and
>> > source, so I did one more step(or it cannot build alsa).
>> >
>> > Did anyone try the same and get it working?
>> >
>> > Thanks!
>> > Jian
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From renjian at gmail.com  Sun Oct 24 14:28:20 2010
From: renjian at gmail.com (Jian Ren)
Date: Sun, 24 Oct 2010 17:28:20 -0400
Subject: [Freeswitch-users] Help needed about building snd_dummy with
 the dummy.c in skypopen
In-Reply-To: <AANLkTiku3YBi004AZrC_-vW_Kc01x4NQbR3rtf_yxeNT@mail.gmail.com>
References: <AANLkTikQ_vj3Ue1MYs7x1pVj8kiCgFbeKBTL=97PddX=@mail.gmail.com>
	<AANLkTik5BEvEi=P_+cEL2kwjfrTQYLGh7MNtF14-8ROE@mail.gmail.com>
	<AANLkTikkXGL7H=1Sfhnzub454ob4MeCe7asoxXJi9myT@mail.gmail.com>
	<AANLkTiku3YBi004AZrC_-vW_Kc01x4NQbR3rtf_yxeNT@mail.gmail.com>
Message-ID: <AANLkTik6aj0dRqL4BqkRgT9vvp+7OgQtWT=abJ0F2EZ5@mail.gmail.com>

Ok, I found out the new error was caused by missing of 32bit compatibility
libraries, after I ran this:

apt-get -y install ia32-libs lib32asound2 libc6-i386 lib32gcc1 \
lib32stdc++6 lib32ncurses5 lib32z1

Now I am getting exactly the same problem as if under virtualBox or
VMPlayer. So it's not a VM problem.
Please help!
Thanks!
Jian


On Sat, Oct 23, 2010 at 10:19 PM, Jian Ren <renjian at gmail.com> wrote:

> Hi, I tried real machine. Got different error:
> FATAL: Error inserting
> snd_dummy(/lib/modules/2.6.24-26-server/kernel/sound/drivers/snd-dummy.ko):
> Unknown symbol in module, or unknown parameter (see demsg).
> Then if I type demsg, got the attached screen.
> Please help.
>
> Thanks!
>
>
> On Fri, Oct 22, 2010 at 11:33 AM, Jian Ren <renjian at gmail.com> wrote:
>
>> Tried under VMPlayer, got the same problem, will try real machine later.
>> Attached is the screen shot.
>>
>> Thanks!
>> Jian
>>
>>
>> On Thu, Oct 21, 2010 at 4:45 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>>
>>> Maybe is a problem with virtualbox. Also, is known that
>>> fs+mod_skypopen+skype clients does not works in virtualbox.
>>>
>>> Try it on a real (hardware) machine.
>>>
>>> Or (but is less popular) in a xen like virtual machine.
>>>
>>> -giovanni
>>>
>>> On Thu, Oct 21, 2010 at 4:21 PM, Jian Ren <renjian at gmail.com> wrote:
>>> > Hi,
>>> > I am trying to follow this wiki:
>>> > http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>> > to setup FS and Skypopen on Ubuntu 8.04(64bit server inside virualbox).
>>> >
>>> > There is one step asks me to build snd_dummy:
>>> >
>>> http://wiki.freeswitch.org/wiki/Skypopen_Skype_Endpoint_and_Trunk#ALSA_and_the_custom_snd-dummy
>>> > The section is like below:
>>> >
>>> > Ubuntu, Debian
>>> >
>>> > Note, Ubuntu may have alsa sound drivers installed at
>>> /lib/modules/<your
>>> > kernel version>/ubuntu/sound/..., you may need to remove the dir to
>>> allow
>>> > modprobe search from the default place: /lib/modules/<your kernel
>>> > verision>/kernel/sound/ (don't forget to rerun /sbin/depmod after
>>> removing
>>> > the old sound directory. Eg: /sbin/depmod -a 2.6.24-24-server).
>>> >
>>> > ./configure --with-redhat=no \
>>> > --with-cards=dummy,usb-audio,hda-intel,hrtimer,rtctimer \
>>> > --with-card-options=all
>>> > make && make install
>>> >
>>> > After the first make && make install, copy
>>> mod_skypopen/configs/alsa/dummy.c
>>> > to alsa-driver-1.0.20sound/drivers/dummy.c .
>>> >
>>> > make && make install #again :)
>>> >
>>> > The problem is after I did all of them, when I entered modprobe
>>> snd_dummy,
>>> > it always crashed and returned as "killed" on my terminal, I could see
>>> it
>>> > dumped out a huge block or data(numbers, etc) on the host but don't
>>> > understand the meaning. While if I used the original dummy.c inside
>>> > alsa-driver-1.0.20, it worked fine.
>>> > Besides, the default ubuntu installation doesn't include kernal dev and
>>> > source, so I did one more step(or it cannot build alsa).
>>> >
>>> > Did anyone try the same and get it working?
>>> >
>>> > Thanks!
>>> > Jian
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
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From johnsonw at eallway.com  Sun Oct 24 20:49:10 2010
From: johnsonw at eallway.com (Johnson)
Date: Sun, 24 Oct 2010 20:49:10 -0700 (PDT)
Subject: [Freeswitch-users] FIFO Originate caller ID
In-Reply-To: <1287977517125-5669276.post@n2.nabble.com>
References: <29507085-AE04-4A43-A41A-6D279810CCEC@shaw.ca>
	<88D85720-C24A-43B8-A6E8-7F9179D71625@jerris.com>
	<D85EDE43-C1D0-438C-A273-0016856CA723@shaw.ca>
	<2CCD724A-65E4-4EB6-A813-50F26CC0642A@jerris.com>
	<1287977517125-5669276.post@n2.nabble.com>
Message-ID: <1287978550996-5669301.post@n2.nabble.com>


I am a new vanilla. Who know how to configure or put in new mod to show
incoming callerid on SIP softphone without supporting UPDATE MESSAGE in
Queue or ACD? Any additional prompting is very appreciated. 

Johnson    

-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FIFO-Originate-caller-ID-tp4402117p5669301.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From david.ponzone at ipeva.fr  Mon Oct 25 06:57:47 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 25 Oct 2010 15:57:47 +0200
Subject: [Freeswitch-users] A few questions
In-Reply-To: <AANLkTikoFv4n9Naoq3RBzGJA8cGDwbS1P_aQ3WL9kMfB@mail.gmail.com>
References: <AANLkTikoFv4n9Naoq3RBzGJA8cGDwbS1P_aQ3WL9kMfB@mail.gmail.com>
Message-ID: <4946830E-8008-41CC-AA91-641A2771C099@ipeva.fr>

Mark,

1 and 2. for mod_limit, you need now to load mod_db or mod_hash.
If you use to use mod_limit with hash tables, you need now mod_hash.
So you need to enable it in modules.conf so it is compiled and  
installed.
Make sure you load it in conf/autoload.conf.xml. You can also disable  
mod_db.
mod_limit is not required anymore.

3. You just need to update the G729 codec binary:
fsg729-158-installer

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 24/10/2010 ? 11:04, Mark Campbell-Smith a ?crit :

> HI!
>
> I just updated to the latest git revision, and in my log file I see  
> a few errors at startup:
>
> 1.
> 2010-10-24 19:55:46.930256 [WARNING] mod_limit.c:47 Loading  
> mod_limit - a shim for backwards compatability with the new limit  
> system.  This is deprecated, remove mod_limit and instead load  
> mod_hash and mod_db!
> 2010-10-24 19:55:46.934054 [WARNING] mod_limit.c:51 mod_hash not  
> loaded, trying to load...!
> 2010-10-24 19:55:46.936177 [CRIT] switch_loadable_module.c:928 Error  
> Loading module /usr/local/freeswitch/mod/mod_hash.so
> **/usr/local/freeswitch/mod/mod_hash.so: cannot open shared object  
> file: No such file or directory**
> 2010-10-24 19:55:46.940086 [ERR] mod_limit.c:53 Unable to load  
> mod_hash (module load file routine returned an error)!
>
> 2. I don't use database backend... how do I disable this.
> 2010-10-24 19:55:46.941498 [WARNING] mod_limit.c:59 mod_db not  
> loaded, trying to load...!
> 2010-10-24 19:55:46.945519 [CRIT] switch_loadable_module.c:928 Error  
> Loading module /usr/local/freeswitch/mod/mod_db.so
> **/usr/local/freeswitch/mod/mod_db.so: cannot open shared object  
> file: No such file or directory**
> 2010-10-24 19:55:46.949313 [ERR] mod_limit.c:61 Unable to load  
> mod_db (module load file routine returned an error)!
>
> 3. I have a G729 license.  Do I haver to update the codec file when  
> FS is updated?
> 2010-10-24 19:55:47.326592 [CRIT] switch_loadable_module.c:928 Error  
> Loading module /usr/local/freeswitch/mod/mod_com_g729.so
> **Trying to load an out of date module, please rebuild the module.**
>
> Thanks!
> /Mark
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jeff at jefflenk.com  Mon Oct 25 07:08:12 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 25 Oct 2010 07:08:12 -0700 (PDT)
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION on
 loopback/app=voicemail, default dialplan
In-Reply-To: <AANLkTim66YgK4oZZc5gw+d7RM9XDhW=FsnJv_bi2rSq+@mail.gmail.com>
References: <AANLkTim66YgK4oZZc5gw+d7RM9XDhW=FsnJv_bi2rSq+@mail.gmail.com>
Message-ID: <1288015692904-5670788.post@n2.nabble.com>


see http://jira.freeswitch.org/browse/FS-2795

As far as using lookback my understanding is that is allows attended
transfers to work correctly but I dont have a good understanding of the
problem or how to explain it.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/INCOMPATIBLE-DESTINATION-on-loopback-app-voicemail-default-dialplan-tp5669257p5670788.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rupa at rupa.com  Mon Oct 25 07:15:45 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 25 Oct 2010 09:15:45 -0500
Subject: [Freeswitch-users] FS not detecting public IP address change
In-Reply-To: <AANLkTimCRfKSWG+5OdQY0=HTt2kOqxwk+0vKvaBNdmAd@mail.gmail.com>
References: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>
	<AANLkTinEoTcLmUVcpras8-m8pdkfpE=_r0spz1S50iMF@mail.gmail.com>
	<AANLkTi=MoV27FW0Ukd6K+mqmgnkyNgb9hjFx-vBYV1ak@mail.gmail.com>
	<AANLkTimCRfKSWG+5OdQY0=HTt2kOqxwk+0vKvaBNdmAd@mail.gmail.com>
Message-ID: <AANLkTikfpRi9Shxiqd2w5nz3ZHsBnrHTt+=rHHnvoDJn@mail.gmail.com>

I need the debug logging from the upnp stuff.  It should look something
like:

+OK log level 7 [7]
freeswitch at internal> 2010-10-25 09:13:59.926861 [DEBUG] switch_nat.c:299 got
UPnP keep alive packet:
NOTIFY * HTTP/1.1
HOST:239.255.255.250:1900
Cache-Control:max-age=60
Location:http://192.168.1.1:5000/rootDesc.xml
Server: Tomato UPnP/1.0 MiniUPnPd/1.4
NT:urn:schemas-upnp-org:service:WANIPConnection:1
USN:uuid:00000000-0000-0000-0000-000000000000::urn:schemas-upnp-org:service:WANIPConnection:1
NTS:ssdp:alive

I also run tomato and haven't had trouble with the upnp support so at least
we have that part working.

You should see the above keepalive appear periodically every 30s or so.  You
should see another set of messages when you terminate the DSL connection and
when that dsl connection comes back online.

On Sat, Oct 23, 2010 at 5:36 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Hi!
>
> The router is a ASUS router but it is running the tomato firmware which
> runs miniupnd.
>
> In the debug I see the message: 2010-10-24 08:02:59.178918 [DEBUG]
> sofia.c:956 nua_i_outbound: unknown event 8: 101 NAT detected
>
> This is what I see below. I hope I enabled all debug messages.  I thought I
> saw some xml type messages earlier, but not sure how I enabled them.
>
> nta_outgoing: RTT is 67.958 ms
> outbound(0xb6e07c00): NAT binding changed: [210.xx.xxx.xx]:5080 !=
> [58.xxx.xx.xx]:5080
> nua(0xb6e07c00): event i_outbound 102 NAT binding changed
> nua: nua_application_event: entering
> 2010-10-24 09:05:08.849525 [DEBUG] sofia.c:956 nua_i_outbound: unknown
> event 8: 102 NAT binding changed
>
>
> On Sun, Oct 24, 2010 at 1:08 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> btw: what router are you using?
>>
>>
>> On Sat, Oct 23, 2010 at 9:06 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>>> In a upnp config, FS is depending on the router to notify it of the ip
>>> address change.  It is possible the upnp message(s) aren't formatted as
>>> expected.  If you turn on debug logging you'll get the upnp messages on the
>>> console.  If you can pastebin the capture of those while dropping and
>>> restarting the dsl connection I can maybe see if there is something obvious
>>> going on.  The debug should also have messages related to the processing of
>>> those upnp messages.
>>>
>>> On Sat, Oct 23, 2010 at 7:04 AM, Mark Campbell-Smith <
>>> mcampbellsmith at gmail.com> wrote:
>>>
>>>> HI!
>>>>
>>>> I know this question must have been answered 100's of times....
>>>>
>>>> My adsl is a bit dodgy at the moment and tends to go up and down; which
>>>> means I get assigned a new ip address from my ISP.  FS is nat'd behind a
>>>> upnp capable router.
>>>>
>>>> The problem is that FS does not seem to be detecting the change.   For
>>>> both the internal and external profile, I have auto-nat setup, and I have
>>>> tried stun and host settings in vars.conf.  When I issue a nat_map status I
>>>> see my old IP address, and if I then issue a nat_map reinit, I see the new
>>>> public IP address.  How can I get this to be automatic?
>>>>
>>>> Also the public IP address shown in the nat_map status is not the same
>>>> as shown in the sofia profile internal/external printouts.  Why is this?
>>>>
>>>> I'm sure this is configuration, just not sure what to change.
>>>>
>>>> Thanks!
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From Nabble at slickdeals.endjunk.com  Mon Oct 25 07:18:05 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 25 Oct 2010 07:18:05 -0700 (PDT)
Subject: [Freeswitch-users] Need help on regex
In-Reply-To: <AANLkTimdxQNGuLxYZSH8YukYoT1b2J0nJFGVOUv9_MCQ@mail.gmail.com>
References: <1287954596503-5668666.post@n2.nabble.com>
	<37A7ED80-F6C5-40CD-93F0-C728282E1628@gmail.com>
	<1287970747309-5669081.post@n2.nabble.com>
	<AANLkTimWM-Vfrpuf4ZwfbkwM2vh_2GN9yv4859s2TCb0@mail.gmail.com>
	<AANLkTimdxQNGuLxYZSH8YukYoT1b2J0nJFGVOUv9_MCQ@mail.gmail.com>
Message-ID: <1288016285627-5670827.post@n2.nabble.com>



Steven Ayre wrote:
> 
> Actually, I've just realised there is a way:
> 
> ^(?|\d{3}(0\d{4})\d{2}$|^(\d{3}[1-9]\d{6}))$
> 
> The (?| ) bracket resets the numbering for each alternative, so the
> first set of brackets in each alternative both point to $1.
> 
> -Steve
Steve: Thank you and that does the trick. I try to modify the above regex
filters so that it will become more user friendly using this
expression="^(?|\d{3}(0\d{4})\d+|^(\d{3}[1-9]\d{6}))$". This way, the 1st
regex filter (in bold) will process when the 4th digit is zero and only will
capture starting from the 4th to 8th digits (a total of 5 digits) and will
ignore the rest digits. In other words, if a user dials 12301234 or
1230123456 or 123012345 or 1230123456789, the regex filter will only return
01234. This works to a point, except if the entered number has an exact area
code + 0 + 4 digits. For instance,

10-digits input number     output number   Comments
---------------------      -------------   ----------
    4120123456              01234          Does work
    41201234                               Doesn't work
    412012346               01234          Does work
    0121234567              0121234567     Does work
    1310156434              01564          Does work
    131015643               01564          Does work
    13101564                               Doesn't work
    8231234567              8231234567     Does work
    823123456               823123456      Does work
    0126543210              0126543210     Does work

I can easily fix the above problem by introducing an additional regex filter
into the above expression, i.e.
expression="^(?|\d{3}(0\d{4})|\d{3}(0\d{4})\d+|^(\d{3}[1-9]\d{6}))$".
However, if there is a better way to achieve this with less regex filter, it
sure will be a plus.

Mose: Thank you too even though the ":" doesn't seem to work. However, this
bring an interesting thing for me to read the wiki 
http://wiki.freeswitch.org/wiki/Regular_Expression ReGex  so that I will
understand further.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Need-help-on-regex-tp5668666p5670827.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rupa at rupa.com  Mon Oct 25 07:29:26 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 25 Oct 2010 09:29:26 -0500
Subject: [Freeswitch-users] Need help on regex
In-Reply-To: <1288016285627-5670827.post@n2.nabble.com>
References: <1287954596503-5668666.post@n2.nabble.com>
	<37A7ED80-F6C5-40CD-93F0-C728282E1628@gmail.com>
	<1287970747309-5669081.post@n2.nabble.com>
	<AANLkTimWM-Vfrpuf4ZwfbkwM2vh_2GN9yv4859s2TCb0@mail.gmail.com>
	<AANLkTimdxQNGuLxYZSH8YukYoT1b2J0nJFGVOUv9_MCQ@mail.gmail.com>
	<1288016285627-5670827.post@n2.nabble.com>
Message-ID: <AANLkTikBFEw1-uCty8LReKGZOYZmHFdjJe=+5hqO8Fqd@mail.gmail.com>

I've gotta ask.  What is the reason / business process you are doing that
requires a complex regex/match like this?

On Mon, Oct 25, 2010 at 9:18 AM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> Steve: Thank you and that does the trick. I try to modify the above regex
> filters so that it will become more user friendly using this
> expression="^(?|\d{3}(0\d{4})\d+|^(\d{3}[1-9]\d{6}))$". This way, the 1st
> regex filter (in bold) will process when the 4th digit is zero and only
> will
> capture starting from the 4th to 8th digits (a total of 5 digits) and will
> ignore the rest digits. In other words, if a user dials 12301234 or
> 1230123456 or 123012345 or 1230123456789, the regex filter will only return
> 01234. This works to a point, except if the entered number has an exact
> area
> code + 0 + 4 digits. For instance,
>
> 10-digits input number     output number   Comments
> ---------------------      -------------   ----------
>    4120123456              01234          Does work
>    41201234                               Doesn't work
>    412012346               01234          Does work
>    0121234567              0121234567     Does work
>    1310156434              01564          Does work
>    131015643               01564          Does work
>    13101564                               Doesn't work
>    8231234567              8231234567     Does work
>    823123456               823123456      Does work
>    0126543210              0126543210     Does work
>
> I can easily fix the above problem by introducing an additional regex
> filter
> into the above expression, i.e.
> expression="^(?|\d{3}(0\d{4})|\d{3}(0\d{4})\d+|^(\d{3}[1-9]\d{6}))$".
> However, if there is a better way to achieve this with less regex filter,
> it
> sure will be a plus.
>
> Mose: Thank you too even though the ":" doesn't seem to work. However, this
> bring an interesting thing for me to read the wiki
> http://wiki.freeswitch.org/wiki/Regular_Expression ReGex  so that I will
> understand further.
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
>


-- 
-Rupa
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From Nabble at slickdeals.endjunk.com  Mon Oct 25 08:10:39 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 25 Oct 2010 08:10:39 -0700 (PDT)
Subject: [Freeswitch-users] Need help on regex
In-Reply-To: <AANLkTikBFEw1-uCty8LReKGZOYZmHFdjJe=+5hqO8Fqd@mail.gmail.com>
References: <1287954596503-5668666.post@n2.nabble.com>
	<37A7ED80-F6C5-40CD-93F0-C728282E1628@gmail.com>
	<1287970747309-5669081.post@n2.nabble.com>
	<AANLkTimWM-Vfrpuf4ZwfbkwM2vh_2GN9yv4859s2TCb0@mail.gmail.com>
	<AANLkTimdxQNGuLxYZSH8YukYoT1b2J0nJFGVOUv9_MCQ@mail.gmail.com>
	<1288016285627-5670827.post@n2.nabble.com>
	<AANLkTikBFEw1-uCty8LReKGZOYZmHFdjJe=+5hqO8Fqd@mail.gmail.com>
Message-ID: <1288019439632-5671023.post@n2.nabble.com>



Rupa Schomaker wrote:
> 
> I've gotta ask.  What is the reason / business process you are doing that
> requires a complex regex/match like this?
Some VoSPs have their internal (non-NANPA standard) numbers with no area
codes and of length less than 7 digits while my users are used to with a
10-digits dialing scheme. So, I thought perhaps if I device a scheme to let
the users dial any 10-digits number and when the 1st digit is 0 (after any
area code), then the dialplan will strip off the area code + the rest of
unused digits and will only leave a 5-digits numbers starting with 0.

I hope this makes it clear to you even though the logic behind this may not
make sense. Thanks for asking.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Need-help-on-regex-tp5668666p5671023.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Russell.Mosemann at cune.org  Mon Oct 25 08:13:14 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Mon, 25 Oct 2010 15:13:14 -0000
Subject: [Freeswitch-users] Need help on regex
In-Reply-To: <1288016285627-5670827.post@n2.nabble.com>
Message-ID: <20101025151314.3DC0C3E577F@cuneorg-email.cune.pri>

mazilo <Nabble at slickdeals.endjunk.com> said:

> Mose: Thank you too even though the ":" doesn't seem to work.

(?: is only for grouping and does not capture any values.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From pmhshz at gmail.com  Mon Oct 25 09:13:52 2010
From: pmhshz at gmail.com (MohammedShehzad)
Date: Mon, 25 Oct 2010 21:43:52 +0530
Subject: [Freeswitch-users] Delay in Voicemai's MWI NOTIFY
Message-ID: <AANLkTimbndYg-ed3U8f6f5tNGFu_3aQu+cNohCb576Cs@mail.gmail.com>

Hi all,

Is there any way I can configure FreeSWITCH, to send NOTIFY for
Voicemail's MWI to registered phones whenever new vm comes or deleted?
The issue is that FreeSWITCH sends NOTIFY to update mailbox status
only when phone re-register.
One way I can decrease the register expiry time on phones. But It
seems not proper way. I think NOTIFY should be sent independently when
message get received or deleted.


Regards,
MohammedShehzad


From mario_fs at mgtech.com  Mon Oct 25 09:58:18 2010
From: mario_fs at mgtech.com (Mario Guzman)
Date: Mon, 25 Oct 2010 09:58:18 -0700
Subject: [Freeswitch-users] OS X 10.6 - CORE_SOFTTIMER_MODULE.so &
	CORE_PCM_MODULE.so missing
In-Reply-To: <AANLkTim1KMusg1hUKJqs02tVC1NsGWejBm0gZD-3m1WF@mail.gmail.com>
References: <AANLkTim1KMusg1hUKJqs02tVC1NsGWejBm0gZD-3m1WF@mail.gmail.com>
Message-ID: <596D9321-6514-4F00-8AB6-658548C59467@mgtech.com>

This is what I did Saturday and works. I have been refining this for a month. Will put on the wiki when cleaned I finish and clean it up.

Install FS on osX 10.6.4 as of October 23 2010 using git version

INSTALL APPLE SDK
Per Apple instructions

INSTALL GIT & CREATE SOURCE DIRECTORY
NOTE: This must be done before changing /usr/local owner or the installer won't create the files.
1. Download from: http://code.google.com/p/git-osx-installer/
2. It will launch an easy gui installer.
3. This install creates /usr/local if it did not exist.

CHANGE /usr/local OWNER & GROUP
1. Launch the Terminal application.
2. cd /usr
3. sudo chown -R yourid:yourgroup local   <- required for editing 

CREATE /usr/local/src
This is where we will place the source to all packages
1. cd local  or  cd /usr/local  <- if not already there
2. mkdir src

INSTALL LIBJPEG NEEDED FOR SPANDSP:
IMPORTANT: Most searches will find libjpeg-6, DO NOT USE IT! It took
several hours of trying to load it on osx and never got it to work. V7
was a snap. See:
http://proteus-tech.com/blog/cwt/install-pil-in-snow-leopard/
1. Download libjpeg at www.ijg.org/files/jpegsrc.v7.tar.gz
2. Will download to Downloads directory.
3. Click to uncompress or tar zxvf jpegsrc.v7.tar.gz
4. cd Downloads
5. mv jpeg-7  /usr/local/src
6. cd /usr/local/src/jpeg-7
7. ./configure --enable-shared --enable-static
8. make
9. sudo makemake install  <- need sudo because loaded into /usr/bin

DOWNLOAD FREESWITCH
1. cd src   or  cd /usr/local/src
2. Download git version of FreeSwitch:
git clone git://git.freeswitch.org/freeswitch.git

GENERATE FREESWITCH BASIC MODULES
This must be done to create modules.conf
1. cd freeswitch  or  cd /usr/local/src/freeswitch
2. ./bootstrap.sh   <- Creates modules.conf
3. ./configure

EDIT MODULES.CONF TO INCLUDE FUNCTIONS
1. Launch xCode from the SDK.
2. Use File/Open.. menu
3. While file window is open press command+shift+.(period) to view hidden files.
4. Navigate to and edit /usr/local/src/freeswitch/modules.conf
5. Search for flite and remove #.
6. If you want mp3 emails search for shout and remove #. <- Not recommended!
7. Save the file.

COMPILE FS TO CREATE ADDITIONAL DIRECTORIES AND TEST
This will create libs/flite and other files due to modules.conf changes.
1. cd freeswitch  or  cd /usr/local/src/freeswitch if not already there.
2. make
if you get "error: conflicting types for ?swab?" then you must FIX FLITE
If no errors proceed to INSTALL FREESWITCH

FIX FLITE COMPILE PROBLEM (This may be fixed by the developers).
This is what is causing flite to fail during make. For info visit these URLs:
http://lists.berlios.de/pipermail/festlang-talk/2008-June/001043.html
http://www.clas.ufl.edu/users/mpanning/OSX.html
1. cp /usr/include/string.h /usr/local/src/freeswitch/libs/flite-1.3.99/include
2. chmod +rw+rw+rw /usr/local/src/freeswitch/libs/flite-1.3.99/include/string.h  <- So you can update the file.
3. Launch xCode from the SDK.
4. Use File/Open.. menu
5. While file window is open press command+shift+.(period) to view hidden files.
6. Navigate /usr/local/src/freeswitch/libs/flite-1.3.99/include and select file string.h
7. find "swab" twice and change line:
void swab(const void * __restrict, void * __restrict, ssize_t);
              to (comment the line):
/*void swab(const void * __restrict, void * __restrict, ssize_t);*/
8. Save the file
9. make  <- in the freeswitch directory to recompile.

INSTALL FREESWITCH
1. cd /usr/local/src/freeswitch
2. make install   <- note sudo not needed here
3. make cd-sounds-install
4. make cd-moh-install

On Oct 24, 2010, at 1:23 AM, Mark Duling wrote:

> I am a total freeswitch nubie.  I use macports for everything I downloaded a snapshot tarball as as of yesterday and whipped up a freeswitch port to get it installed (using gmake etc), and it installed fine even with jingle support enabled and started up.  But I get this message when starting - 
> 
> 2010-10-23 23:33:54.168320 [CRIT] switch_loadable_module.c:926 Error Loading module /CORE_SOFTTIMER_MODULE.so
> **dlopen(/CORE_SOFTTIMER_MODULE.so, 6): image not found**
> 2010-10-23 23:33:54.168397 [CRIT] switch_loadable_module.c:926 Error Loading module /CORE_PCM_MODULE.so
> **dlopen(/CORE_PCM_MODULE.so, 6): image not found**
> 
> After looking through the archives just now I saw that the developers of freeswitch only recommend Apple's development tools.  But I don't get any compile errors, yet those modules aren't built or installed, nor I can find any reference to them in the build output at all.  Could anyone point me to what might be going on?  I assume I need these modules -I could not make a test call with x-lite.
> 
> Mark
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Mon Oct 25 10:05:29 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 25 Oct 2010 12:05:29 -0500
Subject: [Freeswitch-users] Delay in Voicemai's MWI NOTIFY
In-Reply-To: <AANLkTimbndYg-ed3U8f6f5tNGFu_3aQu+cNohCb576Cs@mail.gmail.com>
References: <AANLkTimbndYg-ed3U8f6f5tNGFu_3aQu+cNohCb576Cs@mail.gmail.com>
Message-ID: <AANLkTim0wL3hjB8tMtgXmN8o0emN_Jqck4ipq0pYKGPR@mail.gmail.com>

I hope either this sinks in or someone documents it because I have
explained this numerous times.

Either you need to supply the profile name to the voicemail app call

 <action application="voicemail" data="check default ${domain_name}@internal"/>

or

you need to make sure your domain used in the call to voicemail
(${domain_name} in the above example) is aliased to the sip profile
you are using.

you have obviously strayed from the default config where this works as
expected by changing the domain name, the profile name, or the aliases
to no longer align.




On Mon, Oct 25, 2010 at 11:13 AM, MohammedShehzad <pmhshz at gmail.com> wrote:
> Hi all,
>
> Is there any way I can configure FreeSWITCH, to send NOTIFY for
> Voicemail's MWI to registered phones whenever new vm comes or deleted?
> The issue is that FreeSWITCH sends NOTIFY to update mailbox status
> only when phone re-register.
> One way I can decrease the register expiry time on phones. But It
> seems not proper way. I think NOTIFY should be sent independently when
> message get received or deleted.
>
>
> Regards,
> MohammedShehzad
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From gmaruzz at gmail.com  Mon Oct 25 10:09:38 2010
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 25 Oct 2010 19:09:38 +0200
Subject: [Freeswitch-users] Help needed about building snd_dummy with
 the dummy.c in skypopen
In-Reply-To: <AANLkTi=L1ro64h=CBqQ_EUgw8exukt7yno31Fc_O2FTW@mail.gmail.com>
References: <AANLkTikQ_vj3Ue1MYs7x1pVj8kiCgFbeKBTL=97PddX=@mail.gmail.com>
	<AANLkTik5BEvEi=P_+cEL2kwjfrTQYLGh7MNtF14-8ROE@mail.gmail.com>
	<AANLkTikkXGL7H=1Sfhnzub454ob4MeCe7asoxXJi9myT@mail.gmail.com>
	<AANLkTiku3YBi004AZrC_-vW_Kc01x4NQbR3rtf_yxeNT@mail.gmail.com>
	<AANLkTik6aj0dRqL4BqkRgT9vvp+7OgQtWT=abJ0F2EZ5@mail.gmail.com>
	<AANLkTi=L1ro64h=CBqQ_EUgw8exukt7yno31Fc_O2FTW@mail.gmail.com>
Message-ID: <AANLkTikGx+rqDayutiL_Nqhtdo-EwX4k-LKsVza=W=RW@mail.gmail.com>

On Sun, Oct 24, 2010 at 11:32 PM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> After you have installed ALSA, you need to reboot the machine, or to
> rmmod all the snd* modules from the kernel, before to modprobe
> snd-dummy
>

Hello Jian,

I can assure you that the process works, many many people do it and works.

So, we need to find the error.

1) are you sure you have moved away the correct sound modules
directory? check with "uname -a" to know which kernel are you using,
and then move away the sound directory under the correct kernel
directory (you may have more than one kernel and modules installed)

2) you really have to move that directory away, not rename it in place

3) are you sure the ALSA drivers were correctly maked and installed?
Also, don't do the depmod on the command line if you are unsure you
are using the correct kernel in command line (make install will do the
depmod for you, the second one is just a little redundant)

4) if still fails to modprobe:

a) find all the sound directories under the kernel you are actually
using and move them away or delete them
b) delete the /usr/src/alsa-drivers directory
c) do the tar xjf again on the alsa-driver archive you downloaded
d) do configure - make - make install - overwrite the dummy.c with my
dummy.c - make - make install
e) reboot
f) modprobe snd-dummy

Let me know how it goes.

-giovanni














>
>
>>
>> Now I am getting exactly the same problem as if under virtualBox or
>> VMPlayer. So it's not a VM problem.
>> Please help!
>> Thanks!
>> Jian
>>
>> On Sat, Oct 23, 2010 at 10:19 PM, Jian Ren <renjian at gmail.com> wrote:
>>>
>>> Hi, I tried real machine. Got different error:
>>> FATAL: Error inserting
>>> snd_dummy(/lib/modules/2.6.24-26-server/kernel/sound/drivers/snd-dummy.ko):
>>> Unknown symbol in module, or unknown parameter (see demsg).
>>> Then if I type demsg, got the attached screen.
>>> Please help.
>>> Thanks!
>>>
>>> On Fri, Oct 22, 2010 at 11:33 AM, Jian Ren <renjian at gmail.com> wrote:
>>>>
>>>> Tried under VMPlayer, got the same problem, will try real machine later.
>>>> Attached is the screen shot.
>>>>
>>>> Thanks!
>>>> Jian
>>>>
>>>> On Thu, Oct 21, 2010 at 4:45 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>>>> wrote:
>>>>>
>>>>> Maybe is a problem with virtualbox. Also, is known that
>>>>> fs+mod_skypopen+skype clients does not works in virtualbox.
>>>>>
>>>>> Try it on a real (hardware) machine.
>>>>>
>>>>> Or (but is less popular) in a xen like virtual machine.
>>>>>
>>>>> -giovanni
>>>>>
>>>>> On Thu, Oct 21, 2010 at 4:21 PM, Jian Ren <renjian at gmail.com> wrote:
>>>>> > Hi,
>>>>> > I am trying to follow this wiki:
>>>>> > http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
>>>>> > to setup FS and Skypopen on Ubuntu 8.04(64bit server inside
>>>>> > virualbox).
>>>>> >
>>>>> > There is one step asks me to build snd_dummy:
>>>>> >
>>>>> > http://wiki.freeswitch.org/wiki/Skypopen_Skype_Endpoint_and_Trunk#ALSA_and_the_custom_snd-dummy
>>>>> > The section is like below:
>>>>> >
>>>>> > Ubuntu, Debian
>>>>> >
>>>>> > Note, Ubuntu may have alsa sound drivers installed at
>>>>> > /lib/modules/<your
>>>>> > kernel version>/ubuntu/sound/..., you may need to remove the dir to
>>>>> > allow
>>>>> > modprobe search from the default place: /lib/modules/<your kernel
>>>>> > verision>/kernel/sound/ (don't forget to rerun /sbin/depmod after
>>>>> > removing
>>>>> > the old sound directory. Eg: /sbin/depmod -a 2.6.24-24-server).
>>>>> >
>>>>> > ./configure --with-redhat=no \
>>>>> > --with-cards=dummy,usb-audio,hda-intel,hrtimer,rtctimer \
>>>>> > --with-card-options=all
>>>>> > make && make install
>>>>> >
>>>>> > After the first make && make install, copy
>>>>> > mod_skypopen/configs/alsa/dummy.c
>>>>> > to alsa-driver-1.0.20sound/drivers/dummy.c .
>>>>> >
>>>>> > make && make install #again?:)
>>>>> >
>>>>> > The problem is after I did all of them, when I entered modprobe
>>>>> > snd_dummy,
>>>>> > it always crashed and returned as "killed" on my terminal, I could see
>>>>> > it
>>>>> > dumped out a huge block or data(numbers, etc) on the host but don't
>>>>> > understand the meaning. While if I used the original dummy.c inside
>>>>> > alsa-driver-1.0.20, it worked fine.
>>>>> > Besides, the default ubuntu installation doesn't include kernal dev
>>>>> > and
>>>>> > source, so I did one more step(or it cannot build alsa).
>>>>> >
>>>>> > Did anyone try the same and get it working?
>>>>> >
>>>>> > Thanks!
>>>>> > Jian
>>>>> >
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >
>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> Cell : +39-347-2665618
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>
>>
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From pjintheusa at gmail.com  Mon Oct 25 11:00:25 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 25 Oct 2010 14:00:25 -0400
Subject: [Freeswitch-users] Using the bridge | operator for Implementing
	Failover
In-Reply-To: <AANLkTi=JfTPXxAwyPPJ1CuKdmJ1pcys=18EE3apW9D1b@mail.gmail.com>
References: <AANLkTimZonOAnpKJKyy24ZnyJD9HnDPYMju40MYL6CvH@mail.gmail.com>
	<AANLkTimoE7t0LQwK3Nmv8Km-KSHHAtk+n-Teeiv7zhoQ@mail.gmail.com>
	<AANLkTimfXA70zUQKK0cBoKyvjE2OhfaaxckTPMgpYwHu@mail.gmail.com>
	<AANLkTimT0QTVcJKkczFnuwQJMBmgdfjuG7vr6UxxAx0k@mail.gmail.com>
	<AANLkTikomdMk_s3GovN_gkxm6u2U-_p3Y67Jw6bHLLrt@mail.gmail.com>
	<1287872231993-5666526.post@n2.nabble.com>
	<AANLkTi=JfTPXxAwyPPJ1CuKdmJ1pcys=18EE3apW9D1b@mail.gmail.com>
Message-ID: <AANLkTi=p4LQr-hGAisr4_JPd9i4NaGn8ee5sAXOicZNi@mail.gmail.com>

Is anyone using the | to fail over carriers in the why an trying to here?

There must be something I am missing here but I can not see it.

Any help appreciated.

On Sun, Oct 24, 2010 at 11:09 AM, Phillip Jones <pjintheusa at gmail.com>wrote:

> Jeff - Thanks for the response.
>
> <condition field="destination_number" expression="^(3478309992)$">
>       <action application="answer"/>
>       <action application="sleep" data="2000"/>
>        <!--action application="set"
> data="fail_on_single_reject=NO_ANSWER,NORMAL_CLEARING,ALLOTTED_TIMEOUT"/-->
>
>        <action application="set" data="fail_on_single_reject=true"/>
>        <action application="bridge"
> data="{ignore_early_media=true,originate_continue_on_timeout=false}[leg_timeout=10]sofia/gateway/broadvox1/2158824374|[leg_timeout=25]sofia/gateway/broadvox2/2158824374"/>
>  </condition>
> This gives exactly the same result. The 2158824374 is called by each
> gateway specified.
>
> Pj
> On Sat, Oct 23, 2010 at 6:17 PM, Jeff Lenk <jeff at jefflenk.com> wrote:
>
>>
>> what happens with  <action application="set"
>> data="fail_on_single_reject=true"/>
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Using-the-bridge-operator-for-Implementing-Failover-tp5661010p5666526.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From anthony.minessale at gmail.com  Mon Oct 25 11:09:29 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 25 Oct 2010 13:09:29 -0500
Subject: [Freeswitch-users] Using the bridge | operator for Implementing
	Failover
In-Reply-To: <AANLkTi=JfTPXxAwyPPJ1CuKdmJ1pcys=18EE3apW9D1b@mail.gmail.com>
References: <AANLkTimZonOAnpKJKyy24ZnyJD9HnDPYMju40MYL6CvH@mail.gmail.com>
	<AANLkTimoE7t0LQwK3Nmv8Km-KSHHAtk+n-Teeiv7zhoQ@mail.gmail.com>
	<AANLkTimfXA70zUQKK0cBoKyvjE2OhfaaxckTPMgpYwHu@mail.gmail.com>
	<AANLkTimT0QTVcJKkczFnuwQJMBmgdfjuG7vr6UxxAx0k@mail.gmail.com>
	<AANLkTikomdMk_s3GovN_gkxm6u2U-_p3Y67Jw6bHLLrt@mail.gmail.com>
	<1287872231993-5666526.post@n2.nabble.com>
	<AANLkTi=JfTPXxAwyPPJ1CuKdmJ1pcys=18EE3apW9D1b@mail.gmail.com>
Message-ID: <AANLkTinXG9koM_rVFW_gr5EK_KDVK7=-fhFpNw0yVDy7@mail.gmail.com>

can you try latest

On Sun, Oct 24, 2010 at 10:09 AM, Phillip Jones <pjintheusa at gmail.com> wrote:
> Jeff - Thanks for the response.
>
> <condition field="destination_number" expression="^(3478309992)$">
> ????? <action application="answer"/>
> ?? ?? <action application="sleep" data="2000"/>
> ??? ?? <!--action application="set"
> data="fail_on_single_reject=NO_ANSWER,NORMAL_CLEARING,ALLOTTED_TIMEOUT"/-->
> ?????? <action application="set" data="fail_on_single_reject=true"/>
> ?????? <action application="bridge"
> data="{ignore_early_media=true,originate_continue_on_timeout=false}[leg_timeout=10]sofia/gateway/broadvox1/2158824374|[leg_timeout=25]sofia/gateway/broadvox2/2158824374"/>
> ?</condition>
> This gives exactly the same result. The 2158824374 is called by each gateway
> specified.
>
> Pj
> On Sat, Oct 23, 2010 at 6:17 PM, Jeff Lenk <jeff at jefflenk.com> wrote:
>>
>> what happens with ?<action application="set"
>> data="fail_on_single_reject=true"/>
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Using-the-bridge-operator-for-Implementing-Failover-tp5661010p5666526.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From msc at freeswitch.org  Mon Oct 25 12:23:58 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 25 Oct 2010 12:23:58 -0700
Subject: [Freeswitch-users] Memory problems
In-Reply-To: <AANLkTina_J3XMLcrTweeKVw6AaVjJaCsvytFGSo+iPfv@mail.gmail.com>
References: <AANLkTina_J3XMLcrTweeKVw6AaVjJaCsvytFGSo+iPfv@mail.gmail.com>
Message-ID: <AANLkTimHXVGdoBurEAw6NtFN8PnJDa649wib8_9qTTzC@mail.gmail.com>

Look at the *.db files. Are any of them getting really large?
-MC

On Sun, Oct 24, 2010 at 4:27 PM, Babstar <babstar99 at gmail.com> wrote:

> I have been using Freeswitch (git updated regularly) for about 9 months on
> a very low load ubuntu server.
> I have noticed that the after a period of time (sometimes only a few days)
> I am unable to make calls from attached SIP phones.  Logging into the server
> I find freeswitch is taking upto 70% of the memory (measured by top), and I
> can hear the disk being thrashed (swapping, I assume).  Restarting
> Freeswitch fixes all the problems & it then uses only 5% of them memory.
>
> How can I collect more debuging information to help fix the problem?
>
>
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From curriegrad2004 at gmail.com  Mon Oct 25 13:05:12 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Mon, 25 Oct 2010 13:05:12 -0700
Subject: [Freeswitch-users] A few questions
In-Reply-To: <AANLkTikoFv4n9Naoq3RBzGJA8cGDwbS1P_aQ3WL9kMfB@mail.gmail.com>
References: <AANLkTikoFv4n9Naoq3RBzGJA8cGDwbS1P_aQ3WL9kMfB@mail.gmail.com>
Message-ID: <AANLkTikkRcoWHXo4rBeybdsGgq5Xc6s-__dXhzUhfXrF@mail.gmail.com>

1. No such file is a dead give away that the compiler didn't build
your desired module. Edit the modules.conf file and re-build the
softswitch.
2. Again, that's a dead giveaway
3. This is something that the FreeSwitch developers will have to
resolve for you. iirc, they released later version that fixed this
issue 2 or 3 weeks ago.

On Sun, Oct 24, 2010 at 2:04 AM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> HI!
> I just updated to the latest git revision, and in my log file I see a few
> errors at startup:
> 1.
> 2010-10-24 19:55:46.930256 [WARNING] mod_limit.c:47 Loading mod_limit - a
> shim for backwards compatability with the new limit system. ?This is
> deprecated, remove mod_limit and instead load mod_hash and mod_db!
> 2010-10-24 19:55:46.934054 [WARNING] mod_limit.c:51 mod_hash not loaded,
> trying to load...!
> 2010-10-24 19:55:46.936177 [CRIT] switch_loadable_module.c:928 Error Loading
> module /usr/local/freeswitch/mod/mod_hash.so
> **/usr/local/freeswitch/mod/mod_hash.so: cannot open shared object file: No
> such file or directory**
> 2010-10-24 19:55:46.940086 [ERR] mod_limit.c:53 Unable to load mod_hash
> (module load file routine returned an error)!
> 2. I don't use database backend... how do I disable this.
> 2010-10-24 19:55:46.941498 [WARNING] mod_limit.c:59 mod_db not loaded,
> trying to load...!
> 2010-10-24 19:55:46.945519 [CRIT] switch_loadable_module.c:928 Error Loading
> module /usr/local/freeswitch/mod/mod_db.so
> **/usr/local/freeswitch/mod/mod_db.so: cannot open shared object file: No
> such file or directory**
> 2010-10-24 19:55:46.949313 [ERR] mod_limit.c:61 Unable to load mod_db
> (module load file routine returned an error)!
> 3. I have a G729 license. ?Do I haver to update the codec file when FS is
> updated?
> 2010-10-24 19:55:47.326592 [CRIT] switch_loadable_module.c:928 Error Loading
> module /usr/local/freeswitch/mod/mod_com_g729.so
> **Trying to load an out of date module, please rebuild the module.**
> Thanks!
> /Mark
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From pjintheusa at gmail.com  Mon Oct 25 13:25:19 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 25 Oct 2010 16:25:19 -0400
Subject: [Freeswitch-users] Using the bridge | operator for Implementing
	Failover
In-Reply-To: <AANLkTinXG9koM_rVFW_gr5EK_KDVK7=-fhFpNw0yVDy7@mail.gmail.com>
References: <AANLkTimZonOAnpKJKyy24ZnyJD9HnDPYMju40MYL6CvH@mail.gmail.com>
	<AANLkTimoE7t0LQwK3Nmv8Km-KSHHAtk+n-Teeiv7zhoQ@mail.gmail.com>
	<AANLkTimfXA70zUQKK0cBoKyvjE2OhfaaxckTPMgpYwHu@mail.gmail.com>
	<AANLkTimT0QTVcJKkczFnuwQJMBmgdfjuG7vr6UxxAx0k@mail.gmail.com>
	<AANLkTikomdMk_s3GovN_gkxm6u2U-_p3Y67Jw6bHLLrt@mail.gmail.com>
	<1287872231993-5666526.post@n2.nabble.com>
	<AANLkTi=JfTPXxAwyPPJ1CuKdmJ1pcys=18EE3apW9D1b@mail.gmail.com>
	<AANLkTinXG9koM_rVFW_gr5EK_KDVK7=-fhFpNw0yVDy7@mail.gmail.com>
Message-ID: <AANLkTincare8CYw1bTVcG+Z7_DwHHo7CqahQv8qpp_8H@mail.gmail.com>

Anthony,

Thanks. I updated to the latest trunk and retested. Unfortunately I got
exactly the same result - 2158824374 is called by each gateway specified....

Pj



On Mon, Oct 25, 2010 at 2:09 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> can you try latest
>
> On Sun, Oct 24, 2010 at 10:09 AM, Phillip Jones <pjintheusa at gmail.com>
> wrote:
> > Jeff - Thanks for the response.
> >
> > <condition field="destination_number" expression="^(3478309992)$">
> >       <action application="answer"/>
> >       <action application="sleep" data="2000"/>
> >        <!--action application="set"
> >
> data="fail_on_single_reject=NO_ANSWER,NORMAL_CLEARING,ALLOTTED_TIMEOUT"/-->
> >        <action application="set" data="fail_on_single_reject=true"/>
> >        <action application="bridge"
> >
> data="{ignore_early_media=true,originate_continue_on_timeout=false}[leg_timeout=10]sofia/gateway/broadvox1/2158824374|[leg_timeout=25]sofia/gateway/broadvox2/2158824374"/>
> >  </condition>
> > This gives exactly the same result. The 2158824374 is called by each
> gateway
> > specified.
> >
> > Pj
> > On Sat, Oct 23, 2010 at 6:17 PM, Jeff Lenk <jeff at jefflenk.com> wrote:
> >>
> >> what happens with  <action application="set"
> >> data="fail_on_single_reject=true"/>
> >>
> >> --
> >> View this message in context:
> >>
> http://freeswitch-users.2379917.n2.nabble.com/Using-the-bridge-operator-for-Implementing-Failover-tp5661010p5666526.html
> >> Sent from the freeswitch-users mailing list archive at Nabble.com.
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Oct 25 13:45:23 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 25 Oct 2010 13:45:23 -0700
Subject: [Freeswitch-users] transfer problem
In-Reply-To: <19713.1288004756@ccs.covici.com>
References: <13464.1287506974@ccs.covici.com>
	<AANLkTikh67cGuMn3v4D3QZ+AuQ0vWrLE_fRhL5+g1fg7@mail.gmail.com>
	<19713.1288004756@ccs.covici.com>
Message-ID: <AANLkTinA4Egvfr1N-EvLun6rh-BR=WxwutHdOryACnYp@mail.gmail.com>

John,

I see two issues. At line #174 of your pastebin there is a message about
"invalid gateway". I suspect that the gateway name should be "flowroute" and
not "fflowroute".

At line #180 you have an invalid number format error. I would fix the
invalid gateway first and retest as this second error may be a result of the
bad gateway.

-MC

On Mon, Oct 25, 2010 at 4:05 AM, <covici at ccs.covici.com> wrote:

> OK, the debug log is here http://pastebin.freeswitch.org/14307.  The
> relevant part of the configs is below.
>    <extension name="conference_ivr">
>      <condition field="destination_number" expression="^7000$">
>        <action application="answer"/>
> <action application="sleep" data="500"/>
> <action application="set" data="api_hangup_hook=system sayconf -a local  -a
> ${caller_id_number} -a hung -a up"/>
>        <action application="ivr" data="conference_ivr"/>
>      </condition>
>    </extension>
> and the ivr just asks for passwords and send the call to an extension in
> context conferences.
>
> Hope this helps.
>
> Thanks.
>
> Michael Collins <msc at freeswitch.org> wrote:
>
> > Please supply the configs and a debug trace. My guess is that there's
> > probably just a simple config element that is wrong or missing.
> >
> > -MC
> >
> > On Tue, Oct 19, 2010 at 9:49 AM, <covici at ccs.covici.com> wrote:
> >
> > > Hi.  I have an ivr which connects to conferences which I have in their
> > > own context rather than the default.  The problem is that if I call
> > > someone else and have a 3-way between the other party and the
> > > conference, its fine till I hang up and then fs tries to transfer to
> the
> > > correct conference name, but uses default for the context instead.  Is
> > > this expected behavior or should I file a bug?
> > >
> > > Thanks.
> > >
> > > --
> > > Your life is like a penny.  You're going to lose it.  The question is:
> > > How do
> > > you spend it?
> > >
> > >         John Covici
> > >         covici at ccs.covici.com
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> >
> > ----------------------------------------------------
> > Alternatives:
> >
> > ----------------------------------------------------
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From covici at ccs.covici.com  Mon Oct 25 14:11:40 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 25 Oct 2010 17:11:40 -0400
Subject: [Freeswitch-users] transfer problem
In-Reply-To: <AANLkTinA4Egvfr1N-EvLun6rh-BR=WxwutHdOryACnYp@mail.gmail.com>
References: <13464.1287506974@ccs.covici.com>
	<AANLkTikh67cGuMn3v4D3QZ+AuQ0vWrLE_fRhL5+g1fg7@mail.gmail.com>
	<19713.1288004756@ccs.covici.com>
	<AANLkTinA4Egvfr1N-EvLun6rh-BR=WxwutHdOryACnYp@mail.gmail.com>
Message-ID: <28001.1288041100@ccs.covici.com>

Thanks for the correction -- however still same result.  I did not think
it would matter in this case because the call did reach the phone and
everything after that is local.  Question is why does it try to transfer
to the wrong context after the local extension hangs up?

Michael Collins <msc at freeswitch.org> wrote:

> John,
> 
> I see two issues. At line #174 of your pastebin there is a message about
> "invalid gateway". I suspect that the gateway name should be "flowroute" and
> not "fflowroute".
> 
> At line #180 you have an invalid number format error. I would fix the
> invalid gateway first and retest as this second error may be a result of the
> bad gateway.
> 
> -MC
> 
> On Mon, Oct 25, 2010 at 4:05 AM, <covici at ccs.covici.com> wrote:
> 
> > OK, the debug log is here http://pastebin.freeswitch.org/14307.  The
> > relevant part of the configs is below.
> >    <extension name="conference_ivr">
> >      <condition field="destination_number" expression="^7000$">
> >        <action application="answer"/>
> > <action application="sleep" data="500"/>
> > <action application="set" data="api_hangup_hook=system sayconf -a local  -a
> > ${caller_id_number} -a hung -a up"/>
> >        <action application="ivr" data="conference_ivr"/>
> >      </condition>
> >    </extension>
> > and the ivr just asks for passwords and send the call to an extension in
> > context conferences.
> >
> > Hope this helps.
> >
> > Thanks.
> >
> > Michael Collins <msc at freeswitch.org> wrote:
> >
> > > Please supply the configs and a debug trace. My guess is that there's
> > > probably just a simple config element that is wrong or missing.
> > >
> > > -MC
> > >
> > > On Tue, Oct 19, 2010 at 9:49 AM, <covici at ccs.covici.com> wrote:
> > >
> > > > Hi.  I have an ivr which connects to conferences which I have in their
> > > > own context rather than the default.  The problem is that if I call
> > > > someone else and have a 3-way between the other party and the
> > > > conference, its fine till I hang up and then fs tries to transfer to
> > the
> > > > correct conference name, but uses default for the context instead.  Is
> > > > this expected behavior or should I file a bug?
> > > >
> > > > Thanks.
> > > >
> > > > --
> > > > Your life is like a penny.  You're going to lose it.  The question is:
> > > > How do
> > > > you spend it?
> > > >
> > > >         John Covici
> > > >         covici at ccs.covici.com
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > http://www.freeswitch.org
> > > >
> > >
> > > ----------------------------------------------------
> > > Alternatives:
> > >
> > > ----------------------------------------------------
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From mario_fs at mgtech.com  Mon Oct 25 15:26:01 2010
From: mario_fs at mgtech.com (Mario)
Date: Mon, 25 Oct 2010 15:26:01 -0700
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
Message-ID: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>

I really need help on this as I have weeks into this problem. I thought I had it nailed but I guess not. After 5.5 hours I get:
2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed Registration, setting retry to 15 seconds.
2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed Registration, setting retry to 30 seconds.
2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed Registration, setting retry to 15 seconds.
2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed Registration, setting retry to 15 seconds.
2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777

and no way to make/get calls until I restart FS. I did this:
1. log 7
2. sofia profile xxxx siptrace on   for each profile/gateway
3. restarted router

All three did not solve the problem. The trace and log produced no additional lines which is why I am wondering if FS has a problem since the trace shows no SIP activity.

3 gateways with 2 ITSPs
2 DSL/WAN lines, 1 static and 1 dynamic
I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the external static ip.
sofia status profile ... has the right ext ip
nat_map status shows the dynamic (wrong) IP
I tried starting with -nonat but that was worse
the only way to fix is restart FS.

I read the wiki on external nat, auto_nat and everything else many times.
Thanks Mario



From msc at freeswitch.org  Mon Oct 25 15:34:55 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 25 Oct 2010 15:34:55 -0700
Subject: [Freeswitch-users] transfer problem
In-Reply-To: <28001.1288041100@ccs.covici.com>
References: <13464.1287506974@ccs.covici.com>
	<AANLkTikh67cGuMn3v4D3QZ+AuQ0vWrLE_fRhL5+g1fg7@mail.gmail.com>
	<19713.1288004756@ccs.covici.com>
	<AANLkTinA4Egvfr1N-EvLun6rh-BR=WxwutHdOryACnYp@mail.gmail.com>
	<28001.1288041100@ccs.covici.com>
Message-ID: <AANLkTinVAP-5A4KA7gz6hbRUVw7qh6Kc+ty48gs4eucZ@mail.gmail.com>

John,

Perhaps I've missed something. Can you confirm exactly what steps you are
taking? I don't want to make any assumptions. I can see that ext 202 calls
out to 7034754612 and then 202 also makes a call to ext 7000. What I don't
know is if that's a transfer or 3-way or what. Also, can you capture the log
again, this time with a sip trace? Use "sofia global siptrace on".

Thanks,
MC

On Mon, Oct 25, 2010 at 2:11 PM, <covici at ccs.covici.com> wrote:

> Thanks for the correction -- however still same result.  I did not think
> it would matter in this case because the call did reach the phone and
> everything after that is local.  Question is why does it try to transfer
> to the wrong context after the local extension hangs up?
>
> Michael Collins <msc at freeswitch.org> wrote:
>
> > John,
> >
> > I see two issues. At line #174 of your pastebin there is a message about
> > "invalid gateway". I suspect that the gateway name should be "flowroute"
> and
> > not "fflowroute".
> >
> > At line #180 you have an invalid number format error. I would fix the
> > invalid gateway first and retest as this second error may be a result of
> the
> > bad gateway.
> >
> > -MC
> >
> > On Mon, Oct 25, 2010 at 4:05 AM, <covici at ccs.covici.com> wrote:
> >
> > > OK, the debug log is here http://pastebin.freeswitch.org/14307.  The
> > > relevant part of the configs is below.
> > >    <extension name="conference_ivr">
> > >      <condition field="destination_number" expression="^7000$">
> > >        <action application="answer"/>
> > > <action application="sleep" data="500"/>
> > > <action application="set" data="api_hangup_hook=system sayconf -a local
>  -a
> > > ${caller_id_number} -a hung -a up"/>
> > >        <action application="ivr" data="conference_ivr"/>
> > >      </condition>
> > >    </extension>
> > > and the ivr just asks for passwords and send the call to an extension
> in
> > > context conferences.
> > >
> > > Hope this helps.
> > >
> > > Thanks.
> > >
> > > Michael Collins <msc at freeswitch.org> wrote:
> > >
> > > > Please supply the configs and a debug trace. My guess is that there's
> > > > probably just a simple config element that is wrong or missing.
> > > >
> > > > -MC
> > > >
> > > > On Tue, Oct 19, 2010 at 9:49 AM, <covici at ccs.covici.com> wrote:
> > > >
> > > > > Hi.  I have an ivr which connects to conferences which I have in
> their
> > > > > own context rather than the default.  The problem is that if I call
> > > > > someone else and have a 3-way between the other party and the
> > > > > conference, its fine till I hang up and then fs tries to transfer
> to
> > > the
> > > > > correct conference name, but uses default for the context instead.
>  Is
> > > > > this expected behavior or should I file a bug?
> > > > >
> > > > > Thanks.
> > > > >
> > > > > --
> > > > > Your life is like a penny.  You're going to lose it.  The question
> is:
> > > > > How do
> > > > > you spend it?
> > > > >
> > > > >         John Covici
> > > > >         covici at ccs.covici.com
> > > > >
> > > > > _______________________________________________
> > > > > FreeSWITCH-users mailing list
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > UNSUBSCRIBE:
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > > http://www.freeswitch.org
> > > > >
> > > >
> > > > ----------------------------------------------------
> > > > Alternatives:
> > > >
> > > > ----------------------------------------------------
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > http://www.freeswitch.org
> > >
> > > --
> > > Your life is like a penny.  You're going to lose it.  The question is:
> > > How do
> > > you spend it?
> > >
> > >         John Covici
> > >         covici at ccs.covici.com
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> >
> > ----------------------------------------------------
> > Alternatives:
> >
> > ----------------------------------------------------
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Oct 25 15:38:39 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 25 Oct 2010 15:38:39 -0700
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
Message-ID: <AANLkTin0Zfgm6xf-9_G+O-Obvko-05Cpzxt3n9Ukhwir@mail.gmail.com>

Mario,

Try doing "sofia global siptrace on" when your system is in this state. It
will definitely turn on sip trace for every profile on your system. It looks
like these are failed outbound registrations, correct? Are you using domain
names or IP addresses for these gateways? I'm just curious. I've seen
situations where FS could not resolve the domain name for some odd reason.

-MC

On Mon, Oct 25, 2010 at 3:26 PM, Mario <mario_fs at mgtech.com> wrote:

> I really need help on this as I have weeks into this problem. I thought I
> had it nailed but I guess not. After 5.5 hours I get:
> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
> Registration, setting retry to 15 seconds.
> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
> Registration, setting retry to 30 seconds.
> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
> Registration, setting retry to 15 seconds.
> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
> Registration, setting retry to 15 seconds.
> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>
> and no way to make/get calls until I restart FS. I did this:
> 1. log 7
> 2. sofia profile xxxx siptrace on   for each profile/gateway
> 3. restarted router
>
> All three did not solve the problem. The trace and log produced no
> additional lines which is why I am wondering if FS has a problem since the
> trace shows no SIP activity.
>
> 3 gateways with 2 ITSPs
> 2 DSL/WAN lines, 1 static and 1 dynamic
> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is
> the external static ip.
> sofia status profile ... has the right ext ip
> nat_map status shows the dynamic (wrong) IP
> I tried starting with -nonat but that was worse
> the only way to fix is restart FS.
>
> I read the wiki on external nat, auto_nat and everything else many times.
> Thanks Mario
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From pedja at emazu.com  Mon Oct 25 15:42:53 2010
From: pedja at emazu.com (Pedja Delic)
Date: Tue, 26 Oct 2010 00:42:53 +0200
Subject: [Freeswitch-users] Virtual Migration Detected! Syncing Clock
Message-ID: <4CC607ED.6030001@emazu.com>

Hello,
I faced this problem 2 times during test up to 20 concurrent calls:
2010-10-24 23:20:31.552414 [CRIT] switch_time.c:755 Virtual Migration 
Detected! Syncing Clock
2010-10-24 23:20:47.988228 [CRIT] switch_time.c:755 Virtual Migration 
Detected! Syncing Clock
2010-10-24 23:20:52.744154 [CRIT] switch_time.c:755 Virtual Migration 
Detected! Syncing Clock
2010-10-24 23:20:56.027521 [CRIT] switch_time.c:755 Virtual Migration 
Detected! Syncing Clock
2010-10-24 23:20:59.964199 [CRIT] switch_time.c:755 Virtual Migration 
Detected! Syncing Clock
2010-10-24 23:21:05.092063 [CRIT] switch_time.c:755 Virtual Migration 
Detected! Syncing Clock
2010-10-24 23:21:10.620075 [WARNING] switch_scheduler.c:114 Task was 
executed late by 2 seconds 1 heartbeat (core)

The 1st time the whole server went into frozen state. I was able to  
ping network interface but couldn't get response from any service from 
the server. Reboot did the job and server worked fine until error 
occurred the 2nd time when only Freeswitch crashed.
Frequently, during the day, server is faced randomly with high load on 
memory and cpu with up to 20 concurrent calls. Is it possible that high 
cpu load and lack of resource i.e. memory can cause this problem?
Freeswitch consumes  up to 55% of RAM.

Here is box info:
OS: CentOS 5.5 64 bit
CPU: AMD Opteron 1216HE
RAM: 2GB
FreeSWITCH version: 1.0.head (git-8726104 2010-09-15 19-46-23 -0500)
Can anyone drop some light one this issue? Also i want to ask what is 
the recommended configuration for 1000 concurrent calls.

Pedja





From mario_fs at mgtech.com  Mon Oct 25 16:05:54 2010
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 25 Oct 2010 16:05:54 -0700
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <AANLkTin0Zfgm6xf-9_G+O-Obvko-05Cpzxt3n9Ukhwir@mail.gmail.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<AANLkTin0Zfgm6xf-9_G+O-Obvko-05Cpzxt3n9Ukhwir@mail.gmail.com>
Message-ID: <15FC6D87-8988-40B3-AC1E-8E158EEAE87C@mgtech.com>

They look outbound to me and I am using dns names. I was able to get FS to clear this up by doing "nat_map reinit" which is why I think this is a nat problem. I will do the trace you mentioned. I will plug an ip address into one of the gateways to see what happens, they all fail at once. Thanks for responding!
Mario

On Oct 25, 2010, at 3:38 PM, Michael Collins wrote:

> Mario,
> 
> Try doing "sofia global siptrace on" when your system is in this state. It will definitely turn on sip trace for every profile on your system. It looks like these are failed outbound registrations, correct? Are you using domain names or IP addresses for these gateways? I'm just curious. I've seen situations where FS could not resolve the domain name for some odd reason.
> 
> -MC
> 
> On Mon, Oct 25, 2010 at 3:26 PM, Mario <mario_fs at mgtech.com> wrote:
> I really need help on this as I have weeks into this problem. I thought I had it nailed but I guess not. After 5.5 hours I get:
> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed Registration, setting retry to 15 seconds.
> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed Registration, setting retry to 30 seconds.
> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed Registration, setting retry to 15 seconds.
> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed Registration, setting retry to 15 seconds.
> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
> 
> and no way to make/get calls until I restart FS. I did this:
> 1. log 7
> 2. sofia profile xxxx siptrace on   for each profile/gateway
> 3. restarted router
> 
> All three did not solve the problem. The trace and log produced no additional lines which is why I am wondering if FS has a problem since the trace shows no SIP activity.
> 
> 3 gateways with 2 ITSPs
> 2 DSL/WAN lines, 1 static and 1 dynamic
> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the external static ip.
> sofia status profile ... has the right ext ip
> nat_map status shows the dynamic (wrong) IP
> I tried starting with -nonat but that was worse
> the only way to fix is restart FS.
> 
> I read the wiki on external nat, auto_nat and everything else many times.
> Thanks Mario
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Mon Oct 25 16:09:09 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 25 Oct 2010 18:09:09 -0500
Subject: [Freeswitch-users] Virtual Migration Detected! Syncing Clock
In-Reply-To: <4CC607ED.6030001@emazu.com>
References: <4CC607ED.6030001@emazu.com>
Message-ID: <AANLkTik=7dqpzSvGCfM98e19GncnZcTqDTJvV86uoiOY@mail.gmail.com>

Try a multi-core xeon platform.


On Mon, Oct 25, 2010 at 5:42 PM, Pedja Delic <pedja at emazu.com> wrote:
> Hello,
> I faced this problem 2 times during test up to 20 concurrent calls:
> 2010-10-24 23:20:31.552414 [CRIT] switch_time.c:755 Virtual Migration
> Detected! Syncing Clock
> 2010-10-24 23:20:47.988228 [CRIT] switch_time.c:755 Virtual Migration
> Detected! Syncing Clock
> 2010-10-24 23:20:52.744154 [CRIT] switch_time.c:755 Virtual Migration
> Detected! Syncing Clock
> 2010-10-24 23:20:56.027521 [CRIT] switch_time.c:755 Virtual Migration
> Detected! Syncing Clock
> 2010-10-24 23:20:59.964199 [CRIT] switch_time.c:755 Virtual Migration
> Detected! Syncing Clock
> 2010-10-24 23:21:05.092063 [CRIT] switch_time.c:755 Virtual Migration
> Detected! Syncing Clock
> 2010-10-24 23:21:10.620075 [WARNING] switch_scheduler.c:114 Task was
> executed late by 2 seconds 1 heartbeat (core)
>
> The 1st time the whole server went into frozen state. I was able to
> ping network interface but couldn't get response from any service from
> the server. Reboot did the job and server worked fine until error
> occurred the 2nd time when only Freeswitch crashed.
> Frequently, during the day, server is faced randomly with high load on
> memory and cpu with up to 20 concurrent calls. Is it possible that high
> cpu load and lack of resource i.e. memory can cause this problem?
> Freeswitch consumes ?up to 55% of RAM.
>
> Here is box info:
> OS: CentOS 5.5 64 bit
> CPU: AMD Opteron 1216HE
> RAM: 2GB
> FreeSWITCH version: 1.0.head (git-8726104 2010-09-15 19-46-23 -0500)
> Can anyone drop some light one this issue? Also i want to ask what is
> the recommended configuration for 1000 concurrent calls.
>
> Pedja
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From mario_fs at mgtech.com  Mon Oct 25 16:11:21 2010
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 25 Oct 2010 16:11:21 -0700
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
Message-ID: <1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>

Whoops, I am using an IP address for at least one gateway so that is not the problem:
They look outbound to me and I am using dns for 2 and an IP for one so that is not the issue. I was able to get FS to clear this up by doing "nat_map reinit" which is why I think this is a nat problem. I will do the trace you mentioned. I will plug an ip address into one of the gateways to see what happens, they all fail at once. Thanks for responding!
Mario

On Oct 25, 2010, at 3:26 PM, Mario wrote:

> I really need help on this as I have weeks into this problem. I thought I had it nailed but I guess not. After 5.5 hours I get:
> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed Registration, setting retry to 15 seconds.
> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed Registration, setting retry to 30 seconds.
> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed Registration, setting retry to 15 seconds.
> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed Registration, setting retry to 15 seconds.
> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
> 
> and no way to make/get calls until I restart FS. I did this:
> 1. log 7
> 2. sofia profile xxxx siptrace on   for each profile/gateway
> 3. restarted router
> 
> All three did not solve the problem. The trace and log produced no additional lines which is why I am wondering if FS has a problem since the trace shows no SIP activity.
> 
> 3 gateways with 2 ITSPs
> 2 DSL/WAN lines, 1 static and 1 dynamic
> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the external static ip.
> sofia status profile ... has the right ext ip
> nat_map status shows the dynamic (wrong) IP
> I tried starting with -nonat but that was worse
> the only way to fix is restart FS.
> 
> I read the wiki on external nat, auto_nat and everything else many times.
> Thanks Mario
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mcampbellsmith at gmail.com  Mon Oct 25 16:53:47 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 26 Oct 2010 10:53:47 +1100
Subject: [Freeswitch-users] FS not detecting public IP address change
In-Reply-To: <AANLkTikfpRi9Shxiqd2w5nz3ZHsBnrHTt+=rHHnvoDJn@mail.gmail.com>
References: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>
	<AANLkTinEoTcLmUVcpras8-m8pdkfpE=_r0spz1S50iMF@mail.gmail.com>
	<AANLkTi=MoV27FW0Ukd6K+mqmgnkyNgb9hjFx-vBYV1ak@mail.gmail.com>
	<AANLkTimCRfKSWG+5OdQY0=HTt2kOqxwk+0vKvaBNdmAd@mail.gmail.com>
	<AANLkTikfpRi9Shxiqd2w5nz3ZHsBnrHTt+=rHHnvoDJn@mail.gmail.com>
Message-ID: <AANLkTikH3JJCCpL5s_eF2qjjttnEetE0i=nPn87zhBUL@mail.gmail.com>

Hi!

How do I enable debugging to get the UPnP keep alive packets?

Thanks!

On Tue, Oct 26, 2010 at 1:15 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> I need the debug logging from the upnp stuff.  It should look something
> like:
>
> +OK log level 7 [7]
> freeswitch at internal> 2010-10-25 09:13:59.926861 [DEBUG] switch_nat.c:299
> got UPnP keep alive packet:
> NOTIFY * HTTP/1.1
> HOST:239.255.255.250:1900
> Cache-Control:max-age=60
> Location:http://192.168.1.1:5000/rootDesc.xml
> Server: Tomato UPnP/1.0 MiniUPnPd/1.4
> NT:urn:schemas-upnp-org:service:WANIPConnection:1
>
> USN:uuid:00000000-0000-0000-0000-000000000000::urn:schemas-upnp-org:service:WANIPConnection:1
> NTS:ssdp:alive
>
> I also run tomato and haven't had trouble with the upnp support so at least
> we have that part working.
>
> You should see the above keepalive appear periodically every 30s or so.
>  You should see another set of messages when you terminate the DSL
> connection and when that dsl connection comes back online.
>
> On Sat, Oct 23, 2010 at 5:36 PM, Mark Campbell-Smith <
> mcampbellsmith at gmail.com> wrote:
>
>> Hi!
>>
>> The router is a ASUS router but it is running the tomato firmware which
>> runs miniupnd.
>>
>> In the debug I see the message: 2010-10-24 08:02:59.178918 [DEBUG]
>> sofia.c:956 nua_i_outbound: unknown event 8: 101 NAT detected
>>
>> This is what I see below. I hope I enabled all debug messages.  I thought
>> I saw some xml type messages earlier, but not sure how I enabled them.
>>
>> nta_outgoing: RTT is 67.958 ms
>> outbound(0xb6e07c00): NAT binding changed: [210.xx.xxx.xx]:5080 !=
>> [58.xxx.xx.xx]:5080
>> nua(0xb6e07c00): event i_outbound 102 NAT binding changed
>> nua: nua_application_event: entering
>> 2010-10-24 09:05:08.849525 [DEBUG] sofia.c:956 nua_i_outbound: unknown
>> event 8: 102 NAT binding changed
>>
>>
>> On Sun, Oct 24, 2010 at 1:08 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>>> btw: what router are you using?
>>>
>>>
>>> On Sat, Oct 23, 2010 at 9:06 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>>> In a upnp config, FS is depending on the router to notify it of the ip
>>>> address change.  It is possible the upnp message(s) aren't formatted as
>>>> expected.  If you turn on debug logging you'll get the upnp messages on the
>>>> console.  If you can pastebin the capture of those while dropping and
>>>> restarting the dsl connection I can maybe see if there is something obvious
>>>> going on.  The debug should also have messages related to the processing of
>>>> those upnp messages.
>>>>
>>>> On Sat, Oct 23, 2010 at 7:04 AM, Mark Campbell-Smith <
>>>> mcampbellsmith at gmail.com> wrote:
>>>>
>>>>> HI!
>>>>>
>>>>> I know this question must have been answered 100's of times....
>>>>>
>>>>> My adsl is a bit dodgy at the moment and tends to go up and down; which
>>>>> means I get assigned a new ip address from my ISP.  FS is nat'd behind a
>>>>> upnp capable router.
>>>>>
>>>>> The problem is that FS does not seem to be detecting the change.   For
>>>>> both the internal and external profile, I have auto-nat setup, and I have
>>>>> tried stun and host settings in vars.conf.  When I issue a nat_map status I
>>>>> see my old IP address, and if I then issue a nat_map reinit, I see the new
>>>>> public IP address.  How can I get this to be automatic?
>>>>>
>>>>> Also the public IP address shown in the nat_map status is not the same
>>>>> as shown in the sofia profile internal/external printouts.  Why is this?
>>>>>
>>>>> I'm sure this is configuration, just not sure what to change.
>>>>>
>>>>> Thanks!
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From covici at ccs.covici.com  Mon Oct 25 17:06:38 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 25 Oct 2010 20:06:38 -0400
Subject: [Freeswitch-users] transfer problem
In-Reply-To: <AANLkTinVAP-5A4KA7gz6hbRUVw7qh6Kc+ty48gs4eucZ@mail.gmail.com>
References: <13464.1287506974@ccs.covici.com>
	<AANLkTikh67cGuMn3v4D3QZ+AuQ0vWrLE_fRhL5+g1fg7@mail.gmail.com>
	<19713.1288004756@ccs.covici.com>
	<AANLkTinA4Egvfr1N-EvLun6rh-BR=WxwutHdOryACnYp@mail.gmail.com>
	<28001.1288041100@ccs.covici.com>
	<AANLkTinVAP-5A4KA7gz6hbRUVw7qh6Kc+ty48gs4eucZ@mail.gmail.com>
Message-ID: <29557.1288051598@ccs.covici.com>

I can capture the log, but let me tell you the steps.  202 does call
7034754612, then it calls 7000, which is an ivr which transfers to an
extensionn in the context conferences, which in turn puts the caller in
a conference.  So 202 calls 7034754612 and then is in this conference
whose context is not default.  Now I flash the hook on 202, so
7034754612 can hear the conference and also can hear 202.  Now, what I
was hoping would happen is that I could hang up 202 and 4754612 would be
in the conference, but instead 4754612 hangs up and  this seems to be
because the system tries to transfer 4754612 to theconference name but
in the default context and I get 
2010-10-25 17:09:46.137372 [DEBUG] mod_enum.c:204 Unable to lookup NAPTR
record for e164.org: valid domain but no data of requested type
and the 4754612 hangs up.

I hope this clarifies what the problem is.

Michael Collins <msc at freeswitch.org> wrote:

> John,
> 
> Perhaps I've missed something. Can you confirm exactly what steps you are
> taking? I don't want to make any assumptions. I can see that ext 202 calls
> out to 7034754612 and then 202 also makes a call to ext 7000. What I don't
> know is if that's a transfer or 3-way or what. Also, can you capture the log
> again, this time with a sip trace? Use "sofia global siptrace on".
> 
> Thanks,
> MC
> 
> On Mon, Oct 25, 2010 at 2:11 PM, <covici at ccs.covici.com> wrote:
> 
> > Thanks for the correction -- however still same result.  I did not think
> > it would matter in this case because the call did reach the phone and
> > everything after that is local.  Question is why does it try to transfer
> > to the wrong context after the local extension hangs up?
> >
> > Michael Collins <msc at freeswitch.org> wrote:
> >
> > > John,
> > >
> > > I see two issues. At line #174 of your pastebin there is a message about
> > > "invalid gateway". I suspect that the gateway name should be "flowroute"
> > and
> > > not "fflowroute".
> > >
> > > At line #180 you have an invalid number format error. I would fix the
> > > invalid gateway first and retest as this second error may be a result of
> > the
> > > bad gateway.
> > >
> > > -MC
> > >
> > > On Mon, Oct 25, 2010 at 4:05 AM, <covici at ccs.covici.com> wrote:
> > >
> > > > OK, the debug log is here http://pastebin.freeswitch.org/14307.  The
> > > > relevant part of the configs is below.
> > > >    <extension name="conference_ivr">
> > > >      <condition field="destination_number" expression="^7000$">
> > > >        <action application="answer"/>
> > > > <action application="sleep" data="500"/>
> > > > <action application="set" data="api_hangup_hook=system sayconf -a local
> >  -a
> > > > ${caller_id_number} -a hung -a up"/>
> > > >        <action application="ivr" data="conference_ivr"/>
> > > >      </condition>
> > > >    </extension>
> > > > and the ivr just asks for passwords and send the call to an extension
> > in
> > > > context conferences.
> > > >
> > > > Hope this helps.
> > > >
> > > > Thanks.
> > > >
> > > > Michael Collins <msc at freeswitch.org> wrote:
> > > >
> > > > > Please supply the configs and a debug trace. My guess is that there's
> > > > > probably just a simple config element that is wrong or missing.
> > > > >
> > > > > -MC
> > > > >
> > > > > On Tue, Oct 19, 2010 at 9:49 AM, <covici at ccs.covici.com> wrote:
> > > > >
> > > > > > Hi.  I have an ivr which connects to conferences which I have in
> > their
> > > > > > own context rather than the default.  The problem is that if I call
> > > > > > someone else and have a 3-way between the other party and the
> > > > > > conference, its fine till I hang up and then fs tries to transfer
> > to
> > > > the
> > > > > > correct conference name, but uses default for the context instead.
> >  Is
> > > > > > this expected behavior or should I file a bug?
> > > > > >
> > > > > > Thanks.
> > > > > >
> > > > > > --
> > > > > > Your life is like a penny.  You're going to lose it.  The question
> > is:
> > > > > > How do
> > > > > > you spend it?
> > > > > >
> > > > > >         John Covici
> > > > > >         covici at ccs.covici.com
> > > > > >
> > > > > > _______________________________________________
> > > > > > FreeSWITCH-users mailing list
> > > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > > UNSUBSCRIBE:
> > > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > > > http://www.freeswitch.org
> > > > > >
> > > > >
> > > > > ----------------------------------------------------
> > > > > Alternatives:
> > > > >
> > > > > ----------------------------------------------------
> > > > > _______________________________________________
> > > > > FreeSWITCH-users mailing list
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > > http://www.freeswitch.org
> > > >
> > > > --
> > > > Your life is like a penny.  You're going to lose it.  The question is:
> > > > How do
> > > > you spend it?
> > > >
> > > >         John Covici
> > > >         covici at ccs.covici.com
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > http://www.freeswitch.org
> > > >
> > >
> > > ----------------------------------------------------
> > > Alternatives:
> > >
> > > ----------------------------------------------------
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From anthony.minessale at gmail.com  Mon Oct 25 18:20:32 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 25 Oct 2010 20:20:32 -0500
Subject: [Freeswitch-users] transfer problem
In-Reply-To: <29557.1288051598@ccs.covici.com>
References: <13464.1287506974@ccs.covici.com>
	<AANLkTikh67cGuMn3v4D3QZ+AuQ0vWrLE_fRhL5+g1fg7@mail.gmail.com>
	<19713.1288004756@ccs.covici.com>
	<AANLkTinA4Egvfr1N-EvLun6rh-BR=WxwutHdOryACnYp@mail.gmail.com>
	<28001.1288041100@ccs.covici.com>
	<AANLkTinVAP-5A4KA7gz6hbRUVw7qh6Kc+ty48gs4eucZ@mail.gmail.com>
	<29557.1288051598@ccs.covici.com>
Message-ID: <AANLkTi==CoAgm9_foaPEWDpBmME2HWq5hoskDNjE4MA6@mail.gmail.com>

set the variable force_transfer_context to the desired context on all
channels involved

On Mon, Oct 25, 2010 at 7:06 PM,  <covici at ccs.covici.com> wrote:
> I can capture the log, but let me tell you the steps. ?202 does call
> 7034754612, then it calls 7000, which is an ivr which transfers to an
> extensionn in the context conferences, which in turn puts the caller in
> a conference. ?So 202 calls 7034754612 and then is in this conference
> whose context is not default. ?Now I flash the hook on 202, so
> 7034754612 can hear the conference and also can hear 202. ?Now, what I
> was hoping would happen is that I could hang up 202 and 4754612 would be
> in the conference, but instead 4754612 hangs up and ?this seems to be
> because the system tries to transfer 4754612 to theconference name but
> in the default context and I get
> 2010-10-25 17:09:46.137372 [DEBUG] mod_enum.c:204 Unable to lookup NAPTR
> record for e164.org: valid domain but no data of requested type
> and the 4754612 hangs up.
>
> I hope this clarifies what the problem is.
>
> Michael Collins <msc at freeswitch.org> wrote:
>
>> John,
>>
>> Perhaps I've missed something. Can you confirm exactly what steps you are
>> taking? I don't want to make any assumptions. I can see that ext 202 calls
>> out to 7034754612 and then 202 also makes a call to ext 7000. What I don't
>> know is if that's a transfer or 3-way or what. Also, can you capture the log
>> again, this time with a sip trace? Use "sofia global siptrace on".
>>
>> Thanks,
>> MC
>>
>> On Mon, Oct 25, 2010 at 2:11 PM, <covici at ccs.covici.com> wrote:
>>
>> > Thanks for the correction -- however still same result. ?I did not think
>> > it would matter in this case because the call did reach the phone and
>> > everything after that is local. ?Question is why does it try to transfer
>> > to the wrong context after the local extension hangs up?
>> >
>> > Michael Collins <msc at freeswitch.org> wrote:
>> >
>> > > John,
>> > >
>> > > I see two issues. At line #174 of your pastebin there is a message about
>> > > "invalid gateway". I suspect that the gateway name should be "flowroute"
>> > and
>> > > not "fflowroute".
>> > >
>> > > At line #180 you have an invalid number format error. I would fix the
>> > > invalid gateway first and retest as this second error may be a result of
>> > the
>> > > bad gateway.
>> > >
>> > > -MC
>> > >
>> > > On Mon, Oct 25, 2010 at 4:05 AM, <covici at ccs.covici.com> wrote:
>> > >
>> > > > OK, the debug log is here http://pastebin.freeswitch.org/14307. ?The
>> > > > relevant part of the configs is below.
>> > > > ? ?<extension name="conference_ivr">
>> > > > ? ? ?<condition field="destination_number" expression="^7000$">
>> > > > ? ? ? ?<action application="answer"/>
>> > > > <action application="sleep" data="500"/>
>> > > > <action application="set" data="api_hangup_hook=system sayconf -a local
>> > ?-a
>> > > > ${caller_id_number} -a hung -a up"/>
>> > > > ? ? ? ?<action application="ivr" data="conference_ivr"/>
>> > > > ? ? ?</condition>
>> > > > ? ?</extension>
>> > > > and the ivr just asks for passwords and send the call to an extension
>> > in
>> > > > context conferences.
>> > > >
>> > > > Hope this helps.
>> > > >
>> > > > Thanks.
>> > > >
>> > > > Michael Collins <msc at freeswitch.org> wrote:
>> > > >
>> > > > > Please supply the configs and a debug trace. My guess is that there's
>> > > > > probably just a simple config element that is wrong or missing.
>> > > > >
>> > > > > -MC
>> > > > >
>> > > > > On Tue, Oct 19, 2010 at 9:49 AM, <covici at ccs.covici.com> wrote:
>> > > > >
>> > > > > > Hi. ?I have an ivr which connects to conferences which I have in
>> > their
>> > > > > > own context rather than the default. ?The problem is that if I call
>> > > > > > someone else and have a 3-way between the other party and the
>> > > > > > conference, its fine till I hang up and then fs tries to transfer
>> > to
>> > > > the
>> > > > > > correct conference name, but uses default for the context instead.
>> > ?Is
>> > > > > > this expected behavior or should I file a bug?
>> > > > > >
>> > > > > > Thanks.
>> > > > > >
>> > > > > > --
>> > > > > > Your life is like a penny. ?You're going to lose it. ?The question
>> > is:
>> > > > > > How do
>> > > > > > you spend it?
>> > > > > >
>> > > > > > ? ? ? ? John Covici
>> > > > > > ? ? ? ? covici at ccs.covici.com
>> > > > > >
>> > > > > > _______________________________________________
>> > > > > > FreeSWITCH-users mailing list
>> > > > > > FreeSWITCH-users at lists.freeswitch.org
>> > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > > > > UNSUBSCRIBE:
>> > > > http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > > > > http://www.freeswitch.org
>> > > > > >
>> > > > >
>> > > > > ----------------------------------------------------
>> > > > > Alternatives:
>> > > > >
>> > > > > ----------------------------------------------------
>> > > > > _______________________________________________
>> > > > > FreeSWITCH-users mailing list
>> > > > > FreeSWITCH-users at lists.freeswitch.org
>> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > > > UNSUBSCRIBE:
>> > http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > > > http://www.freeswitch.org
>> > > >
>> > > > --
>> > > > Your life is like a penny. ?You're going to lose it. ?The question is:
>> > > > How do
>> > > > you spend it?
>> > > >
>> > > > ? ? ? ? John Covici
>> > > > ? ? ? ? covici at ccs.covici.com
>> > > >
>> > > > _______________________________________________
>> > > > FreeSWITCH-users mailing list
>> > > > FreeSWITCH-users at lists.freeswitch.org
>> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > > UNSUBSCRIBE:
>> > http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > > http://www.freeswitch.org
>> > > >
>> > >
>> > > ----------------------------------------------------
>> > > Alternatives:
>> > >
>> > > ----------------------------------------------------
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> >
>> > --
>> > Your life is like a penny. ?You're going to lose it. ?The question is:
>> > How do
>> > you spend it?
>> >
>> > ? ? ? ? John Covici
>> > ? ? ? ? covici at ccs.covici.com
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> ----------------------------------------------------
>> Alternatives:
>>
>> ----------------------------------------------------
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> --
> Your life is like a penny. ?You're going to lose it. ?The question is:
> How do
> you spend it?
>
> ? ? ? ? John Covici
> ? ? ? ? covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From rupa at rupa.com  Mon Oct 25 21:13:35 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 25 Oct 2010 23:13:35 -0500
Subject: [Freeswitch-users] FS not detecting public IP address change
In-Reply-To: <AANLkTikH3JJCCpL5s_eF2qjjttnEetE0i=nPn87zhBUL@mail.gmail.com>
References: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>
	<AANLkTinEoTcLmUVcpras8-m8pdkfpE=_r0spz1S50iMF@mail.gmail.com>
	<AANLkTi=MoV27FW0Ukd6K+mqmgnkyNgb9hjFx-vBYV1ak@mail.gmail.com>
	<AANLkTimCRfKSWG+5OdQY0=HTt2kOqxwk+0vKvaBNdmAd@mail.gmail.com>
	<AANLkTikfpRi9Shxiqd2w5nz3ZHsBnrHTt+=rHHnvoDJn@mail.gmail.com>
	<AANLkTikH3JJCCpL5s_eF2qjjttnEetE0i=nPn87zhBUL@mail.gmail.com>
Message-ID: <AANLkTimzkb7qY8nW3M2afTPn18=OxY6jQye8D9TKr9Ts@mail.gmail.com>

Just turn up console logging to level 7.

/log 7 if using fs_cli

On Mon, Oct 25, 2010 at 6:53 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> Hi!
>
> How do I enable debugging to get the UPnP keep alive packets?
>
> Thanks!
>
> On Tue, Oct 26, 2010 at 1:15 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> I need the debug logging from the upnp stuff.  It should look something
>> like:
>>
>> +OK log level 7 [7]
>> freeswitch at internal> 2010-10-25 09:13:59.926861 [DEBUG] switch_nat.c:299
>> got UPnP keep alive packet:
>> NOTIFY * HTTP/1.1
>> HOST:239.255.255.250:1900
>> Cache-Control:max-age=60
>> Location:http://192.168.1.1:5000/rootDesc.xml
>> Server: Tomato UPnP/1.0 MiniUPnPd/1.4
>> NT:urn:schemas-upnp-org:service:WANIPConnection:1
>>
>> USN:uuid:00000000-0000-0000-0000-000000000000::urn:schemas-upnp-org:service:WANIPConnection:1
>> NTS:ssdp:alive
>>
>> I also run tomato and haven't had trouble with the upnp support so at
>> least we have that part working.
>>
>> You should see the above keepalive appear periodically every 30s or so.
>>  You should see another set of messages when you terminate the DSL
>> connection and when that dsl connection comes back online.
>>
>> On Sat, Oct 23, 2010 at 5:36 PM, Mark Campbell-Smith <
>> mcampbellsmith at gmail.com> wrote:
>>
>>> Hi!
>>>
>>> The router is a ASUS router but it is running the tomato firmware which
>>> runs miniupnd.
>>>
>>> In the debug I see the message: 2010-10-24 08:02:59.178918 [DEBUG]
>>> sofia.c:956 nua_i_outbound: unknown event 8: 101 NAT detected
>>>
>>> This is what I see below. I hope I enabled all debug messages.  I thought
>>> I saw some xml type messages earlier, but not sure how I enabled them.
>>>
>>> nta_outgoing: RTT is 67.958 ms
>>> outbound(0xb6e07c00): NAT binding changed: [210.xx.xxx.xx]:5080 !=
>>> [58.xxx.xx.xx]:5080
>>> nua(0xb6e07c00): event i_outbound 102 NAT binding changed
>>> nua: nua_application_event: entering
>>> 2010-10-24 09:05:08.849525 [DEBUG] sofia.c:956 nua_i_outbound: unknown
>>> event 8: 102 NAT binding changed
>>>
>>>
>>> On Sun, Oct 24, 2010 at 1:08 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>>> btw: what router are you using?
>>>>
>>>>
>>>> On Sat, Oct 23, 2010 at 9:06 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>>
>>>>> In a upnp config, FS is depending on the router to notify it of the ip
>>>>> address change.  It is possible the upnp message(s) aren't formatted as
>>>>> expected.  If you turn on debug logging you'll get the upnp messages on the
>>>>> console.  If you can pastebin the capture of those while dropping and
>>>>> restarting the dsl connection I can maybe see if there is something obvious
>>>>> going on.  The debug should also have messages related to the processing of
>>>>> those upnp messages.
>>>>>
>>>>> On Sat, Oct 23, 2010 at 7:04 AM, Mark Campbell-Smith <
>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>
>>>>>> HI!
>>>>>>
>>>>>> I know this question must have been answered 100's of times....
>>>>>>
>>>>>> My adsl is a bit dodgy at the moment and tends to go up and down;
>>>>>> which means I get assigned a new ip address from my ISP.  FS is nat'd behind
>>>>>> a upnp capable router.
>>>>>>
>>>>>> The problem is that FS does not seem to be detecting the change.   For
>>>>>> both the internal and external profile, I have auto-nat setup, and I have
>>>>>> tried stun and host settings in vars.conf.  When I issue a nat_map status I
>>>>>> see my old IP address, and if I then issue a nat_map reinit, I see the new
>>>>>> public IP address.  How can I get this to be automatic?
>>>>>>
>>>>>> Also the public IP address shown in the nat_map status is not the same
>>>>>> as shown in the sofia profile internal/external printouts.  Why is this?
>>>>>>
>>>>>> I'm sure this is configuration, just not sure what to change.
>>>>>>
>>>>>> Thanks!
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> -Rupa
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From pmhshz at gmail.com  Mon Oct 25 23:03:24 2010
From: pmhshz at gmail.com (MohammedShehzad)
Date: Tue, 26 Oct 2010 11:33:24 +0530
Subject: [Freeswitch-users] Delay in Voicemai's MWI NOTIFY
In-Reply-To: <AANLkTim0wL3hjB8tMtgXmN8o0emN_Jqck4ipq0pYKGPR@mail.gmail.com>
References: <AANLkTimbndYg-ed3U8f6f5tNGFu_3aQu+cNohCb576Cs@mail.gmail.com>
	<AANLkTim0wL3hjB8tMtgXmN8o0emN_Jqck4ipq0pYKGPR@mail.gmail.com>
Message-ID: <AANLkTinM_NZ11GZ+=G18C1PtVT7En-doZ+462YUoYzC1@mail.gmail.com>

Hi,

I have already aliased the sip profile to the domain name used in
voicemail application call.

freeswitch at localhost.localdomain> sofia status

                     Name          Type
       Data      State
=================================================================================================
                 internal       profile
sip:mod_sofia at 192.168.1.33:5260      RUNNING (0)
                 external       profile
sip:mod_sofia at 192.168.1.33:5160      RUNNING (0)
             192.168.1.33         alias
   internal      ALIASED
=================================================================================================
2 profiles 1 alias


Voicemail application is called using below command:
For leaving voicemail:
<action application="voicemail" data="default ${domain_name}
${destination_number}"/>

For checking voicemail:
<action application="voicemail" data="check default ${domain_name}
${username}"/>

One more thing is that I am using number alias here in user directory.
sample user profile is like:
	<user id="654321" number-alias="05-0090E802668F">
Users are registered using alias '05-0090E802668F'. Will it cause any issue?

Thanks Anthony for prompt reply.

Regards,
MohammedShehzad



On Mon, Oct 25, 2010 at 10:35 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> I hope either this sinks in or someone documents it because I have
> explained this numerous times.
>
> Either you need to supply the profile name to the voicemail app call
>
> ?<action application="voicemail" data="check default ${domain_name}@internal"/>
>
> or
>
> you need to make sure your domain used in the call to voicemail
> (${domain_name} in the above example) is aliased to the sip profile
> you are using.
>
> you have obviously strayed from the default config where this works as
> expected by changing the domain name, the profile name, or the aliases
> to no longer align.
>
>
>
>
> On Mon, Oct 25, 2010 at 11:13 AM, MohammedShehzad <pmhshz at gmail.com> wrote:
>> Hi all,
>>
>> Is there any way I can configure FreeSWITCH, to send NOTIFY for
>> Voicemail's MWI to registered phones whenever new vm comes or deleted?
>> The issue is that FreeSWITCH sends NOTIFY to update mailbox status
>> only when phone re-register.
>> One way I can decrease the register expiry time on phones. But It
>> seems not proper way. I think NOTIFY should be sent independently when
>> message get received or deleted.
>>
>>
>> Regards,
>> MohammedShehzad
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From erkan at speedingtrade.com  Tue Oct 26 01:12:22 2010
From: erkan at speedingtrade.com (=?iso-8859-9?B?RXJrYW4g3G5s/A==?=)
Date: Tue, 26 Oct 2010 11:12:22 +0300
Subject: [Freeswitch-users] Sofia_reg.c:816 Can not do authorization without
	a complete from header
Message-ID: <81C2CEF80046FB4F863A60D4347DD33A0C566A@server1.st.local>

Hi all,

 

Today I have many error in my FS.

Sofia_reg.c:816  Can not do authorization without a complete  from header

 

What does it mean?

Is there a problem in my FS or by the client?

The client work since 3 months without errors, until tomorrow morning the many sofia_reg 816 errors is during up.

The wiki doesn't give any information's about this error.

 

Thank you for some information's.

 

 

Many error during up 

2010-10-26 11:01:58.448513 [ERR] sofia_reg.c:816 Can not do authorization without a complete from header

2010-10-26 11:01:59.168193 [ERR] sofia_reg.c:816 Can not do authorization without a complete from header

2010-10-26 11:01:59.320527 [NOTICE] sofia.c:3739 Channel [sofia/external/0xxxxx42336] has been answe

2010-10-26 11:01:59.379117 [NOTICE] switch_ivr_bridge.c:306 Channel [sofia/internal/99014 at xxxx

2010-10-26 11:01:59.470908 [ERR] sofia_reg.c:816 Can not do authorization without a complete from header

2010-10-26 11:01:59.992359 [ERR] sofia_reg.c:816 Can not do authorization without a complete from header

2010-10-26 11:02:00.293121 [ERR] sofia_reg.c:816 Can not do authorization without a complete from header

2010-10-26 11:02:00.394677 [ERR] sofia_reg.c:816 Can not do authorization without a complete from header

2010-10-26 11:02:00.595836 [ERR] sofia_reg.c:816 Can not do authorization without a complete from header

2010-10-26 11:02:01.311611 [ERR] sofia_reg.c:816 Can not do authorization without a complete from header

2010-10-26 11:02:02.30315 [ERR] sofia_reg.c:816 Can not do authorization without a complete from header

2010-10-26 11:02:02.655275 [ERR] sofia_reg.c:816 Can not do authorization without a complete from header

2010-10-26 11:02:03.820239 [ERR] sofia_reg.c:816 Can not do authorization without a complete from header

2010-10-26 11:02:03.989174 [ERR] sofia_reg.c:816 Can not do authorization without a complete from header

2010-10-26 11:02:04.784045 [NOTICE] sofia.c:3849 Hangup sofia/external/0xxx463995 [CS_EXCHANGE_MED

2010-10-26 11:02:04.799669 [NOTICE] switch_ivr_bridge.c:419 Hangup sofia/internal/99015 at xxxxxx [

2010-10-26 11:02:04.799669 [NOTICE] switch_core_session.c:1086 Session 1517 (sofia/external/0xxxx3046

2010-10-26 11:02:04.799669 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/0xxx046

2010-10-26 11:02:04.803575 [NOTICE] switch_core_session.c:1086 Session 1516 (sofia/internal/99015 at xxxx

2010-10-26 11:02:04.803575 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/99015 at xxxx

2010-10-26 11:02:05.160974 [ERR] sofia_reg.c:816 Can not do authorization without a complete from header

2010-10-26 11:02:05.626764 [ERR] sofia_reg.c:816 Can not do authorization without a complete from header

2010-10-26 11:02:05.746874 [NOTICE] sofia.c:3849 Hangup sofia/internal/24000 at xxxxxx [CS_EXECUTE]

2010-10-26 11:02:05.760545 [NOTICE] switch_ivr_bridge.c:503 Hangup sofia/external/0xxxxx38 [CS

2010-10-26 11:02:05.760545 [NOTICE] switch_core_session.c:1086 Session 1513 (sofia/external/0xxx5203

2010-10-26 11:02:05.760545 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/xxxx5203

2010-10-26 11:02:05.764451 [NOTICE] switch_core_session.c:1086 Session 1512 (sofia/internal/24000 at xxxx

2010-10-26 11:02:05.764451 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/24000 at xxx

2010-10-26 11:02:05.932409 [ERR] sofia_reg.c:816 Can not do authorization without a complete from header

2010-10-26 11:02:06.132591 [ERR] sofia_reg.c:816 Can not do authorization without a complete from header

2010-10-26 11:02:06.382575 [NOTICE] sofia.c:322 Hangup sofia/internal/24001 at xxxxxx [CS_EXECUTE]

2010-10-26 11:02:06.400152 [NOTICE] switch_ivr_bridge.c:503 Hangup sofia/external/00xxxxxxxxxx5671 [CS_

2010-10-26 11:02:06.458742 [NOTICE] switch_core_session.c:1086 Session 1379 (sofia/external/xxx37

2010-10-26 11:02:06.458742 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/0xxxx137

2010-10-26 11:02:06.460695 [NOTICE] switch_core_session.c:1086 Session 1378 (sofia/internal/24001 at xxxx

2010-10-26 11:02:06.460695 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/24001 at xxxx

2010-10-26 11:02:06.585687 [ERR] sofia_reg.c:816 Can not do authorization without a complete from header

2010-10-26 11:02:06.849342 [ERR] sofia_reg.c:816 Can not do authorization without a complete from header

2010-10-26 11:02:07.200882 [ERR] sofia_reg.c:816 Can not do authorization without a complete from header

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From erkan at speedingtrade.com  Tue Oct 26 01:44:42 2010
From: erkan at speedingtrade.com (=?iso-8859-9?B?RXJrYW4g3G5s/A==?=)
Date: Tue, 26 Oct 2010 11:44:42 +0300
Subject: [Freeswitch-users] Timeout problem
Message-ID: <81C2CEF80046FB4F863A60D4347DD33A0C566B@server1.st.local>

Hi all,

 

I have sometimes a really interesting problem.

Sometimes my server is not reachable from xlite.

Xlite give me Request Timeout error.

 

Also if I start xlite on my server they can not be registered on FS.

If I try   "telnet xxx.xxx.xxx.xxx 5060" a connection is established.

 

HTTP and all other services are available. But FS are not available.

For test I installed another sip server on the machine (Voipswitch Demo) so I see that works.

 

But FS gives not response.

I check my network and many other things.

Also I change my port on vars.xml

 

  <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060" />

  <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061" />

 

But nothing is changed.

After 1 or 2 hours FS is available and works again.

This problem is during up 1 time in 1 month.

 

I don`t understand what is wrong.

Maybe a mistake in my configuration?

Maybe everybody have an idea?

 

 

Greetings

Erkan

 

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From pmhshz at gmail.com  Tue Oct 26 01:59:05 2010
From: pmhshz at gmail.com (MohammedShehzad)
Date: Tue, 26 Oct 2010 14:29:05 +0530
Subject: [Freeswitch-users] Delay in Voicemai's MWI NOTIFY
In-Reply-To: <AANLkTinM_NZ11GZ+=G18C1PtVT7En-doZ+462YUoYzC1@mail.gmail.com>
References: <AANLkTimbndYg-ed3U8f6f5tNGFu_3aQu+cNohCb576Cs@mail.gmail.com>
	<AANLkTim0wL3hjB8tMtgXmN8o0emN_Jqck4ipq0pYKGPR@mail.gmail.com>
	<AANLkTinM_NZ11GZ+=G18C1PtVT7En-doZ+462YUoYzC1@mail.gmail.com>
Message-ID: <AANLkTinquYb=+7-q4-tutUKPaKf5MxytoNsWNNekwzCV@mail.gmail.com>

I think I found the solution/work around:

Updated wiki.
http://wiki.freeswitch.org/wiki/Mod_voicemail#2._Force_MWI_Info_in_Sofia

Regards,
MohammedShehzad

Double check your number-alias settings. I think you need the alphanumeric
> value in the id and the numeric value in the number-alias.
>
> -MC
>
> On Tue, Oct 26, 2010 at 11:33 AM, MohammedShehzad <pmhshz at gmail.com>wrote:
>
>> Hi,
>>
>> I have already aliased the sip profile to the domain name used in
>> voicemail application call.
>>
>> freeswitch at localhost.localdomain> sofia status
>>
>>                     Name          Type
>>       Data      State
>>
>> =================================================================================================
>>                 internal       profile
>> sip:mod_sofia at 192.168.1.33:5260      RUNNING (0)
>>                 external       profile
>> sip:mod_sofia at 192.168.1.33:5160      RUNNING (0)
>>             192.168.1.33         alias
>>   internal      ALIASED
>>
>> =================================================================================================
>> 2 profiles 1 alias
>>
>>
>> Voicemail application is called using below command:
>> For leaving voicemail:
>> <action application="voicemail" data="default ${domain_name}
>> ${destination_number}"/>
>>
>> For checking voicemail:
>> <action application="voicemail" data="check default ${domain_name}
>> ${username}"/>
>>
>> One more thing is that I am using number alias here in user directory.
>> sample user profile is like:
>>        <user id="654321" number-alias="05-0090E802668F">
>> Users are registered using alias '05-0090E802668F'. Will it cause any
>> issue?
>>
>> Thanks Anthony for prompt reply.
>>
>> Regards,
>> MohammedShehzad
>>
>>
>>
>> On Mon, Oct 25, 2010 at 10:35 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > I hope either this sinks in or someone documents it because I have
>> > explained this numerous times.
>> >
>> > Either you need to supply the profile name to the voicemail app call
>> >
>> >  <action application="voicemail" data="check default
>> ${domain_name}@internal"/>
>> >
>> > or
>> >
>> > you need to make sure your domain used in the call to voicemail
>> > (${domain_name} in the above example) is aliased to the sip profile
>> > you are using.
>> >
>> > you have obviously strayed from the default config where this works as
>> > expected by changing the domain name, the profile name, or the aliases
>> > to no longer align.
>> >
>> >
>> >
>> >
>> > On Mon, Oct 25, 2010 at 11:13 AM, MohammedShehzad <pmhshz at gmail.com>
>> wrote:
>> >> Hi all,
>> >>
>> >> Is there any way I can configure FreeSWITCH, to send NOTIFY for
>> >> Voicemail's MWI to registered phones whenever new vm comes or deleted?
>> >> The issue is that FreeSWITCH sends NOTIFY to update mailbox status
>> >> only when phone re-register.
>> >> One way I can decrease the register expiry time on phones. But It
>> >> seems not proper way. I think NOTIFY should be sent independently when
>> >> message get received or deleted.
>> >>
>> >>
>> >> Regards,
>> >> MohammedShehzad
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> 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 <MSN%3Aanthony_minessale at hotmail.com>
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>
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From abubacker at bksystems.co.in  Tue Oct 26 02:40:36 2010
From: abubacker at bksystems.co.in (abubacker)
Date: Tue, 26 Oct 2010 15:10:36 +0530
Subject: [Freeswitch-users] Is anyone familiar with Telemaque?
In-Reply-To: <4CAC4EB4.6080901@telemaque.fr>
References: <AANLkTimib+3T7L9sP28HN6BREJF5ca2j2nou-84YU=Ln@mail.gmail.com>	<4CAAE5DE.6080407@telemaque.fr>	<4CABF9D5.7000501@bksys.co.in>
	<4CAC4EB4.6080901@telemaque.fr>
Message-ID: <4CC6A214.1080601@bksys.co.in>

On Wednesday 06 October 2010 03:55 PM, Tristan Mah? wrote:
> Hi,
>
> The telco cluster is powered by IBM servers ( from dual Xeon/8gb ram 
> to dual quad core/16gb ram depending on the node ) with FC13 and 
> latest FS git ( well almost always, as of today I'm a week late, but 
> will upgrade soon again to benefit of the amazing work the dev do ).
>
> If you have more precise questions, don't hesitate, I'm sorry if I 
> can't reveal all the details on how we're doing things,but I'm sure 
> you can understand that...
>
> Regards,
>
> Tristan.
> Le 10/06/2010 06:23 AM, abubacker a ?crit :
>> On Tuesday 05 October 2010 02:16 PM, Tristan Mah? wrote:
>>> There's someone working for them on list :)
>>>
>>> Feel free to ask me questions Michael.
>>>
>>> Regards,
>>>
>>> Gled.
>>>
>>> Le 10/04/2010 08:39 PM, Michael Collins a ?crit :
>>>> Dear FreeSWITCH Community,
>>>>
>>>> Some of you may have stumbled upon this news item:
>>>>
>>>> http://blogs.oracle.com/mysql/2010/10/innovating_with_open_source_call_center_services.html
>>>>
>>>> Evidently there is a company in France (Telemaque) who is using 
>>>> MySQL + Kamailio + FreeSWITCH for some heavy duty call processing. 
>>>> If you are at all familiar with this company please let me know. 
>>>> I'd like to learn more about what they are doing. (Je ne parle pas 
>>>> boucoup de Francais :( )
>>>>
>>>> Thanks for your help,
>>>> MC
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>      
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>    
>>
>> "Feel free to ask me questions Michael"
>>
>> I think I must use this opportunity ,
>> please specify the software ( OS ) and the hardwares required to 
>> handle the heavy duty call
>> processing and also specify the FreeSWITCH version.
>> I guess you could answer this very precisely.
>>
>> Thanks in Advance !
>> -- 
>> Best regards,
>> N.Abubacker ,
>> Associate system engineer ,
>> bk systems pvt ltd ,
>> Ph : 9144-43902701
>>
>> Disclaimer:http://www.bksystems.co.in/email-policy
>>      
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>      
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
Dear Tristan ,
Knew this is an old post , but even though I wanted this to continue to 
get some more details ,
what mod you are using  to handle more calls ? ( hope its mod_fifo )
I am working in mod_fifo , some times it throws socketerror and makes 
the entire freeswitch hang
when more than 25 calls in a queue.

My question is can u give me some details about the queue and member 
configuration ( if possible )

If you are not using mod_fifo , just tell me the general configuration 
required to fine tune the
freeswitch.

Thanks Again !

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

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

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From erkan at speedingtrade.com  Tue Oct 26 03:34:15 2010
From: erkan at speedingtrade.com (=?iso-8859-9?B?RXJrYW4g3G5s/A==?=)
Date: Tue, 26 Oct 2010 13:34:15 +0300
Subject: [Freeswitch-users] Timeout problem
References: <81C2CEF80046FB4F863A60D4347DD33A0C566B@server1.st.local>
Message-ID: <81C2CEF80046FB4F863A60D4347DD33A0C566C@server1.st.local>

In my vars.xml under ip settings are stun:stun.freeswitch.org

 

?s that possible if the stun server not reachable or not available so the FS gives the explained problems?

 

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Erkan ?nl?
Sent: Tuesday, October 26, 2010 11:45 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Timeout problem

 

Hi all,

 

I have sometimes a really interesting problem.

Sometimes my server is not reachable from xlite.

Xlite give me Request Timeout error.

 

Also if I start xlite on my server they can not be registered on FS.

If I try   "telnet xxx.xxx.xxx.xxx 5060" a connection is established.

 

HTTP and all other services are available. But FS are not available.

For test I installed another sip server on the machine (Voipswitch Demo) so I see that works.

 

But FS gives not response.

I check my network and many other things.

Also I change my port on vars.xml

 

  <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060" />

  <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061" />

 

But nothing is changed.

After 1 or 2 hours FS is available and works again.

This problem is during up 1 time in 1 month.

 

I don`t understand what is wrong.

Maybe a mistake in my configuration?

Maybe everybody have an idea?

 

 

Greetings

Erkan

 

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From a.afzali2003 at gmail.com  Tue Oct 26 03:46:30 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Tue, 26 Oct 2010 14:16:30 +0330
Subject: [Freeswitch-users] Error Loading mod_h323
Message-ID: <AANLkTi=ezKDAZMAsJ1zYKKg24Qfc4Yjru--Uj7JN17t-@mail.gmail.com>

Hi Guys,

I've successfully made & installed mod_h323 on CentOS 5.5 x86_64 bit OS. But
It can not be loaded because this :

freeswitch at internal> load mod_h323
+OK Reloading XML
-ERR [module load file routine returned an error]

2010-10-26 14:04:50.438317 [INFO] mod_enum.c:808 ENUM Reloaded
freeswitch at internal> 2010-10-26 14:04:50.438317 [INFO] switch_time.c:950
Timezone reloaded 530 definitions
2010-10-26 14:04:50.494262 [CRIT] switch_loadable_module.c:928 Error Loading
module /usr/local/freeswitch/mod/mod_h323.so
**/usr/local/freeswitch/mod/mod_h323.so: undefined symbol:
_ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infoi**

Although I've found a same thread on this list "Mod_h323 on openvz (64
bit)", it seems did not get final response. Maybe it's about loading another
share library which is not in path (LD_LIBRARY_PATH ?)

Appreciate all comments,
-- afshin
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From steveayre at gmail.com  Tue Oct 26 04:09:59 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 26 Oct 2010 12:09:59 +0100
Subject: [Freeswitch-users] Error Loading mod_h323
In-Reply-To: <AANLkTi=ezKDAZMAsJ1zYKKg24Qfc4Yjru--Uj7JN17t-@mail.gmail.com>
References: <AANLkTi=ezKDAZMAsJ1zYKKg24Qfc4Yjru--Uj7JN17t-@mail.gmail.com>
Message-ID: <AANLkTikdkZyOzuyVmbVEPfPjf9FE2Hagd3oqpSvSsQzW@mail.gmail.com>

It probably means you're using the wrong version of h323plus. The API
changes frequently, and it's trying to use a function that doesn't
seem to exist in your version. LD_LIBRARY_PATH should be ok since
it'll give a different error if it can't load the library.

See http://wiki.freeswitch.org/wiki/Mod_h323

You'll want ptlib 2.8.2:
$ svn co http://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/tags/v2_8_2/
ptlib
And h323plus 20100525 or trunk:
$ wget http://waix.dl.sourceforge.net/project/openh323gk/Sources/2.3.2/h323plus-20100525.tar.gz
$ cvs -d:pserver:anonymous at h323plus.cvs.sourceforge.net:/cvsroot/h323plus
checkout h323plus

-Steve



On 26 October 2010 11:46, afshin afzali <a.afzali2003 at gmail.com> wrote:
> Hi Guys,
>
> I've successfully made & installed mod_h323 on CentOS 5.5 x86_64 bit OS. But
> It can not be loaded because this :
>
> freeswitch at internal> load mod_h323
> +OK Reloading XML
> -ERR [module load file routine returned an error]
>
> 2010-10-26 14:04:50.438317 [INFO] mod_enum.c:808 ENUM Reloaded
> freeswitch at internal> 2010-10-26 14:04:50.438317 [INFO] switch_time.c:950
> Timezone reloaded 530 definitions
> 2010-10-26 14:04:50.494262 [CRIT] switch_loadable_module.c:928 Error Loading
> module /usr/local/freeswitch/mod/mod_h323.so
> **/usr/local/freeswitch/mod/mod_h323.so: undefined symbol:
> _ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infoi**
>
> Although I've found a same thread on this list "Mod_h323 on openvz (64
> bit)", it seems did not get final response. Maybe it's about loading another
> share library which is not in path (LD_LIBRARY_PATH ?)
>
> Appreciate all comments,
> -- afshin
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From a.afzali2003 at gmail.com  Tue Oct 26 04:17:01 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Tue, 26 Oct 2010 14:47:01 +0330
Subject: [Freeswitch-users] Error Loading mod_h323
In-Reply-To: <AANLkTikdkZyOzuyVmbVEPfPjf9FE2Hagd3oqpSvSsQzW@mail.gmail.com>
References: <AANLkTi=ezKDAZMAsJ1zYKKg24Qfc4Yjru--Uj7JN17t-@mail.gmail.com>
	<AANLkTikdkZyOzuyVmbVEPfPjf9FE2Hagd3oqpSvSsQzW@mail.gmail.com>
Message-ID: <AANLkTi=uMrS5W+4Dey=vDhgr_z3WogDx3CeCKPjhfJm8@mail.gmail.com>

Hi Steve,

Thanks to your reply,
As you said I've check out the ptlib 2.8.2 & h323plus trunk! maybe I should
use 20100525 ?

Regards,
-- afshin

On Tue, Oct 26, 2010 at 2:39 PM, Steven Ayre <steveayre at gmail.com> wrote:

> It probably means you're using the wrong version of h323plus. The API
> changes frequently, and it's trying to use a function that doesn't
> seem to exist in your version. LD_LIBRARY_PATH should be ok since
> it'll give a different error if it can't load the library.
>
> See http://wiki.freeswitch.org/wiki/Mod_h323
>
> You'll want ptlib 2.8.2:
> $ svn co
> http://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/tags/v2_8_2/
> ptlib
> And h323plus 20100525 or trunk:
> $ wget
> http://waix.dl.sourceforge.net/project/openh323gk/Sources/2.3.2/h323plus-20100525.tar.gz
> $ cvs -d:pserver:anonymous at h323plus.cvs.sourceforge.net:/cvsroot/h323plus
> checkout h323plus
>
> -Steve
>
>
>
> On 26 October 2010 11:46, afshin afzali <a.afzali2003 at gmail.com> wrote:
> > Hi Guys,
> >
> > I've successfully made & installed mod_h323 on CentOS 5.5 x86_64 bit OS.
> But
> > It can not be loaded because this :
> >
> > freeswitch at internal> load mod_h323
> > +OK Reloading XML
> > -ERR [module load file routine returned an error]
> >
> > 2010-10-26 14:04:50.438317 [INFO] mod_enum.c:808 ENUM Reloaded
> > freeswitch at internal> 2010-10-26 14:04:50.438317 [INFO] switch_time.c:950
> > Timezone reloaded 530 definitions
> > 2010-10-26 14:04:50.494262 [CRIT] switch_loadable_module.c:928 Error
> Loading
> > module /usr/local/freeswitch/mod/mod_h323.so
> > **/usr/local/freeswitch/mod/mod_h323.so: undefined symbol:
> > _ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infoi**
> >
> > Although I've found a same thread on this list "Mod_h323 on openvz (64
> > bit)", it seems did not get final response. Maybe it's about loading
> another
> > share library which is not in path (LD_LIBRARY_PATH ?)
> >
> > Appreciate all comments,
> > -- afshin
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From peter.olsson at visionutveckling.se  Tue Oct 26 04:20:09 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 26 Oct 2010 13:20:09 +0200
Subject: [Freeswitch-users] Error Loading mod_h323
In-Reply-To: <AANLkTi=ezKDAZMAsJ1zYKKg24Qfc4Yjru--Uj7JN17t-@mail.gmail.com>
References: <AANLkTi=ezKDAZMAsJ1zYKKg24Qfc4Yjru--Uj7JN17t-@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57E630C4CE@cooper>

Add -DP_64BIT in the Makefile for LOCAL_CFLAGS and rebuild the module.

Mod_h323's Makefile doesn't detect the platform automatically for you... I tried this myself a couple of days ago, and it seems to work.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r afshin afzali
Skickat: den 26 oktober 2010 12:47
Till: freeswitch-users
?mne: [Freeswitch-users] Error Loading mod_h323

Hi Guys,

I've successfully made & installed mod_h323 on CentOS 5.5 x86_64 bit OS. But It can not be loaded because this :

freeswitch at internal> load mod_h323
+OK Reloading XML
-ERR [module load file routine returned an error]

2010-10-26 14:04:50.438317 [INFO] mod_enum.c:808 ENUM Reloaded
freeswitch at internal> 2010-10-26 14:04:50.438317 [INFO] switch_time.c:950 Timezone reloaded 530 definitions
2010-10-26 14:04:50.494262 [CRIT] switch_loadable_module.c:928 Error Loading module /usr/local/freeswitch/mod/mod_h323.so
**/usr/local/freeswitch/mod/mod_h323.so: undefined symbol: _ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infoi**

Although I've found a same thread on this list "Mod_h323 on openvz (64 bit)", it seems did not get final response. Maybe it's about loading another share library which is not in path (LD_LIBRARY_PATH ?)

Appreciate all comments,
-- afshin
!DSPAM:4cc6b31132935800614202!
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From a.afzali2003 at gmail.com  Tue Oct 26 04:30:22 2010
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Tue, 26 Oct 2010 15:00:22 +0330
Subject: [Freeswitch-users] Error Loading mod_h323
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57E630C4CE@cooper>
References: <AANLkTi=ezKDAZMAsJ1zYKKg24Qfc4Yjru--Uj7JN17t-@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57E630C4CE@cooper>
Message-ID: <AANLkTikvKcVpQ2se3=j-sd-z8op7yWv0Ek4rRoi-s=Us@mail.gmail.com>

Thanks Peter , Thanks Steve
By adding LOCAL_CFLAGS it's done.

Yours,
-- afshin


On Tue, Oct 26, 2010 at 2:50 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  Add -DP_64BIT in the Makefile for LOCAL_CFLAGS and rebuild the module.
>
>
>
> Mod_h323?s Makefile doesn?t detect the platform automatically for you... I
> tried this myself a couple of days ago, and it seems to work.
>
>
>
> /Peter
>
>
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *afshin afzali
> *Skickat:* den 26 oktober 2010 12:47
> *Till:* freeswitch-users
> *?mne:* [Freeswitch-users] Error Loading mod_h323
>
>
>
> Hi Guys,
>
> I've successfully made & installed mod_h323 on CentOS 5.5 x86_64 bit OS.
> But It can not be loaded because this :
>
> freeswitch at internal> load mod_h323
> +OK Reloading XML
> -ERR [module load file routine returned an error]
>
> 2010-10-26 14:04:50.438317 [INFO] mod_enum.c:808 ENUM Reloaded
> freeswitch at internal> 2010-10-26 14:04:50.438317 [INFO] switch_time.c:950
> Timezone reloaded 530 definitions
> 2010-10-26 14:04:50.494262 [CRIT] switch_loadable_module.c:928 Error
> Loading module /usr/local/freeswitch/mod/mod_h323.so
> **/usr/local/freeswitch/mod/mod_h323.so: undefined symbol:
> _ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infoi**
>
> Although I've found a same thread on this list "Mod_h323 on openvz (64
> bit)", it seems did not get final response. Maybe it's about loading another
> share library which is not in path (LD_LIBRARY_PATH ?)
>
> Appreciate all comments,
> -- afshin
> !DSPAM:4cc6b31132935800614202!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Mon Oct 25 21:32:46 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 26 Oct 2010 15:32:46 +1100
Subject: [Freeswitch-users] FS not detecting public IP address change
In-Reply-To: <AANLkTimzkb7qY8nW3M2afTPn18=OxY6jQye8D9TKr9Ts@mail.gmail.com>
References: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>
	<AANLkTinEoTcLmUVcpras8-m8pdkfpE=_r0spz1S50iMF@mail.gmail.com>
	<AANLkTi=MoV27FW0Ukd6K+mqmgnkyNgb9hjFx-vBYV1ak@mail.gmail.com>
	<AANLkTimCRfKSWG+5OdQY0=HTt2kOqxwk+0vKvaBNdmAd@mail.gmail.com>
	<AANLkTikfpRi9Shxiqd2w5nz3ZHsBnrHTt+=rHHnvoDJn@mail.gmail.com>
	<AANLkTikH3JJCCpL5s_eF2qjjttnEetE0i=nPn87zhBUL@mail.gmail.com>
	<AANLkTimzkb7qY8nW3M2afTPn18=OxY6jQye8D9TKr9Ts@mail.gmail.com>
Message-ID: <AANLkTimR2x5AQPNiDDt4fZ8F=z_Wep3TBVR9Lob_72OA@mail.gmail.com>

:)  that was the first thing I did, but I get nothing; no keep alive
packets.

in the logfile at startup I see:

2010-10-26 10:19:45.484258 [INFO] switch_nat.c:410 Scanning for NAT
2010-10-26 10:19:45.485991 [DEBUG] switch_nat.c:167 Checking for PMP 1/5
2010-10-26 10:19:45.488575 [INFO] switch_nat.c:423 NAT detected type: pmp,
ExtIP: '203.xxx.xxx.xxx'
2010-10-26 10:19:45.490683 [DEBUG] switch_nat.c:256 NAT thread configured
2010-10-26 10:19:45.492180 [DEBUG] switch_nat.c:267 NAT thread started

And nat_map status shows me which ports have been forwarded and the external
ip address (which is sometimes wrong).  I also see this in the router gui.

But I never see the keep alive packets.

On Tue, Oct 26, 2010 at 3:13 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> Just turn up console logging to level 7.
>
> /log 7 if using fs_cli
>
>
> On Mon, Oct 25, 2010 at 6:53 PM, Mark Campbell-Smith <
> mcampbellsmith at gmail.com> wrote:
>
>> Hi!
>>
>> How do I enable debugging to get the UPnP keep alive packets?
>>
>> Thanks!
>>
>> On Tue, Oct 26, 2010 at 1:15 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>>> I need the debug logging from the upnp stuff.  It should look something
>>> like:
>>>
>>> +OK log level 7 [7]
>>> freeswitch at internal> 2010-10-25 09:13:59.926861 [DEBUG] switch_nat.c:299
>>> got UPnP keep alive packet:
>>> NOTIFY * HTTP/1.1
>>> HOST:239.255.255.250:1900
>>> Cache-Control:max-age=60
>>> Location:http://192.168.1.1:5000/rootDesc.xml
>>> Server: Tomato UPnP/1.0 MiniUPnPd/1.4
>>> NT:urn:schemas-upnp-org:service:WANIPConnection:1
>>>
>>> USN:uuid:00000000-0000-0000-0000-000000000000::urn:schemas-upnp-org:service:WANIPConnection:1
>>> NTS:ssdp:alive
>>>
>>> I also run tomato and haven't had trouble with the upnp support so at
>>> least we have that part working.
>>>
>>> You should see the above keepalive appear periodically every 30s or so.
>>>  You should see another set of messages when you terminate the DSL
>>> connection and when that dsl connection comes back online.
>>>
>>> On Sat, Oct 23, 2010 at 5:36 PM, Mark Campbell-Smith <
>>> mcampbellsmith at gmail.com> wrote:
>>>
>>>> Hi!
>>>>
>>>> The router is a ASUS router but it is running the tomato firmware which
>>>> runs miniupnd.
>>>>
>>>> In the debug I see the message: 2010-10-24 08:02:59.178918 [DEBUG]
>>>> sofia.c:956 nua_i_outbound: unknown event 8: 101 NAT detected
>>>>
>>>> This is what I see below. I hope I enabled all debug messages.  I
>>>> thought I saw some xml type messages earlier, but not sure how I enabled
>>>> them.
>>>>
>>>> nta_outgoing: RTT is 67.958 ms
>>>> outbound(0xb6e07c00): NAT binding changed: [210.xx.xxx.xx]:5080 !=
>>>> [58.xxx.xx.xx]:5080
>>>> nua(0xb6e07c00): event i_outbound 102 NAT binding changed
>>>> nua: nua_application_event: entering
>>>> 2010-10-24 09:05:08.849525 [DEBUG] sofia.c:956 nua_i_outbound: unknown
>>>> event 8: 102 NAT binding changed
>>>>
>>>>
>>>> On Sun, Oct 24, 2010 at 1:08 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>>
>>>>> btw: what router are you using?
>>>>>
>>>>>
>>>>> On Sat, Oct 23, 2010 at 9:06 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>>>
>>>>>> In a upnp config, FS is depending on the router to notify it of the ip
>>>>>> address change.  It is possible the upnp message(s) aren't formatted as
>>>>>> expected.  If you turn on debug logging you'll get the upnp messages on the
>>>>>> console.  If you can pastebin the capture of those while dropping and
>>>>>> restarting the dsl connection I can maybe see if there is something obvious
>>>>>> going on.  The debug should also have messages related to the processing of
>>>>>> those upnp messages.
>>>>>>
>>>>>> On Sat, Oct 23, 2010 at 7:04 AM, Mark Campbell-Smith <
>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>
>>>>>>> HI!
>>>>>>>
>>>>>>> I know this question must have been answered 100's of times....
>>>>>>>
>>>>>>> My adsl is a bit dodgy at the moment and tends to go up and down;
>>>>>>> which means I get assigned a new ip address from my ISP.  FS is nat'd behind
>>>>>>> a upnp capable router.
>>>>>>>
>>>>>>> The problem is that FS does not seem to be detecting the change.
>>>>>>> For both the internal and external profile, I have auto-nat setup, and I
>>>>>>> have tried stun and host settings in vars.conf.  When I issue a nat_map
>>>>>>> status I see my old IP address, and if I then issue a nat_map reinit, I see
>>>>>>> the new public IP address.  How can I get this to be automatic?
>>>>>>>
>>>>>>> Also the public IP address shown in the nat_map status is not the
>>>>>>> same as shown in the sofia profile internal/external printouts.  Why is
>>>>>>> this?
>>>>>>>
>>>>>>> I'm sure this is configuration, just not sure what to change.
>>>>>>>
>>>>>>> Thanks!
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> -Rupa
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> -Rupa
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Tue Oct 26 04:24:20 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Tue, 26 Oct 2010 22:24:20 +1100
Subject: [Freeswitch-users] FS not detecting public IP address change
In-Reply-To: <AANLkTimR2x5AQPNiDDt4fZ8F=z_Wep3TBVR9Lob_72OA@mail.gmail.com>
References: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>
	<AANLkTinEoTcLmUVcpras8-m8pdkfpE=_r0spz1S50iMF@mail.gmail.com>
	<AANLkTi=MoV27FW0Ukd6K+mqmgnkyNgb9hjFx-vBYV1ak@mail.gmail.com>
	<AANLkTimCRfKSWG+5OdQY0=HTt2kOqxwk+0vKvaBNdmAd@mail.gmail.com>
	<AANLkTikfpRi9Shxiqd2w5nz3ZHsBnrHTt+=rHHnvoDJn@mail.gmail.com>
	<AANLkTikH3JJCCpL5s_eF2qjjttnEetE0i=nPn87zhBUL@mail.gmail.com>
	<AANLkTimzkb7qY8nW3M2afTPn18=OxY6jQye8D9TKr9Ts@mail.gmail.com>
	<AANLkTimR2x5AQPNiDDt4fZ8F=z_Wep3TBVR9Lob_72OA@mail.gmail.com>
Message-ID: <AANLkTi=JgCME8u=27kUyLGU3cMP2nZpFpu7Zd5-4tpj0@mail.gmail.com>

OK..  tcpdump gave me this and I still do not see anything on FS except for
this line:

2010-10-26 22:02:51.235608 [DEBUG] sofia.c:957 nua_i_outbound: unknown event
8: 102 NAT binding changed

Hypertext Transfer Protocol
    NOTIFY * HTTP/1.1\r\n
    HOST:239.255.255.250:1900\r\n
    Cache-Control:max-age=120\r\n
    Location:http://192.168.1.1:1278/rootDesc.xml\r\n
    Server: Tomato UPnP/1.0 MiniUPnPd/1.4\r\n
    NT:urn:schemas-upnp-org:service:WANIPConnection:1\r\n

 USN:uuid:882dbe81-c28a-4992-baf7-410c05f1caf4::urn:schemas-upnp-org:service:WANIPConnection:1\r\n
    NTS:ssdp:alive\r\n
    \r\n

This looks the same as you posted before, so why do I get the unknown event
8 (is this related)?

Thanks!
On Tue, Oct 26, 2010 at 3:32 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> :)  that was the first thing I did, but I get nothing; no keep alive
> packets.
>
> in the logfile at startup I see:
>
> 2010-10-26 10:19:45.484258 [INFO] switch_nat.c:410 Scanning for NAT
> 2010-10-26 10:19:45.485991 [DEBUG] switch_nat.c:167 Checking for PMP 1/5
> 2010-10-26 10:19:45.488575 [INFO] switch_nat.c:423 NAT detected type: pmp,
> ExtIP: '203.xxx.xxx.xxx'
> 2010-10-26 10:19:45.490683 [DEBUG] switch_nat.c:256 NAT thread configured
> 2010-10-26 10:19:45.492180 [DEBUG] switch_nat.c:267 NAT thread started
>
> And nat_map status shows me which ports have been forwarded and the
> external ip address (which is sometimes wrong).  I also see this in the
> router gui.
>
> But I never see the keep alive packets.
>
> On Tue, Oct 26, 2010 at 3:13 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> Just turn up console logging to level 7.
>>
>> /log 7 if using fs_cli
>>
>>
>> On Mon, Oct 25, 2010 at 6:53 PM, Mark Campbell-Smith <
>> mcampbellsmith at gmail.com> wrote:
>>
>>> Hi!
>>>
>>> How do I enable debugging to get the UPnP keep alive packets?
>>>
>>> Thanks!
>>>
>>> On Tue, Oct 26, 2010 at 1:15 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>>> I need the debug logging from the upnp stuff.  It should look something
>>>> like:
>>>>
>>>> +OK log level 7 [7]
>>>> freeswitch at internal> 2010-10-25 09:13:59.926861 [DEBUG]
>>>> switch_nat.c:299 got UPnP keep alive packet:
>>>> NOTIFY * HTTP/1.1
>>>> HOST:239.255.255.250:1900
>>>> Cache-Control:max-age=60
>>>> Location:http://192.168.1.1:5000/rootDesc.xml
>>>> Server: Tomato UPnP/1.0 MiniUPnPd/1.4
>>>> NT:urn:schemas-upnp-org:service:WANIPConnection:1
>>>>
>>>> USN:uuid:00000000-0000-0000-0000-000000000000::urn:schemas-upnp-org:service:WANIPConnection:1
>>>> NTS:ssdp:alive
>>>>
>>>> I also run tomato and haven't had trouble with the upnp support so at
>>>> least we have that part working.
>>>>
>>>> You should see the above keepalive appear periodically every 30s or so.
>>>>  You should see another set of messages when you terminate the DSL
>>>> connection and when that dsl connection comes back online.
>>>>
>>>> On Sat, Oct 23, 2010 at 5:36 PM, Mark Campbell-Smith <
>>>> mcampbellsmith at gmail.com> wrote:
>>>>
>>>>> Hi!
>>>>>
>>>>> The router is a ASUS router but it is running the tomato firmware which
>>>>> runs miniupnd.
>>>>>
>>>>> In the debug I see the message: 2010-10-24 08:02:59.178918 [DEBUG]
>>>>> sofia.c:956 nua_i_outbound: unknown event 8: 101 NAT detected
>>>>>
>>>>> This is what I see below. I hope I enabled all debug messages.  I
>>>>> thought I saw some xml type messages earlier, but not sure how I enabled
>>>>> them.
>>>>>
>>>>> nta_outgoing: RTT is 67.958 ms
>>>>> outbound(0xb6e07c00): NAT binding changed: [210.xx.xxx.xx]:5080 !=
>>>>> [58.xxx.xx.xx]:5080
>>>>> nua(0xb6e07c00): event i_outbound 102 NAT binding changed
>>>>> nua: nua_application_event: entering
>>>>> 2010-10-24 09:05:08.849525 [DEBUG] sofia.c:956 nua_i_outbound: unknown
>>>>> event 8: 102 NAT binding changed
>>>>>
>>>>>
>>>>> On Sun, Oct 24, 2010 at 1:08 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>>>
>>>>>> btw: what router are you using?
>>>>>>
>>>>>>
>>>>>> On Sat, Oct 23, 2010 at 9:06 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>
>>>>>>> In a upnp config, FS is depending on the router to notify it of the
>>>>>>> ip address change.  It is possible the upnp message(s) aren't formatted as
>>>>>>> expected.  If you turn on debug logging you'll get the upnp messages on the
>>>>>>> console.  If you can pastebin the capture of those while dropping and
>>>>>>> restarting the dsl connection I can maybe see if there is something obvious
>>>>>>> going on.  The debug should also have messages related to the processing of
>>>>>>> those upnp messages.
>>>>>>>
>>>>>>> On Sat, Oct 23, 2010 at 7:04 AM, Mark Campbell-Smith <
>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>
>>>>>>>> HI!
>>>>>>>>
>>>>>>>> I know this question must have been answered 100's of times....
>>>>>>>>
>>>>>>>> My adsl is a bit dodgy at the moment and tends to go up and down;
>>>>>>>> which means I get assigned a new ip address from my ISP.  FS is nat'd behind
>>>>>>>> a upnp capable router.
>>>>>>>>
>>>>>>>> The problem is that FS does not seem to be detecting the change.
>>>>>>>> For both the internal and external profile, I have auto-nat setup, and I
>>>>>>>> have tried stun and host settings in vars.conf.  When I issue a nat_map
>>>>>>>> status I see my old IP address, and if I then issue a nat_map reinit, I see
>>>>>>>> the new public IP address.  How can I get this to be automatic?
>>>>>>>>
>>>>>>>> Also the public IP address shown in the nat_map status is not the
>>>>>>>> same as shown in the sofia profile internal/external printouts.  Why is
>>>>>>>> this?
>>>>>>>>
>>>>>>>> I'm sure this is configuration, just not sure what to change.
>>>>>>>>
>>>>>>>> Thanks!
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> -Rupa
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> -Rupa
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From abubacker at bksystems.co.in  Tue Oct 26 06:29:56 2010
From: abubacker at bksystems.co.in (abubacker)
Date: Tue, 26 Oct 2010 18:59:56 +0530
Subject: [Freeswitch-users] nowait.G7221 error
Message-ID: <4CC6D7D4.7080507@bksys.co.in>

Dear all,
What this error denotes and how to resolve this , I always got this 
error when I am doing
queue manipulation using mod_fifo

[ERR] mod_native_file.c:74 Error opening 
/usr/local/freeswitch/sounds/en/us/callie/nowait.G7221

Thanks in advance !

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

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



From rupa at rupa.com  Tue Oct 26 06:43:38 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 26 Oct 2010 08:43:38 -0500
Subject: [Freeswitch-users] FS not detecting public IP address change
In-Reply-To: <AANLkTi=JgCME8u=27kUyLGU3cMP2nZpFpu7Zd5-4tpj0@mail.gmail.com>
References: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>
	<AANLkTinEoTcLmUVcpras8-m8pdkfpE=_r0spz1S50iMF@mail.gmail.com>
	<AANLkTi=MoV27FW0Ukd6K+mqmgnkyNgb9hjFx-vBYV1ak@mail.gmail.com>
	<AANLkTimCRfKSWG+5OdQY0=HTt2kOqxwk+0vKvaBNdmAd@mail.gmail.com>
	<AANLkTikfpRi9Shxiqd2w5nz3ZHsBnrHTt+=rHHnvoDJn@mail.gmail.com>
	<AANLkTikH3JJCCpL5s_eF2qjjttnEetE0i=nPn87zhBUL@mail.gmail.com>
	<AANLkTimzkb7qY8nW3M2afTPn18=OxY6jQye8D9TKr9Ts@mail.gmail.com>
	<AANLkTimR2x5AQPNiDDt4fZ8F=z_Wep3TBVR9Lob_72OA@mail.gmail.com>
	<AANLkTi=JgCME8u=27kUyLGU3cMP2nZpFpu7Zd5-4tpj0@mail.gmail.com>
Message-ID: <AANLkTikL=kvbfLQx3E-vCaErRM-ARnU3_YSAvKo67Uvy@mail.gmail.com>

What do you have set for loglevel in switch.conf.xml?  My guess is that you
have it set to something other than debug.  If that is set to (say) info,
then info is as low as you can go.  Setting the console log level to debug
will not give you debug output.

sofia is unaware of upnp so a keepalive packet should have no impact on it's
operation.  I'm not familiar enough with sofia to tell what triggers that
event.

On Tue, Oct 26, 2010 at 6:24 AM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> OK..  tcpdump gave me this and I still do not see anything on FS except for
> this line:
>
> 2010-10-26 22:02:51.235608 [DEBUG] sofia.c:957 nua_i_outbound: unknown
> event 8: 102 NAT binding changed
>
> Hypertext Transfer Protocol
>     NOTIFY * HTTP/1.1\r\n
>     HOST:239.255.255.250:1900\r\n
>     Cache-Control:max-age=120\r\n
>     Location:http://192.168.1.1:1278/rootDesc.xml\r\n<http://192.168.1.1:1278/rootDesc.xml%5Cr%5Cn>
>     Server: Tomato UPnP/1.0 MiniUPnPd/1.4\r\n
>     NT:urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>
>  USN:uuid:882dbe81-c28a-4992-baf7-410c05f1caf4::urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>     NTS:ssdp:alive\r\n
>     \r\n
>
> This looks the same as you posted before, so why do I get the unknown event
> 8 (is this related)?
>
> Thanks!
>
> On Tue, Oct 26, 2010 at 3:32 PM, Mark Campbell-Smith <
> mcampbellsmith at gmail.com> wrote:
>
>> :)  that was the first thing I did, but I get nothing; no keep alive
>> packets.
>>
>> in the logfile at startup I see:
>>
>> 2010-10-26 10:19:45.484258 [INFO] switch_nat.c:410 Scanning for NAT
>> 2010-10-26 10:19:45.485991 [DEBUG] switch_nat.c:167 Checking for PMP 1/5
>> 2010-10-26 10:19:45.488575 [INFO] switch_nat.c:423 NAT detected type: pmp,
>> ExtIP: '203.xxx.xxx.xxx'
>> 2010-10-26 10:19:45.490683 [DEBUG] switch_nat.c:256 NAT thread configured
>> 2010-10-26 10:19:45.492180 [DEBUG] switch_nat.c:267 NAT thread started
>>
>> And nat_map status shows me which ports have been forwarded and the
>> external ip address (which is sometimes wrong).  I also see this in the
>> router gui.
>>
>> But I never see the keep alive packets.
>>
>> On Tue, Oct 26, 2010 at 3:13 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>>> Just turn up console logging to level 7.
>>>
>>> /log 7 if using fs_cli
>>>
>>>
>>> On Mon, Oct 25, 2010 at 6:53 PM, Mark Campbell-Smith <
>>> mcampbellsmith at gmail.com> wrote:
>>>
>>>> Hi!
>>>>
>>>> How do I enable debugging to get the UPnP keep alive packets?
>>>>
>>>> Thanks!
>>>>
>>>> On Tue, Oct 26, 2010 at 1:15 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>>
>>>>> I need the debug logging from the upnp stuff.  It should look something
>>>>> like:
>>>>>
>>>>> +OK log level 7 [7]
>>>>> freeswitch at internal> 2010-10-25 09:13:59.926861 [DEBUG]
>>>>> switch_nat.c:299 got UPnP keep alive packet:
>>>>> NOTIFY * HTTP/1.1
>>>>> HOST:239.255.255.250:1900
>>>>> Cache-Control:max-age=60
>>>>> Location:http://192.168.1.1:5000/rootDesc.xml
>>>>> Server: Tomato UPnP/1.0 MiniUPnPd/1.4
>>>>> NT:urn:schemas-upnp-org:service:WANIPConnection:1
>>>>>
>>>>> USN:uuid:00000000-0000-0000-0000-000000000000::urn:schemas-upnp-org:service:WANIPConnection:1
>>>>> NTS:ssdp:alive
>>>>>
>>>>> I also run tomato and haven't had trouble with the upnp support so at
>>>>> least we have that part working.
>>>>>
>>>>> You should see the above keepalive appear periodically every 30s or so.
>>>>>  You should see another set of messages when you terminate the DSL
>>>>> connection and when that dsl connection comes back online.
>>>>>
>>>>> On Sat, Oct 23, 2010 at 5:36 PM, Mark Campbell-Smith <
>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>
>>>>>> Hi!
>>>>>>
>>>>>> The router is a ASUS router but it is running the tomato firmware
>>>>>> which runs miniupnd.
>>>>>>
>>>>>> In the debug I see the message: 2010-10-24 08:02:59.178918 [DEBUG]
>>>>>> sofia.c:956 nua_i_outbound: unknown event 8: 101 NAT detected
>>>>>>
>>>>>> This is what I see below. I hope I enabled all debug messages.  I
>>>>>> thought I saw some xml type messages earlier, but not sure how I enabled
>>>>>> them.
>>>>>>
>>>>>> nta_outgoing: RTT is 67.958 ms
>>>>>> outbound(0xb6e07c00): NAT binding changed: [210.xx.xxx.xx]:5080 !=
>>>>>> [58.xxx.xx.xx]:5080
>>>>>> nua(0xb6e07c00): event i_outbound 102 NAT binding changed
>>>>>> nua: nua_application_event: entering
>>>>>> 2010-10-24 09:05:08.849525 [DEBUG] sofia.c:956 nua_i_outbound: unknown
>>>>>> event 8: 102 NAT binding changed
>>>>>>
>>>>>>
>>>>>> On Sun, Oct 24, 2010 at 1:08 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>
>>>>>>> btw: what router are you using?
>>>>>>>
>>>>>>>
>>>>>>> On Sat, Oct 23, 2010 at 9:06 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>
>>>>>>>> In a upnp config, FS is depending on the router to notify it of the
>>>>>>>> ip address change.  It is possible the upnp message(s) aren't formatted as
>>>>>>>> expected.  If you turn on debug logging you'll get the upnp messages on the
>>>>>>>> console.  If you can pastebin the capture of those while dropping and
>>>>>>>> restarting the dsl connection I can maybe see if there is something obvious
>>>>>>>> going on.  The debug should also have messages related to the processing of
>>>>>>>> those upnp messages.
>>>>>>>>
>>>>>>>> On Sat, Oct 23, 2010 at 7:04 AM, Mark Campbell-Smith <
>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> HI!
>>>>>>>>>
>>>>>>>>> I know this question must have been answered 100's of times....
>>>>>>>>>
>>>>>>>>> My adsl is a bit dodgy at the moment and tends to go up and down;
>>>>>>>>> which means I get assigned a new ip address from my ISP.  FS is nat'd behind
>>>>>>>>> a upnp capable router.
>>>>>>>>>
>>>>>>>>> The problem is that FS does not seem to be detecting the change.
>>>>>>>>> For both the internal and external profile, I have auto-nat setup, and I
>>>>>>>>> have tried stun and host settings in vars.conf.  When I issue a nat_map
>>>>>>>>> status I see my old IP address, and if I then issue a nat_map reinit, I see
>>>>>>>>> the new public IP address.  How can I get this to be automatic?
>>>>>>>>>
>>>>>>>>> Also the public IP address shown in the nat_map status is not the
>>>>>>>>> same as shown in the sofia profile internal/external printouts.  Why is
>>>>>>>>> this?
>>>>>>>>>
>>>>>>>>> I'm sure this is configuration, just not sure what to change.
>>>>>>>>>
>>>>>>>>> Thanks!
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> -Rupa
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> -Rupa
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> -Rupa
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From steveayre at gmail.com  Tue Oct 26 07:12:03 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 26 Oct 2010 15:12:03 +0100
Subject: [Freeswitch-users] nowait.G7221 error
In-Reply-To: <4CC6D7D4.7080507@bksys.co.in>
References: <4CC6D7D4.7080507@bksys.co.in>
Message-ID: <AANLkTimarRSWW7Ym8euCVANuKE3r0fzLehSoqpewNW1P@mail.gmail.com>

The file does not exist.

mod_native_file It is trying to play a file named nowait that has
already been encoded with the codec you're using on the call (G7221 in
this case). The file doesn't exist though.

Create it, or adjust your dialplan if you don't require it.

-Steve


On 26 October 2010 14:29, abubacker <abubacker at bksystems.co.in> wrote:
> Dear all,
> What this error denotes and how to resolve this , I always got this
> error when I am doing
> queue manipulation using mod_fifo
>
> [ERR] mod_native_file.c:74 Error opening
> /usr/local/freeswitch/sounds/en/us/callie/nowait.G7221
>
> Thanks in advance !
>
> --
> Best regards,
> N.Abubacker ,
> Associate system engineer ,
> bk systems pvt ltd ,
> Ph : 9144-43902701
>
> Disclaimer: http://www.bksystems.co.in/email-policy
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From anthony.minessale at gmail.com  Tue Oct 26 07:12:52 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 Oct 2010 09:12:52 -0500
Subject: [Freeswitch-users] Sofia_reg.c:816 Can not do authorization
 without a complete from header
In-Reply-To: <81C2CEF80046FB4F863A60D4347DD33A0C566A@server1.st.local>
References: <81C2CEF80046FB4F863A60D4347DD33A0C566A@server1.st.local>
Message-ID: <AANLkTi=Nb6TSfsZKjGx9cJNPM7HU=aPD4cHAhKALw_pM@mail.gmail.com>

use the source luke.

if (!to_user || !to_host) {
        switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Can
not do authorization without a complete header in REGISTER request
from %s:%d\n",
                          network_ip, network_port);


2010/10/26 Erkan ?nl? <erkan at speedingtrade.com>:
> Hi all,
>
>
>
> Today I have many error in my FS.
>
> Sofia_reg.c:816? Can not do authorization without a complete? from header
>
>
>
> What does it mean?
>
> Is there a problem in my FS or by the client?
>
> The client work since 3 months without errors, until tomorrow morning the
> many sofia_reg 816 errors is during up.
>
> The wiki doesn?t give any information?s about this error.
>
>
>
> Thank you for some information?s.
>
>
>
>
>
> Many error during up
>
> 2010-10-26 11:01:58.448513 [ERR] sofia_reg.c:816 Can not do authorization
> without a complete from header
>
> 2010-10-26 11:01:59.168193 [ERR] sofia_reg.c:816 Can not do authorization
> without a complete from header
>
> 2010-10-26 11:01:59.320527 [NOTICE] sofia.c:3739 Channel
> [sofia/external/0xxxxx42336] has been answe
>
> 2010-10-26 11:01:59.379117 [NOTICE] switch_ivr_bridge.c:306 Channel
> [sofia/internal/99014 at xxxx
>
> 2010-10-26 11:01:59.470908 [ERR] sofia_reg.c:816 Can not do authorization
> without a complete from header
>
> 2010-10-26 11:01:59.992359 [ERR] sofia_reg.c:816 Can not do authorization
> without a complete from header
>
> 2010-10-26 11:02:00.293121 [ERR] sofia_reg.c:816 Can not do authorization
> without a complete from header
>
> 2010-10-26 11:02:00.394677 [ERR] sofia_reg.c:816 Can not do authorization
> without a complete from header
>
> 2010-10-26 11:02:00.595836 [ERR] sofia_reg.c:816 Can not do authorization
> without a complete from header
>
> 2010-10-26 11:02:01.311611 [ERR] sofia_reg.c:816 Can not do authorization
> without a complete from header
>
> 2010-10-26 11:02:02.30315 [ERR] sofia_reg.c:816 Can not do authorization
> without a complete from header
>
> 2010-10-26 11:02:02.655275 [ERR] sofia_reg.c:816 Can not do authorization
> without a complete from header
>
> 2010-10-26 11:02:03.820239 [ERR] sofia_reg.c:816 Can not do authorization
> without a complete from header
>
> 2010-10-26 11:02:03.989174 [ERR] sofia_reg.c:816 Can not do authorization
> without a complete from header
>
> 2010-10-26 11:02:04.784045 [NOTICE] sofia.c:3849 Hangup
> sofia/external/0xxx463995 [CS_EXCHANGE_MED
>
> 2010-10-26 11:02:04.799669 [NOTICE] switch_ivr_bridge.c:419 Hangup
> sofia/internal/99015 at xxxxxx [
>
> 2010-10-26 11:02:04.799669 [NOTICE] switch_core_session.c:1086 Session 1517
> (sofia/external/0xxxx3046
>
> 2010-10-26 11:02:04.799669 [NOTICE] switch_core_session.c:1088 Close Channel
> sofia/external/0xxx046
>
> 2010-10-26 11:02:04.803575 [NOTICE] switch_core_session.c:1086 Session 1516
> (sofia/internal/99015 at xxxx
>
> 2010-10-26 11:02:04.803575 [NOTICE] switch_core_session.c:1088 Close Channel
> sofia/internal/99015 at xxxx
>
> 2010-10-26 11:02:05.160974 [ERR] sofia_reg.c:816 Can not do authorization
> without a complete from header
>
> 2010-10-26 11:02:05.626764 [ERR] sofia_reg.c:816 Can not do authorization
> without a complete from header
>
> 2010-10-26 11:02:05.746874 [NOTICE] sofia.c:3849 Hangup
> sofia/internal/24000 at xxxxxx [CS_EXECUTE]
>
> 2010-10-26 11:02:05.760545 [NOTICE] switch_ivr_bridge.c:503 Hangup
> sofia/external/0xxxxx38 [CS
>
> 2010-10-26 11:02:05.760545 [NOTICE] switch_core_session.c:1086 Session 1513
> (sofia/external/0xxx5203
>
> 2010-10-26 11:02:05.760545 [NOTICE] switch_core_session.c:1088 Close Channel
> sofia/external/xxxx5203
>
> 2010-10-26 11:02:05.764451 [NOTICE] switch_core_session.c:1086 Session 1512
> (sofia/internal/24000 at xxxx
>
> 2010-10-26 11:02:05.764451 [NOTICE] switch_core_session.c:1088 Close Channel
> sofia/internal/24000 at xxx
>
> 2010-10-26 11:02:05.932409 [ERR] sofia_reg.c:816 Can not do authorization
> without a complete from header
>
> 2010-10-26 11:02:06.132591 [ERR] sofia_reg.c:816 Can not do authorization
> without a complete from header
>
> 2010-10-26 11:02:06.382575 [NOTICE] sofia.c:322 Hangup
> sofia/internal/24001 at xxxxxx [CS_EXECUTE]
>
> 2010-10-26 11:02:06.400152 [NOTICE] switch_ivr_bridge.c:503 Hangup
> sofia/external/00xxxxxxxxxx5671 [CS_
>
> 2010-10-26 11:02:06.458742 [NOTICE] switch_core_session.c:1086 Session 1379
> (sofia/external/xxx37
>
> 2010-10-26 11:02:06.458742 [NOTICE] switch_core_session.c:1088 Close Channel
> sofia/external/0xxxx137
>
> 2010-10-26 11:02:06.460695 [NOTICE] switch_core_session.c:1086 Session 1378
> (sofia/internal/24001 at xxxx
>
> 2010-10-26 11:02:06.460695 [NOTICE] switch_core_session.c:1088 Close Channel
> sofia/internal/24001 at xxxx
>
> 2010-10-26 11:02:06.585687 [ERR] sofia_reg.c:816 Can not do authorization
> without a complete from header
>
> 2010-10-26 11:02:06.849342 [ERR] sofia_reg.c:816 Can not do authorization
> without a complete from header
>
> 2010-10-26 11:02:07.200882 [ERR] sofia_reg.c:816 Can not do authorization
> without a complete from header
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From anthony.minessale at gmail.com  Tue Oct 26 07:26:52 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 Oct 2010 09:26:52 -0500
Subject: [Freeswitch-users] Is anyone familiar with Telemaque?
In-Reply-To: <4CC6A214.1080601@bksys.co.in>
References: <AANLkTimib+3T7L9sP28HN6BREJF5ca2j2nou-84YU=Ln@mail.gmail.com>
	<4CAAE5DE.6080407@telemaque.fr> <4CABF9D5.7000501@bksys.co.in>
	<4CAC4EB4.6080901@telemaque.fr> <4CC6A214.1080601@bksys.co.in>
Message-ID: <AANLkTink3nf0K413UuhKahWeeQtL0aL2UuepiX+NzdFm@mail.gmail.com>

Cool Let's have a presentation at ClueCon MMXI


On Tue, Oct 26, 2010 at 4:40 AM, abubacker <abubacker at bksystems.co.in> wrote:
> On Wednesday 06 October 2010 03:55 PM, Tristan Mah? wrote:
>
> Hi,
>
> The telco cluster is powered by IBM servers ( from dual Xeon/8gb ram to dual
> quad core/16gb ram depending on the node ) with FC13 and latest FS git (
> well almost always, as of today I'm a week late, but will upgrade soon again
> to benefit of the amazing work the dev do ).
>
> If you have more precise questions, don't hesitate, I'm sorry if I can't
> reveal all the details on how we're doing things,but I'm sure you can
> understand that...
>
> Regards,
>
> Tristan.
> Le 10/06/2010 06:23 AM, abubacker a ?crit?:
>
> On Tuesday 05 October 2010 02:16 PM, Tristan Mah? wrote:
>
> There's someone working for them on list :)
>
> Feel free to ask me questions Michael.
>
> Regards,
>
> Gled.
>
> Le 10/04/2010 08:39 PM, Michael Collins a ?crit?:
>
> Dear FreeSWITCH Community,
>
> Some of you may have stumbled upon this news item:
>
> http://blogs.oracle.com/mysql/2010/10/innovating_with_open_source_call_center_services.html
>
> Evidently there is a company in France (Telemaque) who is using MySQL +
> Kamailio + FreeSWITCH for some heavy duty call processing. If you are at all
> familiar with this company please let me know. I'd like to learn more about
> what they are doing. (Je ne parle pas boucoup de Francais :( )
>
> Thanks for your help,
> MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> "Feel free to ask me questions Michael"
>
> I think I must use this opportunity ,
> please specify the software ( OS ) and the hardwares required to handle the
> heavy duty call
> processing and also specify the FreeSWITCH version.
> I guess you could answer this very precisely.
>
> Thanks in Advance !
>
> --
> Best regards,
> N.Abubacker ,
> Associate system engineer ,
> bk systems pvt ltd ,
> Ph : 9144-43902701
>
> Disclaimer: http://www.bksystems.co.in/email-policy
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> Dear Tristan ,
> Knew this is an old post , but even though I wanted this to continue to get
> some more details ,
> what mod you are using? to handle more calls ? ( hope its mod_fifo )
> I am working in mod_fifo , some times it throws socketerror and makes the
> entire freeswitch hang
> when more than 25 calls in a queue.
>
> My question is can u give me some details about the queue and member
> configuration ( if possible )
>
> If you are not using mod_fifo , just tell me the general configuration
> required to fine tune the
> freeswitch.
>
> Thanks Again !
>
> --
> Best regards,
> N.Abubacker ,
> Associate system engineer ,
> bk systems pvt ltd ,
> Ph : 9144-43902701
>
> Disclaimer: http://www.bksystems.co.in/email-policy
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From oscav at hotmail.fr  Tue Oct 26 08:19:07 2010
From: oscav at hotmail.fr (Oscav)
Date: Tue, 26 Oct 2010 08:19:07 -0700 (PDT)
Subject: [Freeswitch-users]  How to know if bridge was terminated by key?
Message-ID: <30058660.post@talk.nabble.com>


Hi,

Does anyone know if it's possible to know if a bridge was interrupted by the
key specified in "bridge_terminate_key" while also using
"hangup_after_bridge=false" ?? This is useful to trigger some specific
actions different from a bridge gone in timeout.

Thanks
-- 
View this message in context: http://old.nabble.com/How-to-know-if-bridge-was-terminated-by-key--tp30058660p30058660.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From david.villasmil.work at gmail.com  Tue Oct 26 08:34:29 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 26 Oct 2010 17:34:29 +0200
Subject: [Freeswitch-users] xml_mod_cdr
Message-ID: <AANLkTi=NQ9HC98Q3QPDOkfvx45RiFWSmHuq7VWvnaKfZ@mail.gmail.com>

Hello Guys,

I'm trying to configure mod_xml_cdr and i keep on getting this error on startup:

2010-10-26 17:06:06.364270 [CRIT] switch_loadable_module.c:882 Error
Loading module /usr/local/freeswitch/mod/mod_xml_cdr.so
**/usr/local/freeswitch/mod/mod_xml_cdr.so: undefined symbol:
switch_channel_get_variable**

my xml config is the following:

<configuration name="xml_cdr.conf" description="XML CDR CURL logger">
  <settings>
    <param name="url" value="http://server_ip/cdrpost/post.php"/>
    <param name="retries" value="2"/>
    <param name="log-dir" value="/var/log/freeswitch"/>
    <param name="log-b-leg" value="true"/>
    <param name="prefix-a-leg" value="true"/>
    <param name="encode" value="true"/>
    <param name="err-log-dir" value="/var/log/freeswitch/errors"/>
  </settings>
</configuration>

My FreeSWITCH version is:

2010-10-26 17:25:38.227987 [CONSOLE] switch_core.c:1605
FreeSWITCH Version 1.0.head (git-) Started.


Can anyone help me out?


Thanks!


David


From balabaev.m at gmail.com  Tue Oct 26 02:27:52 2010
From: balabaev.m at gmail.com (Maxim Balabaev)
Date: Tue, 26 Oct 2010 13:27:52 +0400
Subject: [Freeswitch-users] tcmalloc
Message-ID: <AANLkTik+Oj3xD9DK2G=koDsmVarmgCdqPGjQ900WQ7NL@mail.gmail.com>

Why not include configure options for tcmalloc (google perftools)?
Such as this patch -
http://oss.axsentis.de/people/stkn/freeswitch/freeswitch-r7716-alternative-memory-allocators.patch
I compiled with LIBS='-ltcmalloc' and works just fine. Maybe 0,1% vore
memory used.
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From renjian at gmail.com  Tue Oct 26 07:44:27 2010
From: renjian at gmail.com (Jian Ren)
Date: Tue, 26 Oct 2010 10:44:27 -0400
Subject: [Freeswitch-users] Help needed about building snd_dummy with
 the dummy.c in skypopen
In-Reply-To: <AANLkTi=L1ro64h=CBqQ_EUgw8exukt7yno31Fc_O2FTW@mail.gmail.com>
References: <AANLkTikQ_vj3Ue1MYs7x1pVj8kiCgFbeKBTL=97PddX=@mail.gmail.com>
	<AANLkTik5BEvEi=P_+cEL2kwjfrTQYLGh7MNtF14-8ROE@mail.gmail.com>
	<AANLkTikkXGL7H=1Sfhnzub454ob4MeCe7asoxXJi9myT@mail.gmail.com>
	<AANLkTiku3YBi004AZrC_-vW_Kc01x4NQbR3rtf_yxeNT@mail.gmail.com>
	<AANLkTik6aj0dRqL4BqkRgT9vvp+7OgQtWT=abJ0F2EZ5@mail.gmail.com>
	<AANLkTi=L1ro64h=CBqQ_EUgw8exukt7yno31Fc_O2FTW@mail.gmail.com>
Message-ID: <AANLkTim7Kfj_+755YWEPHbTRkGdP9CEjP-FRRQnSMzNL@mail.gmail.com>

Problem found and solved.
There are two dummy.c under alsa-driver-1.0.20, I overwrote the one under
drivers, which is the wrong one. Based on the wiki, it should be the one
under alsa-driver-1.0.20/sound/drivers.
Besides, I need to reboot after rebuilding the new dummy.c.

Thank Giovanni for helping me out.

Jian

On Sun, Oct 24, 2010 at 5:32 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> On Sun, Oct 24, 2010 at 11:28 PM, Jian Ren <renjian at gmail.com> wrote:
> > Ok, I found out the new error was caused by missing of 32bit
> compatibility
> > libraries, after I ran this:
> >
> > apt-get -y install ia32-libs lib32asound2 libc6-i386 lib32gcc1 \
> > lib32stdc++6 lib32ncurses5 lib32z1
>
> Btw, in the wiki is written about the 32bit libraries.
>
> If you follow the wiki, it works. I've done that many times.
>
> After you have installed ALSA, you need to reboot the machine, or to
> rmmod all the snd* modules from the kernel, before to modprobe
> snd-dummy
>
> -giovanni
>
>
> >
> > Now I am getting exactly the same problem as if under virtualBox or
> > VMPlayer. So it's not a VM problem.
> > Please help!
> > Thanks!
> > Jian
> >
> > On Sat, Oct 23, 2010 at 10:19 PM, Jian Ren <renjian at gmail.com> wrote:
> >>
> >> Hi, I tried real machine. Got different error:
> >> FATAL: Error inserting
> >>
> snd_dummy(/lib/modules/2.6.24-26-server/kernel/sound/drivers/snd-dummy.ko):
> >> Unknown symbol in module, or unknown parameter (see demsg).
> >> Then if I type demsg, got the attached screen.
> >> Please help.
> >> Thanks!
> >>
> >> On Fri, Oct 22, 2010 at 11:33 AM, Jian Ren <renjian at gmail.com> wrote:
> >>>
> >>> Tried under VMPlayer, got the same problem, will try real machine
> later.
> >>> Attached is the screen shot.
> >>>
> >>> Thanks!
> >>> Jian
> >>>
> >>> On Thu, Oct 21, 2010 at 4:45 PM, Giovanni Maruzzelli <
> gmaruzz at gmail.com>
> >>> wrote:
> >>>>
> >>>> Maybe is a problem with virtualbox. Also, is known that
> >>>> fs+mod_skypopen+skype clients does not works in virtualbox.
> >>>>
> >>>> Try it on a real (hardware) machine.
> >>>>
> >>>> Or (but is less popular) in a xen like virtual machine.
> >>>>
> >>>> -giovanni
> >>>>
> >>>> On Thu, Oct 21, 2010 at 4:21 PM, Jian Ren <renjian at gmail.com> wrote:
> >>>> > Hi,
> >>>> > I am trying to follow this wiki:
> >>>> > http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk
> >>>> > to setup FS and Skypopen on Ubuntu 8.04(64bit server inside
> >>>> > virualbox).
> >>>> >
> >>>> > There is one step asks me to build snd_dummy:
> >>>> >
> >>>> >
> http://wiki.freeswitch.org/wiki/Skypopen_Skype_Endpoint_and_Trunk#ALSA_and_the_custom_snd-dummy
> >>>> > The section is like below:
> >>>> >
> >>>> > Ubuntu, Debian
> >>>> >
> >>>> > Note, Ubuntu may have alsa sound drivers installed at
> >>>> > /lib/modules/<your
> >>>> > kernel version>/ubuntu/sound/..., you may need to remove the dir to
> >>>> > allow
> >>>> > modprobe search from the default place: /lib/modules/<your kernel
> >>>> > verision>/kernel/sound/ (don't forget to rerun /sbin/depmod after
> >>>> > removing
> >>>> > the old sound directory. Eg: /sbin/depmod -a 2.6.24-24-server).
> >>>> >
> >>>> > ./configure --with-redhat=no \
> >>>> > --with-cards=dummy,usb-audio,hda-intel,hrtimer,rtctimer \
> >>>> > --with-card-options=all
> >>>> > make && make install
> >>>> >
> >>>> > After the first make && make install, copy
> >>>> > mod_skypopen/configs/alsa/dummy.c
> >>>> > to alsa-driver-1.0.20sound/drivers/dummy.c .
> >>>> >
> >>>> > make && make install #again :)
> >>>> >
> >>>> > The problem is after I did all of them, when I entered modprobe
> >>>> > snd_dummy,
> >>>> > it always crashed and returned as "killed" on my terminal, I could
> see
> >>>> > it
> >>>> > dumped out a huge block or data(numbers, etc) on the host but don't
> >>>> > understand the meaning. While if I used the original dummy.c inside
> >>>> > alsa-driver-1.0.20, it worked fine.
> >>>> > Besides, the default ubuntu installation doesn't include kernal dev
> >>>> > and
> >>>> > source, so I did one more step(or it cannot build alsa).
> >>>> >
> >>>> > Did anyone try the same and get it working?
> >>>> >
> >>>> > Thanks!
> >>>> > Jian
> >>>> >
> >>>> >
> >>>> > _______________________________________________
> >>>> > FreeSWITCH-users mailing list
> >>>> > FreeSWITCH-users at lists.freeswitch.org
> >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> >
> >>>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> > http://www.freeswitch.org
> >>>> >
> >>>> >
> >>>>
> >>>>
> >>>>
> >>>> --
> >>>> Sincerely,
> >>>>
> >>>> Giovanni Maruzzelli
> >>>> Cell : +39-347-2665618
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
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From wstephen80 at gmail.com  Tue Oct 26 08:37:54 2010
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 26 Oct 2010 17:37:54 +0200
Subject: [Freeswitch-users] FreeTDM - Screening Indicator Field
Message-ID: <AANLkTin-5ncn2NvGJo1Qmd5EkjUqf=xAaSr1_c_Kd79J@mail.gmail.com>

Hi,
I have a server with Freeswitch (latest GIT revision) + FreeTDM with some
Sangoma A108 boards.
I have the necessity to set, for outgoing calls, the ISDN information
element "Screening Indicator" of calling party number.
I have see that some information element can be set in the
"freetdm.conf.xml" file (i.e. calling and called typer of number or
numbering plan) but I have not found any reference to the screening
indicator.
If there is no way to specify this information element for each trunk, it's
possible to change its default value (that is 01=user provided, verified and
passed)?
Thanks,

Stephen
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From mario_fs at mgtech.com  Tue Oct 26 09:09:48 2010
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 26 Oct 2010 09:09:48 -0700
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
Message-ID: <407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>

I ran the global trace during the problem and it is at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone", "accttwo", "acct3". The trace includes phones since it was global. I am using:
    <param name="ext-rtp-ip" value="autonat:my-static.ip"/>
    <param name="ext-sip-ip" value="autonat:my-static.ip"/> 

I tried dumping nat and removing the autonat: above and using -nonat but that did not work, registration proceeded but no calls inbound.

On Oct 25, 2010, at 4:11 PM, Mario G wrote:

> Whoops, I am using an IP address for at least one gateway so that is not the problem:
> They look outbound to me and I am using dns for 2 and an IP for one so that is not the issue. I was able to get FS to clear this up by doing "nat_map reinit" which is why I think this is a nat problem. I will do the trace you mentioned. I will plug an ip address into one of the gateways to see what happens, they all fail at once. Thanks for responding!
> Mario
> 
> On Oct 25, 2010, at 3:26 PM, Mario wrote:
> 
>> I really need help on this as I have weeks into this problem. I thought I had it nailed but I guess not. After 5.5 hours I get:
>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed Registration, setting retry to 15 seconds.
>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed Registration, setting retry to 30 seconds.
>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed Registration, setting retry to 15 seconds.
>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed Registration, setting retry to 15 seconds.
>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>> 
>> and no way to make/get calls until I restart FS. I did this:
>> 1. log 7
>> 2. sofia profile xxxx siptrace on   for each profile/gateway
>> 3. restarted router
>> 
>> All three did not solve the problem. The trace and log produced no additional lines which is why I am wondering if FS has a problem since the trace shows no SIP activity.
>> 
>> 3 gateways with 2 ITSPs
>> 2 DSL/WAN lines, 1 static and 1 dynamic
>> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the external static ip.
>> sofia status profile ... has the right ext ip
>> nat_map status shows the dynamic (wrong) IP
>> I tried starting with -nonat but that was worse
>> the only way to fix is restart FS.
>> 
>> I read the wiki on external nat, auto_nat and everything else many times.
>> Thanks Mario
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From xyangni at gmail.com  Tue Oct 26 09:09:41 2010
From: xyangni at gmail.com (xuyan yang)
Date: Wed, 27 Oct 2010 00:09:41 +0800
Subject: [Freeswitch-users] How to get answer state from fs_cli?
Message-ID: <AANLkTiniEwo8mQn8eVCK9U_RAzPqt6ZhdQnBU0qhG8tX@mail.gmail.com>

Hi,

According to channel_variable page I have tried uuid_getvar <uuid> state.
But is always return _undef_
I can get other variables such as destination in this way but not answer
state.
What should I do to get this information in command or ESL mode? Thanks.
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From mario_fs at mgtech.com  Tue Oct 26 09:14:32 2010
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 26 Oct 2010 09:14:32 -0700
Subject: [Freeswitch-users] FS not detecting public IP address change
In-Reply-To: <AANLkTikL=kvbfLQx3E-vCaErRM-ARnU3_YSAvKo67Uvy@mail.gmail.com>
References: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>
	<AANLkTinEoTcLmUVcpras8-m8pdkfpE=_r0spz1S50iMF@mail.gmail.com>
	<AANLkTi=MoV27FW0Ukd6K+mqmgnkyNgb9hjFx-vBYV1ak@mail.gmail.com>
	<AANLkTimCRfKSWG+5OdQY0=HTt2kOqxwk+0vKvaBNdmAd@mail.gmail.com>
	<AANLkTikfpRi9Shxiqd2w5nz3ZHsBnrHTt+=rHHnvoDJn@mail.gmail.com>
	<AANLkTikH3JJCCpL5s_eF2qjjttnEetE0i=nPn87zhBUL@mail.gmail.com>
	<AANLkTimzkb7qY8nW3M2afTPn18=OxY6jQye8D9TKr9Ts@mail.gmail.com>
	<AANLkTimR2x5AQPNiDDt4fZ8F=z_Wep3TBVR9Lob_72OA@mail.gmail.com>
	<AANLkTi=JgCME8u=27kUyLGU3cMP2nZpFpu7Zd5-4tpj0@mail.gmail.com>
	<AANLkTikL=kvbfLQx3E-vCaErRM-ARnU3_YSAvKo67Uvy@mail.gmail.com>
Message-ID: <AF5D2E2E-F8A7-44AB-8EA0-D240A28A51F6@mgtech.com>

What you said: "And nat_map status shows me which ports have been forwarded and the external ip address (which is sometimes wrong).  I also see this in the router gui." is also happening to me on a Linksys/Cisco RV042 router. Can't keep FS working for more than 2-5 hours without doing a restart or nat_map reinit. I think the routers upnp has a problem with dual wans, I have one static and one dynamic DSL.

On Oct 26, 2010, at 6:43 AM, Rupa Schomaker wrote:

> What do you have set for loglevel in switch.conf.xml?  My guess is that you have it set to something other than debug.  If that is set to (say) info, then info is as low as you can go.  Setting the console log level to debug will not give you debug output.
> 
> sofia is unaware of upnp so a keepalive packet should have no impact on it's operation.  I'm not familiar enough with sofia to tell what triggers that event.
> 
> On Tue, Oct 26, 2010 at 6:24 AM, Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
> OK..  tcpdump gave me this and I still do not see anything on FS except for this line:
> 
> 2010-10-26 22:02:51.235608 [DEBUG] sofia.c:957 nua_i_outbound: unknown event 8: 102 NAT binding changed
> 
> Hypertext Transfer Protocol
>     NOTIFY * HTTP/1.1\r\n
>     HOST:239.255.255.250:1900\r\n
>     Cache-Control:max-age=120\r\n
>     Location:http://192.168.1.1:1278/rootDesc.xml\r\n
>     Server: Tomato UPnP/1.0 MiniUPnPd/1.4\r\n
>     NT:urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>     USN:uuid:882dbe81-c28a-4992-baf7-410c05f1caf4::urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>     NTS:ssdp:alive\r\n
>     \r\n
> 
> This looks the same as you posted before, so why do I get the unknown event 8 (is this related)?
> 
> Thanks!
> 
> On Tue, Oct 26, 2010 at 3:32 PM, Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
> :)  that was the first thing I did, but I get nothing; no keep alive packets.
> 
> in the logfile at startup I see:
> 
> 2010-10-26 10:19:45.484258 [INFO] switch_nat.c:410 Scanning for NAT
> 2010-10-26 10:19:45.485991 [DEBUG] switch_nat.c:167 Checking for PMP 1/5
> 2010-10-26 10:19:45.488575 [INFO] switch_nat.c:423 NAT detected type: pmp, ExtIP: '203.xxx.xxx.xxx'
> 2010-10-26 10:19:45.490683 [DEBUG] switch_nat.c:256 NAT thread configured
> 2010-10-26 10:19:45.492180 [DEBUG] switch_nat.c:267 NAT thread started
> 
> And nat_map status shows me which ports have been forwarded and the external ip address (which is sometimes wrong).  I also see this in the router gui.  
> 
> But I never see the keep alive packets.
> 
> On Tue, Oct 26, 2010 at 3:13 PM, Rupa Schomaker <rupa at rupa.com> wrote:
> Just turn up console logging to level 7.
> 
> /log 7 if using fs_cli
> 
> 
> On Mon, Oct 25, 2010 at 6:53 PM, Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
> Hi!
> 
> How do I enable debugging to get the UPnP keep alive packets?
> 
> Thanks!
> 
> On Tue, Oct 26, 2010 at 1:15 AM, Rupa Schomaker <rupa at rupa.com> wrote:
> I need the debug logging from the upnp stuff.  It should look something like:
> 
> +OK log level 7 [7]
> freeswitch at internal> 2010-10-25 09:13:59.926861 [DEBUG] switch_nat.c:299 got UPnP keep alive packet:
> NOTIFY * HTTP/1.1
> HOST:239.255.255.250:1900
> Cache-Control:max-age=60
> Location:http://192.168.1.1:5000/rootDesc.xml
> Server: Tomato UPnP/1.0 MiniUPnPd/1.4
> NT:urn:schemas-upnp-org:service:WANIPConnection:1
> USN:uuid:00000000-0000-0000-0000-000000000000::urn:schemas-upnp-org:service:WANIPConnection:1
> NTS:ssdp:alive
> 
> I also run tomato and haven't had trouble with the upnp support so at least we have that part working.
> 
> You should see the above keepalive appear periodically every 30s or so.  You should see another set of messages when you terminate the DSL connection and when that dsl connection comes back online.
> 
> On Sat, Oct 23, 2010 at 5:36 PM, Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
> Hi!
> 
> The router is a ASUS router but it is running the tomato firmware which runs miniupnd.
> 
> In the debug I see the message: 2010-10-24 08:02:59.178918 [DEBUG] sofia.c:956 nua_i_outbound: unknown event 8: 101 NAT detected
> 
> This is what I see below. I hope I enabled all debug messages.  I thought I saw some xml type messages earlier, but not sure how I enabled them.
> 
> nta_outgoing: RTT is 67.958 ms
> outbound(0xb6e07c00): NAT binding changed: [210.xx.xxx.xx]:5080 != [58.xxx.xx.xx]:5080
> nua(0xb6e07c00): event i_outbound 102 NAT binding changed
> nua: nua_application_event: entering
> 2010-10-24 09:05:08.849525 [DEBUG] sofia.c:956 nua_i_outbound: unknown event 8: 102 NAT binding changed
> 
> 
> On Sun, Oct 24, 2010 at 1:08 AM, Rupa Schomaker <rupa at rupa.com> wrote:
> btw: what router are you using?
> 
> 
> On Sat, Oct 23, 2010 at 9:06 AM, Rupa Schomaker <rupa at rupa.com> wrote:
> In a upnp config, FS is depending on the router to notify it of the ip address change.  It is possible the upnp message(s) aren't formatted as expected.  If you turn on debug logging you'll get the upnp messages on the console.  If you can pastebin the capture of those while dropping and restarting the dsl connection I can maybe see if there is something obvious going on.  The debug should also have messages related to the processing of those upnp messages.
> 
> On Sat, Oct 23, 2010 at 7:04 AM, Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
> HI!
> 
> I know this question must have been answered 100's of times....
> 
> My adsl is a bit dodgy at the moment and tends to go up and down; which means I get assigned a new ip address from my ISP.  FS is nat'd behind a upnp capable router.
> 
> The problem is that FS does not seem to be detecting the change.   For both the internal and external profile, I have auto-nat setup, and I have tried stun and host settings in vars.conf.  When I issue a nat_map status I see my old IP address, and if I then issue a nat_map reinit, I see the new public IP address.  How can I get this to be automatic?
> 
> Also the public IP address shown in the nat_map status is not the same as shown in the sofia profile internal/external printouts.  Why is this?
> 
> I'm sure this is configuration, just not sure what to change.
> 
> Thanks!
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> -Rupa
> 
> 
> 
> -- 
> -Rupa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> -Rupa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> -Rupa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> -Rupa
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mario_fs at mgtech.com  Tue Oct 26 10:01:10 2010
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 26 Oct 2010 10:01:10 -0700
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
	<407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>
Message-ID: <EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>

From the TSP:
"I have enabled the SIP trace on your account. We are not currently seeing any registration attempts to your account within the last 15 minutes. Please restart FreeSwitch so that registration attempts begin again. Thank you. ". So FS is not getting past router.

On Oct 26, 2010, at 9:09 AM, Mario G wrote:

> I ran the global trace during the problem and it is at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone", "accttwo", "acct3". The trace includes phones since it was global. I am using:
>     <param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>     <param name="ext-sip-ip" value="autonat:my-static.ip"/> 
> 
> I tried dumping nat and removing the autonat: above and using -nonat but that did not work, registration proceeded but no calls inbound.
> 
> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
> 
>> Whoops, I am using an IP address for at least one gateway so that is not the problem:
>> They look outbound to me and I am using dns for 2 and an IP for one so that is not the issue. I was able to get FS to clear this up by doing "nat_map reinit" which is why I think this is a nat problem. I will do the trace you mentioned. I will plug an ip address into one of the gateways to see what happens, they all fail at once. Thanks for responding!
>> Mario
>> 
>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>> 
>>> I really need help on this as I have weeks into this problem. I thought I had it nailed but I guess not. After 5.5 hours I get:
>>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed Registration, setting retry to 15 seconds.
>>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed Registration, setting retry to 30 seconds.
>>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed Registration, setting retry to 15 seconds.
>>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed Registration, setting retry to 15 seconds.
>>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>>> 
>>> and no way to make/get calls until I restart FS. I did this:
>>> 1. log 7
>>> 2. sofia profile xxxx siptrace on   for each profile/gateway
>>> 3. restarted router
>>> 
>>> All three did not solve the problem. The trace and log produced no additional lines which is why I am wondering if FS has a problem since the trace shows no SIP activity.
>>> 
>>> 3 gateways with 2 ITSPs
>>> 2 DSL/WAN lines, 1 static and 1 dynamic
>>> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the external static ip.
>>> sofia status profile ... has the right ext ip
>>> nat_map status shows the dynamic (wrong) IP
>>> I tried starting with -nonat but that was worse
>>> the only way to fix is restart FS.
>>> 
>>> I read the wiki on external nat, auto_nat and everything else many times.
>>> Thanks Mario
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Tue Oct 26 10:15:58 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 Oct 2010 12:15:58 -0500
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
	<407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>
	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
Message-ID: <AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>

add

<variable name="sip-force-expires" value="30"/>

to the <variables> section of your <user>

you have it at 600 and the nat mapping is timing out while the 600
seconds is ticking away



On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
> From the TSP:
> "I have enabled the SIP trace on your account. We are not currently seeing
> any registration attempts to your account within the last 15 minutes. Please
> restart FreeSwitch so that registration attempts begin again. Thank you.?".
> So FS is not getting past router.
> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>
> I ran the global trace during the problem and it is
> at?http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
> "accttwo", "acct3". The trace includes phones since it was global. I am
> using:
> ?? ?<param name="ext-rtp-ip" value="autonat:my-static.ip"/>
> ? ? <param name="ext-sip-ip" value="autonat:my-static.ip"/>
> I tried dumping nat and removing the autonat: above and using -nonat but
> that did not work, registration proceeded but no calls inbound.
> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>
> Whoops, I am using an IP address for at least one gateway so that is not the
> problem:
> They look outbound to me and I am using dns for 2 and an IP for one so that
> is not the issue. I was able to get FS to clear this up by doing "nat_map
> reinit" which is why I think this is a nat problem. I will do the trace you
> mentioned. I will plug an ip address into one of the gateways to see what
> happens, they all fail at once. Thanks for responding!
> Mario
>
> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>
> I really need help on this as I have weeks into this problem. I thought I
> had it nailed but I guess not. After 5.5 hours I get:
>
> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
> Registration, setting retry to 15 seconds.
>
> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>
> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>
> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
> Registration, setting retry to 30 seconds.
>
> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
> Registration, setting retry to 15 seconds.
>
> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
> Registration, setting retry to 15 seconds.
>
> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>
> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>
> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>
> and no way to make/get calls until I restart FS. I did this:
>
> 1. log 7
>
> 2. sofia profile xxxx siptrace on ??for each profile/gateway
>
> 3. restarted router
>
> All three did not solve the problem. The trace and log produced no
> additional lines which is why I am wondering if FS has a problem since the
> trace shows no SIP activity.
>
> 3 gateways with 2 ITSPs
>
> 2 DSL/WAN lines, 1 static and 1 dynamic
>
> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
> external static ip.
>
> sofia status profile ... has the right ext ip
>
> nat_map status shows the dynamic (wrong) IP
>
> I tried starting with -nonat but that was worse
>
> the only way to fix is restart FS.
>
> I read the wiki on external nat, auto_nat and everything else many times.
>
> Thanks Mario
>
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From mario_fs at mgtech.com  Tue Oct 26 10:34:20 2010
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 26 Oct 2010 10:34:20 -0700
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
	<407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>
	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
	<AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
Message-ID: <6D9FD5E2-21FA-4630-8CF4-A87ADB30AC87@mgtech.com>

I made the change. I had no idea the settings for the inside phones effected nat for the outside sip accounts. I was looking into aggressive-nat- detection since the internal profile status always shows the right external static IP but the nat_ap status always shows the dynamic ip. Crossing fingers/etc since this problem is 85% of time (weeks!) into FS changeover. Thanks!
Mario

On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:

> add
> 
> <variable name="sip-force-expires" value="30"/>
> 
> to the <variables> section of your <user>
> 
> you have it at 600 and the nat mapping is timing out while the 600
> seconds is ticking away
> 
> 
> 
> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
>> From the TSP:
>> "I have enabled the SIP trace on your account. We are not currently seeing
>> any registration attempts to your account within the last 15 minutes. Please
>> restart FreeSwitch so that registration attempts begin again. Thank you. ".
>> So FS is not getting past router.
>> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>> 
>> I ran the global trace during the problem and it is
>> at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
>> "accttwo", "acct3". The trace includes phones since it was global. I am
>> using:
>>     <param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>>     <param name="ext-sip-ip" value="autonat:my-static.ip"/>
>> I tried dumping nat and removing the autonat: above and using -nonat but
>> that did not work, registration proceeded but no calls inbound.
>> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>> 
>> Whoops, I am using an IP address for at least one gateway so that is not the
>> problem:
>> They look outbound to me and I am using dns for 2 and an IP for one so that
>> is not the issue. I was able to get FS to clear this up by doing "nat_map
>> reinit" which is why I think this is a nat problem. I will do the trace you
>> mentioned. I will plug an ip address into one of the gateways to see what
>> happens, they all fail at once. Thanks for responding!
>> Mario
>> 
>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>> 
>> I really need help on this as I have weeks into this problem. I thought I
>> had it nailed but I guess not. After 5.5 hours I get:
>> 
>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>> Registration, setting retry to 15 seconds.
>> 
>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>> 
>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>> 
>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
>> Registration, setting retry to 30 seconds.
>> 
>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>> Registration, setting retry to 15 seconds.
>> 
>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>> Registration, setting retry to 15 seconds.
>> 
>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>> 
>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>> 
>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>> 
>> and no way to make/get calls until I restart FS. I did this:
>> 
>> 1. log 7
>> 
>> 2. sofia profile xxxx siptrace on   for each profile/gateway
>> 
>> 3. restarted router
>> 
>> All three did not solve the problem. The trace and log produced no
>> additional lines which is why I am wondering if FS has a problem since the
>> trace shows no SIP activity.
>> 
>> 3 gateways with 2 ITSPs
>> 
>> 2 DSL/WAN lines, 1 static and 1 dynamic
>> 
>> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
>> external static ip.
>> 
>> sofia status profile ... has the right ext ip
>> 
>> nat_map status shows the dynamic (wrong) IP
>> 
>> I tried starting with -nonat but that was worse
>> 
>> the only way to fix is restart FS.
>> 
>> I read the wiki on external nat, auto_nat and everything else many times.
>> 
>> Thanks Mario
>> 
>> 
>> _______________________________________________
>> 
>> FreeSWITCH-users mailing list
>> 
>> FreeSWITCH-users at lists.freeswitch.org
>> 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From pjintheusa at gmail.com  Tue Oct 26 10:53:59 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 26 Oct 2010 13:53:59 -0400
Subject: [Freeswitch-users] Using the bridge | operator for Implementing
	Failover
In-Reply-To: <AANLkTincare8CYw1bTVcG+Z7_DwHHo7CqahQv8qpp_8H@mail.gmail.com>
References: <AANLkTimZonOAnpKJKyy24ZnyJD9HnDPYMju40MYL6CvH@mail.gmail.com>
	<AANLkTimoE7t0LQwK3Nmv8Km-KSHHAtk+n-Teeiv7zhoQ@mail.gmail.com>
	<AANLkTimfXA70zUQKK0cBoKyvjE2OhfaaxckTPMgpYwHu@mail.gmail.com>
	<AANLkTimT0QTVcJKkczFnuwQJMBmgdfjuG7vr6UxxAx0k@mail.gmail.com>
	<AANLkTikomdMk_s3GovN_gkxm6u2U-_p3Y67Jw6bHLLrt@mail.gmail.com>
	<1287872231993-5666526.post@n2.nabble.com>
	<AANLkTi=JfTPXxAwyPPJ1CuKdmJ1pcys=18EE3apW9D1b@mail.gmail.com>
	<AANLkTinXG9koM_rVFW_gr5EK_KDVK7=-fhFpNw0yVDy7@mail.gmail.com>
	<AANLkTincare8CYw1bTVcG+Z7_DwHHo7CqahQv8qpp_8H@mail.gmail.com>
Message-ID: <AANLkTi=8uj_6U4NG=R+Xh2psYpV15ubPgQ=yTe6VhBuH@mail.gmail.com>

Anthony - Any other thoughts on this? Anyone?

On Mon, Oct 25, 2010 at 4:25 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Anthony,
>
> Thanks. I updated to the latest trunk and retested. Unfortunately I got
> exactly the same result - 2158824374 is called by each gateway specified....
>
> Pj
>
>
>
>
> On Mon, Oct 25, 2010 at 2:09 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> can you try latest
>>
>> On Sun, Oct 24, 2010 at 10:09 AM, Phillip Jones <pjintheusa at gmail.com>
>> wrote:
>> > Jeff - Thanks for the response.
>> >
>> > <condition field="destination_number" expression="^(3478309992)$">
>> >       <action application="answer"/>
>> >       <action application="sleep" data="2000"/>
>> >        <!--action application="set"
>> >
>> data="fail_on_single_reject=NO_ANSWER,NORMAL_CLEARING,ALLOTTED_TIMEOUT"/-->
>> >        <action application="set" data="fail_on_single_reject=true"/>
>> >        <action application="bridge"
>> >
>> data="{ignore_early_media=true,originate_continue_on_timeout=false}[leg_timeout=10]sofia/gateway/broadvox1/2158824374|[leg_timeout=25]sofia/gateway/broadvox2/2158824374"/>
>> >  </condition>
>> > This gives exactly the same result. The 2158824374 is called by each
>> gateway
>> > specified.
>> >
>> > Pj
>> > On Sat, Oct 23, 2010 at 6:17 PM, Jeff Lenk <jeff at jefflenk.com> wrote:
>> >>
>> >> what happens with  <action application="set"
>> >> data="fail_on_single_reject=true"/>
>> >>
>> >> --
>> >> View this message in context:
>> >>
>> http://freeswitch-users.2379917.n2.nabble.com/Using-the-bridge-operator-for-Implementing-Failover-tp5661010p5666526.html
>> >> Sent from the freeswitch-users mailing list archive at Nabble.com.
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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 <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
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From anthony.minessale at gmail.com  Tue Oct 26 11:51:06 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 Oct 2010 13:51:06 -0500
Subject: [Freeswitch-users] Using the bridge | operator for Implementing
	Failover
In-Reply-To: <AANLkTi=8uj_6U4NG=R+Xh2psYpV15ubPgQ=yTe6VhBuH@mail.gmail.com>
References: <AANLkTimZonOAnpKJKyy24ZnyJD9HnDPYMju40MYL6CvH@mail.gmail.com>
	<AANLkTimoE7t0LQwK3Nmv8Km-KSHHAtk+n-Teeiv7zhoQ@mail.gmail.com>
	<AANLkTimfXA70zUQKK0cBoKyvjE2OhfaaxckTPMgpYwHu@mail.gmail.com>
	<AANLkTimT0QTVcJKkczFnuwQJMBmgdfjuG7vr6UxxAx0k@mail.gmail.com>
	<AANLkTikomdMk_s3GovN_gkxm6u2U-_p3Y67Jw6bHLLrt@mail.gmail.com>
	<1287872231993-5666526.post@n2.nabble.com>
	<AANLkTi=JfTPXxAwyPPJ1CuKdmJ1pcys=18EE3apW9D1b@mail.gmail.com>
	<AANLkTinXG9koM_rVFW_gr5EK_KDVK7=-fhFpNw0yVDy7@mail.gmail.com>
	<AANLkTincare8CYw1bTVcG+Z7_DwHHo7CqahQv8qpp_8H@mail.gmail.com>
	<AANLkTi=8uj_6U4NG=R+Xh2psYpV15ubPgQ=yTe6VhBuH@mail.gmail.com>
Message-ID: <AANLkTinYywmVD3RLPLQUD9gnNCLNyOZT_DvqLf50JPX8@mail.gmail.com>

I don't think you actually updated:
see below


freeswitch at deathstar.freeswitch.org> originate
{fail_on_single_reject=ALLOTTED_TIMEOUT}error/ALLOTTED_TIMEOUT|sofia/internal/1235 at conference.freeswitch.org
1234
2010-10-26 13:44:01.607864 [ERR] switch_ivr_originate.c:2612 Cannot
create outgoing channel of type [error] cause: [ALLOTTED_TIMEOUT]

-ERR ALLOTTED_TIMEOUT

freeswitch at deathstar.freeswitch.org> originate
error/ALLOTTED_TIMEOUT|sofia/internal/1235 at conference.freeswitch.org
1234                                        2010-10-26 13:44:09.407870
[ERR] switch_ivr_originate.c:2612 Cannot create outgoing channel of
type [error] cause: [ALLOTTED_TIMEOUT]
2010-10-26 13:44:09.407870 [NOTICE] switch_channel.c:784 New Channel
sofia/internal/1235 at conference.freeswitch.org
[91bdc528-23cd-429e-9d46-c9e97950599a]
2010-10-26 13:44:09.413867 [WARNING] switch_core_port_allocator.c:78
Rounding odd end port 65535 to 65534
2010-10-26 13:44:09.573868 [INFO] sofia.c:709
sofia/internal/1235 at conference.freeswitch.org Update Callee ID to
"Imperial March" <sip:66 at 74.112.133.77>
2010-10-26 13:44:09.577867 [NOTICE] sofia.c:4659 Ring-Ready
sofia/internal/1235 at conference.freeswitch.org!
2010-10-26 13:44:09.681871 [NOTICE] sofia.c:5160 Channel
[sofia/internal/1235 at conference.freeswitch.org] has been answered
2010-10-26 13:44:09.683869 [NOTICE] switch_ivr.c:1507 Transfer
sofia/internal/1235 at conference.freeswitch.org to XML[1234 at default]
2010-10-26 13:44:09.683869 [INFO] mod_dialplan_xml.c:331 Processing
Imperial March <66>->1234 in context default

+OK 91bdc528-23cd-429e-9d46-c9e97950599a

2010-10-26 13:44:09.685878 [INFO] switch_ivr_async.c:2991 Bound A-Leg:
*1 transfer::-both 3000
freeswitch at deathstar.freeswitch.org> 2010-10-26 13:44:09.685878 [INFO]
switch_core_session.c:1176
sofia/internal/1235 at conference.freeswitch.org setting session
heartbeat to 5 second(s).
2010-10-26 13:44:09.685878 [NOTICE] switch_channel.c:784 New Channel
sofia/internal/sr at conference.freeswitch.org
[80b3929b-a77f-4b4f-b9d2-433edbb36e30]
2010-10-26 13:44:09.895864 [INFO] sofia.c:709
sofia/internal/sr at conference.freeswitch.org Update Callee ID to "sr"
<sip:sr at 74.112.133.77>
2010-10-26 13:44:09.901868 [NOTICE] sofia.c:5160 Channel
[sofia/internal/sr at conference.freeswitch.org] has been answered









On Tue, Oct 26, 2010 at 12:53 PM, Phillip Jones <pjintheusa at gmail.com> wrote:
> Anthony - Any other thoughts on this? Anyone?
>
> On Mon, Oct 25, 2010 at 4:25 PM, Phillip Jones <pjintheusa at gmail.com> wrote:
>>
>> Anthony,
>>
>> Thanks. I updated to the latest trunk and retested. Unfortunately I got
>> exactly the same result - 2158824374 is called by each gateway specified....
>>
>> Pj
>>
>>
>>
>> On Mon, Oct 25, 2010 at 2:09 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>>
>>> can you try latest
>>>
>>> On Sun, Oct 24, 2010 at 10:09 AM, Phillip Jones <pjintheusa at gmail.com>
>>> wrote:
>>> > Jeff - Thanks for the response.
>>> >
>>> > <condition field="destination_number" expression="^(3478309992)$">
>>> > ????? <action application="answer"/>
>>> > ?? ?? <action application="sleep" data="2000"/>
>>> > ??? ?? <!--action application="set"
>>> >
>>> > data="fail_on_single_reject=NO_ANSWER,NORMAL_CLEARING,ALLOTTED_TIMEOUT"/-->
>>> > ?????? <action application="set" data="fail_on_single_reject=true"/>
>>> > ?????? <action application="bridge"
>>> >
>>> > data="{ignore_early_media=true,originate_continue_on_timeout=false}[leg_timeout=10]sofia/gateway/broadvox1/2158824374|[leg_timeout=25]sofia/gateway/broadvox2/2158824374"/>
>>> > ?</condition>
>>> > This gives exactly the same result. The 2158824374 is called by each
>>> > gateway
>>> > specified.
>>> >
>>> > Pj
>>> > On Sat, Oct 23, 2010 at 6:17 PM, Jeff Lenk <jeff at jefflenk.com> wrote:
>>> >>
>>> >> what happens with ?<action application="set"
>>> >> data="fail_on_single_reject=true"/>
>>> >>
>>> >> --
>>> >> View this message in context:
>>> >>
>>> >> http://freeswitch-users.2379917.n2.nabble.com/Using-the-bridge-operator-for-Implementing-Failover-tp5661010p5666526.html
>>> >> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > 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
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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
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FreeSWITCH Developer Conference
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googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From covici at ccs.covici.com  Tue Oct 26 12:21:21 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 26 Oct 2010 15:21:21 -0400
Subject: [Freeswitch-users] transfer problem
In-Reply-To: <AANLkTi==CoAgm9_foaPEWDpBmME2HWq5hoskDNjE4MA6@mail.gmail.com>
References: <13464.1287506974@ccs.covici.com>
	<AANLkTikh67cGuMn3v4D3QZ+AuQ0vWrLE_fRhL5+g1fg7@mail.gmail.com>
	<19713.1288004756@ccs.covici.com>
	<AANLkTinA4Egvfr1N-EvLun6rh-BR=WxwutHdOryACnYp@mail.gmail.com>
	<28001.1288041100@ccs.covici.com>
	<AANLkTinVAP-5A4KA7gz6hbRUVw7qh6Kc+ty48gs4eucZ@mail.gmail.com>
	<29557.1288051598@ccs.covici.com>
	<AANLkTi==CoAgm9_foaPEWDpBmME2HWq5hoskDNjE4MA6@mail.gmail.com>
Message-ID: <12571.1288120881@ccs.covici.com>

HI.  Well, Here is the extension in conferences context, where I did an
export of the variable, but it is still not working -- how do I get the
variable to be on the original channel in this case 7034754612 as well
as the 202?

<extension name="utah">
<condition field="destination_number" expression="^utah$">
<action application="export" data="force_transfer_context=conferences"/>
<action application="system" data="sayconf -a local -a utah -a
${caller_id_number}  &amp;"/>
<action application="playback" data="conference/conf-muted.wav"/>
<action application="playback" data="conference/press_0_to_unmute.wav"/>
<action application="conference" data="utah at all"/>
</condition>
</extension>

Anthony Minessale <anthony.minessale at gmail.com> wrote:

> set the variable force_transfer_context to the desired context on all
> channels involved
> 
> On Mon, Oct 25, 2010 at 7:06 PM,  <covici at ccs.covici.com> wrote:
> > I can capture the log, but let me tell you the steps. ?202 does call
> > 7034754612, then it calls 7000, which is an ivr which transfers to an
> > extensionn in the context conferences, which in turn puts the caller in
> > a conference. ?So 202 calls 7034754612 and then is in this conference
> > whose context is not default. ?Now I flash the hook on 202, so
> > 7034754612 can hear the conference and also can hear 202. ?Now, what I
> > was hoping would happen is that I could hang up 202 and 4754612 would be
> > in the conference, but instead 4754612 hangs up and ?this seems to be
> > because the system tries to transfer 4754612 to theconference name but
> > in the default context and I get
> > 2010-10-25 17:09:46.137372 [DEBUG] mod_enum.c:204 Unable to lookup NAPTR
> > record for e164.org: valid domain but no data of requested type
> > and the 4754612 hangs up.
> >
> > I hope this clarifies what the problem is.
> >
> > Michael Collins <msc at freeswitch.org> wrote:
> >
> >> John,
> >>
> >> Perhaps I've missed something. Can you confirm exactly what steps you are
> >> taking? I don't want to make any assumptions. I can see that ext 202 calls
> >> out to 7034754612 and then 202 also makes a call to ext 7000. What I don't
> >> know is if that's a transfer or 3-way or what. Also, can you capture the log
> >> again, this time with a sip trace? Use "sofia global siptrace on".
> >>
> >> Thanks,
> >> MC
> >>
> >> On Mon, Oct 25, 2010 at 2:11 PM, <covici at ccs.covici.com> wrote:
> >>
> >> > Thanks for the correction -- however still same result. ?I did not think
> >> > it would matter in this case because the call did reach the phone and
> >> > everything after that is local. ?Question is why does it try to transfer
> >> > to the wrong context after the local extension hangs up?
> >> >
> >> > Michael Collins <msc at freeswitch.org> wrote:
> >> >
> >> > > John,
> >> > >
> >> > > I see two issues. At line #174 of your pastebin there is a message about
> >> > > "invalid gateway". I suspect that the gateway name should be "flowroute"
> >> > and
> >> > > not "fflowroute".
> >> > >
> >> > > At line #180 you have an invalid number format error. I would fix the
> >> > > invalid gateway first and retest as this second error may be a result of
> >> > the
> >> > > bad gateway.
> >> > >
> >> > > -MC
> >> > >
> >> > > On Mon, Oct 25, 2010 at 4:05 AM, <covici at ccs.covici.com> wrote:
> >> > >
> >> > > > OK, the debug log is here http://pastebin.freeswitch.org/14307. ?The
> >> > > > relevant part of the configs is below.
> >> > > > ? ?<extension name="conference_ivr">
> >> > > > ? ? ?<condition field="destination_number" expression="^7000$">
> >> > > > ? ? ? ?<action application="answer"/>
> >> > > > <action application="sleep" data="500"/>
> >> > > > <action application="set" data="api_hangup_hook=system sayconf -a local
> >> > ?-a
> >> > > > ${caller_id_number} -a hung -a up"/>
> >> > > > ? ? ? ?<action application="ivr" data="conference_ivr"/>
> >> > > > ? ? ?</condition>
> >> > > > ? ?</extension>
> >> > > > and the ivr just asks for passwords and send the call to an extension
> >> > in
> >> > > > context conferences.
> >> > > >
> >> > > > Hope this helps.
> >> > > >
> >> > > > Thanks.
> >> > > >
> >> > > > Michael Collins <msc at freeswitch.org> wrote:
> >> > > >
> >> > > > > Please supply the configs and a debug trace. My guess is that there's
> >> > > > > probably just a simple config element that is wrong or missing.
> >> > > > >
> >> > > > > -MC
> >> > > > >
> >> > > > > On Tue, Oct 19, 2010 at 9:49 AM, <covici at ccs.covici.com> wrote:
> >> > > > >
> >> > > > > > Hi. ?I have an ivr which connects to conferences which I have in
> >> > their
> >> > > > > > own context rather than the default. ?The problem is that if I call
> >> > > > > > someone else and have a 3-way between the other party and the
> >> > > > > > conference, its fine till I hang up and then fs tries to transfer
> >> > to
> >> > > > the
> >> > > > > > correct conference name, but uses default for the context instead.
> >> > ?Is
> >> > > > > > this expected behavior or should I file a bug?
> >> > > > > >
> >> > > > > > Thanks.
> >> > > > > >
> >> > > > > > --
> >> > > > > > Your life is like a penny. ?You're going to lose it. ?The question
> >> > is:
> >> > > > > > How do
> >> > > > > > you spend it?
> >> > > > > >
> >> > > > > > ? ? ? ? John Covici
> >> > > > > > ? ? ? ? covici at ccs.covici.com
> >> > > > > >
> >> > > > > > _______________________________________________
> >> > > > > > FreeSWITCH-users mailing list
> >> > > > > > FreeSWITCH-users at lists.freeswitch.org
> >> > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > > > > > UNSUBSCRIBE:
> >> > > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > > > > > http://www.freeswitch.org
> >> > > > > >
> >> > > > >
> >> > > > > ----------------------------------------------------
> >> > > > > Alternatives:
> >> > > > >
> >> > > > > ----------------------------------------------------
> >> > > > > _______________________________________________
> >> > > > > FreeSWITCH-users mailing list
> >> > > > > FreeSWITCH-users at lists.freeswitch.org
> >> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > > > > UNSUBSCRIBE:
> >> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > > > > http://www.freeswitch.org
> >> > > >
> >> > > > --
> >> > > > Your life is like a penny. ?You're going to lose it. ?The question is:
> >> > > > How do
> >> > > > you spend it?
> >> > > >
> >> > > > ? ? ? ? John Covici
> >> > > > ? ? ? ? covici at ccs.covici.com
> >> > > >
> >> > > > _______________________________________________
> >> > > > FreeSWITCH-users mailing list
> >> > > > FreeSWITCH-users at lists.freeswitch.org
> >> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > > > UNSUBSCRIBE:
> >> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > > > http://www.freeswitch.org
> >> > > >
> >> > >
> >> > > ----------------------------------------------------
> >> > > Alternatives:
> >> > >
> >> > > ----------------------------------------------------
> >> > > _______________________________________________
> >> > > FreeSWITCH-users mailing list
> >> > > FreeSWITCH-users at lists.freeswitch.org
> >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > > http://www.freeswitch.org
> >> >
> >> > --
> >> > Your life is like a penny. ?You're going to lose it. ?The question is:
> >> > How do
> >> > you spend it?
> >> >
> >> > ? ? ? ? John Covici
> >> > ? ? ? ? covici at ccs.covici.com
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >> ----------------------------------------------------
> >> Alternatives:
> >>
> >> ----------------------------------------------------
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > --
> > Your life is like a penny. ?You're going to lose it. ?The question is:
> > How do
> > you spend it?
> >
> > ? ? ? ? John Covici
> > ? ? ? ? covici at ccs.covici.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From pjintheusa at gmail.com  Tue Oct 26 13:00:55 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 26 Oct 2010 16:00:55 -0400
Subject: [Freeswitch-users] Using the bridge | operator for Implementing
	Failover
In-Reply-To: <AANLkTinYywmVD3RLPLQUD9gnNCLNyOZT_DvqLf50JPX8@mail.gmail.com>
References: <AANLkTimZonOAnpKJKyy24ZnyJD9HnDPYMju40MYL6CvH@mail.gmail.com>
	<AANLkTimoE7t0LQwK3Nmv8Km-KSHHAtk+n-Teeiv7zhoQ@mail.gmail.com>
	<AANLkTimfXA70zUQKK0cBoKyvjE2OhfaaxckTPMgpYwHu@mail.gmail.com>
	<AANLkTimT0QTVcJKkczFnuwQJMBmgdfjuG7vr6UxxAx0k@mail.gmail.com>
	<AANLkTikomdMk_s3GovN_gkxm6u2U-_p3Y67Jw6bHLLrt@mail.gmail.com>
	<1287872231993-5666526.post@n2.nabble.com>
	<AANLkTi=JfTPXxAwyPPJ1CuKdmJ1pcys=18EE3apW9D1b@mail.gmail.com>
	<AANLkTinXG9koM_rVFW_gr5EK_KDVK7=-fhFpNw0yVDy7@mail.gmail.com>
	<AANLkTincare8CYw1bTVcG+Z7_DwHHo7CqahQv8qpp_8H@mail.gmail.com>
	<AANLkTi=8uj_6U4NG=R+Xh2psYpV15ubPgQ=yTe6VhBuH@mail.gmail.com>
	<AANLkTinYywmVD3RLPLQUD9gnNCLNyOZT_DvqLf50JPX8@mail.gmail.com>
Message-ID: <AANLkTinLgu+y7kvSe5c2jmcPYGwHpj526c88n=NSCKRW@mail.gmail.com>

Anthony - thanks for the response and you patience!

>From your example the fail_on_single_reject must be passed in the {} before
the dial string. Not set on the A leg as I was doing:

This does not work:

<action application="set" data="fail_on_single_reject=ALLOTTED_TIMEOUT"/>
<action application="bridge"
data="{ignore_early_media=true,originate_continue_on_timeout=false}[leg_timeout=10]sofia/gateway/broadvox1/2158824374|[leg_timeout=25]sofia/gateway/broadvox2/2158824374"/>

But this works now:

<action application="bridge"
data="{fail_on_single_reject=ALLOTTED_TIMEOUT,ignore_early_media=true,originate_continue_on_timeout=false}[leg_timeout=10]sofia/gateway/broadvox1/2158824374|[leg_timeout=25]sofia/gateway/broadvox2/2158824374"/>


Thanks again.




On Tue, Oct 26, 2010 at 2:51 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I don't think you actually updated:
> see below
>
>
> freeswitch at deathstar.freeswitch.org> originate
>
> {fail_on_single_reject=ALLOTTED_TIMEOUT}error/ALLOTTED_TIMEOUT|sofia/internal/
> 1235 at conference.freeswitch.org
> 1234
> 2010-10-26 13:44:01.607864 [ERR] switch_ivr_originate.c:2612 Cannot
> create outgoing channel of type [error] cause: [ALLOTTED_TIMEOUT]
>
> -ERR ALLOTTED_TIMEOUT
>
> freeswitch at deathstar.freeswitch.org> originate
> error/ALLOTTED_TIMEOUT|sofia/internal/1235 at conference.freeswitch.org
> 1234                                        2010-10-26 13:44:09.407870
> [ERR] switch_ivr_originate.c:2612 Cannot create outgoing channel of
> type [error] cause: [ALLOTTED_TIMEOUT]
> 2010-10-26 13:44:09.407870 [NOTICE] switch_channel.c:784 New Channel
> sofia/internal/1235 at conference.freeswitch.org
> [91bdc528-23cd-429e-9d46-c9e97950599a]
> 2010-10-26 13:44:09.413867 [WARNING] switch_core_port_allocator.c:78
> Rounding odd end port 65535 to 65534
> 2010-10-26 13:44:09.573868 [INFO] sofia.c:709
> sofia/internal/1235 at conference.freeswitch.org Update Callee ID to
> "Imperial March" <sip:66 at 74.112.133.77 <sip%3A66 at 74.112.133.77>>
> 2010-10-26 13:44:09.577867 [NOTICE] sofia.c:4659 Ring-Ready
> sofia/internal/1235 at conference.freeswitch.org!
> 2010-10-26 13:44:09.681871 [NOTICE] sofia.c:5160 Channel
> [sofia/internal/1235 at conference.freeswitch.org] has been answered
> 2010-10-26 13:44:09.683869 [NOTICE] switch_ivr.c:1507 Transfer
> sofia/internal/1235 at conference.freeswitch.org to XML[1234 at default]
> 2010-10-26 13:44:09.683869 [INFO] mod_dialplan_xml.c:331 Processing
> Imperial March <66>->1234 in context default
>
> +OK 91bdc528-23cd-429e-9d46-c9e97950599a
>
> 2010-10-26 13:44:09.685878 [INFO] switch_ivr_async.c:2991 Bound A-Leg:
> *1 transfer::-both 3000
> freeswitch at deathstar.freeswitch.org> 2010-10-26 13:44:09.685878 [INFO]
> switch_core_session.c:1176
> sofia/internal/1235 at conference.freeswitch.org setting session
> heartbeat to 5 second(s).
> 2010-10-26 13:44:09.685878 [NOTICE] switch_channel.c:784 New Channel
> sofia/internal/sr at conference.freeswitch.org
> [80b3929b-a77f-4b4f-b9d2-433edbb36e30]
> 2010-10-26 13:44:09.895864 [INFO] sofia.c:709
> sofia/internal/sr at conference.freeswitch.org Update Callee ID to "sr"
> <sip:sr at 74.112.133.77 <sip%3Asr at 74.112.133.77>>
> 2010-10-26 13:44:09.901868 [NOTICE] sofia.c:5160 Channel
> [sofia/internal/sr at conference.freeswitch.org] has been answered
>
>
>
>
>
>
>
>
>
> On Tue, Oct 26, 2010 at 12:53 PM, Phillip Jones <pjintheusa at gmail.com>
> wrote:
> > Anthony - Any other thoughts on this? Anyone?
> >
> > On Mon, Oct 25, 2010 at 4:25 PM, Phillip Jones <pjintheusa at gmail.com>
> wrote:
> >>
> >> Anthony,
> >>
> >> Thanks. I updated to the latest trunk and retested. Unfortunately I got
> >> exactly the same result - 2158824374 is called by each gateway
> specified....
> >>
> >> Pj
> >>
> >>
> >>
> >> On Mon, Oct 25, 2010 at 2:09 PM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >>>
> >>> can you try latest
> >>>
> >>> On Sun, Oct 24, 2010 at 10:09 AM, Phillip Jones <pjintheusa at gmail.com>
> >>> wrote:
> >>> > Jeff - Thanks for the response.
> >>> >
> >>> > <condition field="destination_number" expression="^(3478309992)$">
> >>> >       <action application="answer"/>
> >>> >       <action application="sleep" data="2000"/>
> >>> >        <!--action application="set"
> >>> >
> >>> >
> data="fail_on_single_reject=NO_ANSWER,NORMAL_CLEARING,ALLOTTED_TIMEOUT"/-->
> >>> >        <action application="set" data="fail_on_single_reject=true"/>
> >>> >        <action application="bridge"
> >>> >
> >>> >
> data="{ignore_early_media=true,originate_continue_on_timeout=false}[leg_timeout=10]sofia/gateway/broadvox1/2158824374|[leg_timeout=25]sofia/gateway/broadvox2/2158824374"/>
> >>> >  </condition>
> >>> > This gives exactly the same result. The 2158824374 is called by each
> >>> > gateway
> >>> > specified.
> >>> >
> >>> > Pj
> >>> > On Sat, Oct 23, 2010 at 6:17 PM, Jeff Lenk <jeff at jefflenk.com>
> wrote:
> >>> >>
> >>> >> what happens with  <action application="set"
> >>> >> data="fail_on_single_reject=true"/>
> >>> >>
> >>> >> --
> >>> >> View this message in context:
> >>> >>
> >>> >>
> http://freeswitch-users.2379917.n2.nabble.com/Using-the-bridge-operator-for-Implementing-Failover-tp5661010p5666526.html
> >>> >> Sent from the freeswitch-users mailing list archive at Nabble.com.
> >>> >>
> >>> >> _______________________________________________
> >>> >> FreeSWITCH-users mailing list
> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>
> >>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> http://www.freeswitch.org
> >>> >
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > 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<MSN%3Aanthony_minessale at hotmail.com>
> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> IRC: irc.freenode.net #freeswitch
> >>>
> >>> FreeSWITCH Developer Conference
> >>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> pstn:+19193869900
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Tue Oct 26 13:22:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 Oct 2010 15:22:49 -0500
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <6D9FD5E2-21FA-4630-8CF4-A87ADB30AC87@mgtech.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
	<407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>
	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
	<AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
	<6D9FD5E2-21FA-4630-8CF4-A87ADB30AC87@mgtech.com>
Message-ID: <AANLkTi=BiDiXLgEoj_MfwmZbGjvXoOuNJgUYdetciNCj@mail.gmail.com>

You should be setting the req freq to a low number on the outbound gateways
The examples you showed had a series of inbound reg

also set expire-seconds to 30 in your gateway xml


The problem is if you are not constantly sending traffic to the box
the nat mapping will go away.

If you are in production you should be using a static ip with a static
mapping, any trouble you are having is your own fault for playing with
fire.  The best we can do is tell you how to keep it contained.




On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com> wrote:
> I made the change. I had no idea the settings for the inside phones effected nat for the outside sip accounts. I was looking into aggressive-nat- detection since the internal profile status always shows the right external static IP but the nat_ap status always shows the dynamic ip. Crossing fingers/etc since this problem is 85% of time (weeks!) into FS changeover. Thanks!
> Mario
>
> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
>
>> add
>>
>> <variable name="sip-force-expires" value="30"/>
>>
>> to the <variables> section of your <user>
>>
>> you have it at 600 and the nat mapping is timing out while the 600
>> seconds is ticking away
>>
>>
>>
>> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
>>> From the TSP:
>>> "I have enabled the SIP trace on your account. We are not currently seeing
>>> any registration attempts to your account within the last 15 minutes. Please
>>> restart FreeSwitch so that registration attempts begin again. Thank you. ".
>>> So FS is not getting past router.
>>> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>>>
>>> I ran the global trace during the problem and it is
>>> at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
>>> "accttwo", "acct3". The trace includes phones since it was global. I am
>>> using:
>>> ? ? <param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>>> ? ? <param name="ext-sip-ip" value="autonat:my-static.ip"/>
>>> I tried dumping nat and removing the autonat: above and using -nonat but
>>> that did not work, registration proceeded but no calls inbound.
>>> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>>>
>>> Whoops, I am using an IP address for at least one gateway so that is not the
>>> problem:
>>> They look outbound to me and I am using dns for 2 and an IP for one so that
>>> is not the issue. I was able to get FS to clear this up by doing "nat_map
>>> reinit" which is why I think this is a nat problem. I will do the trace you
>>> mentioned. I will plug an ip address into one of the gateways to see what
>>> happens, they all fail at once. Thanks for responding!
>>> Mario
>>>
>>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>>>
>>> I really need help on this as I have weeks into this problem. I thought I
>>> had it nailed but I guess not. After 5.5 hours I get:
>>>
>>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>>> Registration, setting retry to 15 seconds.
>>>
>>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>
>>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>
>>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
>>> Registration, setting retry to 30 seconds.
>>>
>>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>>> Registration, setting retry to 15 seconds.
>>>
>>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>>> Registration, setting retry to 15 seconds.
>>>
>>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>
>>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>
>>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>>>
>>> and no way to make/get calls until I restart FS. I did this:
>>>
>>> 1. log 7
>>>
>>> 2. sofia profile xxxx siptrace on ? for each profile/gateway
>>>
>>> 3. restarted router
>>>
>>> All three did not solve the problem. The trace and log produced no
>>> additional lines which is why I am wondering if FS has a problem since the
>>> trace shows no SIP activity.
>>>
>>> 3 gateways with 2 ITSPs
>>>
>>> 2 DSL/WAN lines, 1 static and 1 dynamic
>>>
>>> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
>>> external static ip.
>>>
>>> sofia status profile ... has the right ext ip
>>>
>>> nat_map status shows the dynamic (wrong) IP
>>>
>>> I tried starting with -nonat but that was worse
>>>
>>> the only way to fix is restart FS.
>>>
>>> I read the wiki on external nat, auto_nat and everything else many times.
>>>
>>> Thanks Mario
>>>
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From anthony.minessale at gmail.com  Tue Oct 26 13:25:13 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 Oct 2010 15:25:13 -0500
Subject: [Freeswitch-users] transfer problem
In-Reply-To: <12571.1288120881@ccs.covici.com>
References: <13464.1287506974@ccs.covici.com>
	<AANLkTikh67cGuMn3v4D3QZ+AuQ0vWrLE_fRhL5+g1fg7@mail.gmail.com>
	<19713.1288004756@ccs.covici.com>
	<AANLkTinA4Egvfr1N-EvLun6rh-BR=WxwutHdOryACnYp@mail.gmail.com>
	<28001.1288041100@ccs.covici.com>
	<AANLkTinVAP-5A4KA7gz6hbRUVw7qh6Kc+ty48gs4eucZ@mail.gmail.com>
	<29557.1288051598@ccs.covici.com>
	<AANLkTi==CoAgm9_foaPEWDpBmME2HWq5hoskDNjE4MA6@mail.gmail.com>
	<12571.1288120881@ccs.covici.com>
Message-ID: <AANLkTimmgeivgE-+JbCQRcNvRGKhAE1mf9PUAWaJBb98@mail.gmail.com>

you need to learn to post logs to go with "it still doesn't work"
every new change you make needs a new log.


On Tue, Oct 26, 2010 at 2:21 PM,  <covici at ccs.covici.com> wrote:
> HI. ?Well, Here is the extension in conferences context, where I did an
> export of the variable, but it is still not working -- how do I get the
> variable to be on the original channel in this case 7034754612 as well
> as the 202?
>
> <extension name="utah">
> <condition field="destination_number" expression="^utah$">
> <action application="export" data="force_transfer_context=conferences"/>
> <action application="system" data="sayconf -a local -a utah -a
> ${caller_id_number} ?&amp;"/>
> <action application="playback" data="conference/conf-muted.wav"/>
> <action application="playback" data="conference/press_0_to_unmute.wav"/>
> <action application="conference" data="utah at all"/>
> </condition>
> </extension>
>
> Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
>> set the variable force_transfer_context to the desired context on all
>> channels involved
>>
>> On Mon, Oct 25, 2010 at 7:06 PM, ?<covici at ccs.covici.com> wrote:
>> > I can capture the log, but let me tell you the steps. ?202 does call
>> > 7034754612, then it calls 7000, which is an ivr which transfers to an
>> > extensionn in the context conferences, which in turn puts the caller in
>> > a conference. ?So 202 calls 7034754612 and then is in this conference
>> > whose context is not default. ?Now I flash the hook on 202, so
>> > 7034754612 can hear the conference and also can hear 202. ?Now, what I
>> > was hoping would happen is that I could hang up 202 and 4754612 would be
>> > in the conference, but instead 4754612 hangs up and ?this seems to be
>> > because the system tries to transfer 4754612 to theconference name but
>> > in the default context and I get
>> > 2010-10-25 17:09:46.137372 [DEBUG] mod_enum.c:204 Unable to lookup NAPTR
>> > record for e164.org: valid domain but no data of requested type
>> > and the 4754612 hangs up.
>> >
>> > I hope this clarifies what the problem is.
>> >
>> > Michael Collins <msc at freeswitch.org> wrote:
>> >
>> >> John,
>> >>
>> >> Perhaps I've missed something. Can you confirm exactly what steps you are
>> >> taking? I don't want to make any assumptions. I can see that ext 202 calls
>> >> out to 7034754612 and then 202 also makes a call to ext 7000. What I don't
>> >> know is if that's a transfer or 3-way or what. Also, can you capture the log
>> >> again, this time with a sip trace? Use "sofia global siptrace on".
>> >>
>> >> Thanks,
>> >> MC
>> >>
>> >> On Mon, Oct 25, 2010 at 2:11 PM, <covici at ccs.covici.com> wrote:
>> >>
>> >> > Thanks for the correction -- however still same result. ?I did not think
>> >> > it would matter in this case because the call did reach the phone and
>> >> > everything after that is local. ?Question is why does it try to transfer
>> >> > to the wrong context after the local extension hangs up?
>> >> >
>> >> > Michael Collins <msc at freeswitch.org> wrote:
>> >> >
>> >> > > John,
>> >> > >
>> >> > > I see two issues. At line #174 of your pastebin there is a message about
>> >> > > "invalid gateway". I suspect that the gateway name should be "flowroute"
>> >> > and
>> >> > > not "fflowroute".
>> >> > >
>> >> > > At line #180 you have an invalid number format error. I would fix the
>> >> > > invalid gateway first and retest as this second error may be a result of
>> >> > the
>> >> > > bad gateway.
>> >> > >
>> >> > > -MC
>> >> > >
>> >> > > On Mon, Oct 25, 2010 at 4:05 AM, <covici at ccs.covici.com> wrote:
>> >> > >
>> >> > > > OK, the debug log is here http://pastebin.freeswitch.org/14307. ?The
>> >> > > > relevant part of the configs is below.
>> >> > > > ? ?<extension name="conference_ivr">
>> >> > > > ? ? ?<condition field="destination_number" expression="^7000$">
>> >> > > > ? ? ? ?<action application="answer"/>
>> >> > > > <action application="sleep" data="500"/>
>> >> > > > <action application="set" data="api_hangup_hook=system sayconf -a local
>> >> > ?-a
>> >> > > > ${caller_id_number} -a hung -a up"/>
>> >> > > > ? ? ? ?<action application="ivr" data="conference_ivr"/>
>> >> > > > ? ? ?</condition>
>> >> > > > ? ?</extension>
>> >> > > > and the ivr just asks for passwords and send the call to an extension
>> >> > in
>> >> > > > context conferences.
>> >> > > >
>> >> > > > Hope this helps.
>> >> > > >
>> >> > > > Thanks.
>> >> > > >
>> >> > > > Michael Collins <msc at freeswitch.org> wrote:
>> >> > > >
>> >> > > > > Please supply the configs and a debug trace. My guess is that there's
>> >> > > > > probably just a simple config element that is wrong or missing.
>> >> > > > >
>> >> > > > > -MC
>> >> > > > >
>> >> > > > > On Tue, Oct 19, 2010 at 9:49 AM, <covici at ccs.covici.com> wrote:
>> >> > > > >
>> >> > > > > > Hi. ?I have an ivr which connects to conferences which I have in
>> >> > their
>> >> > > > > > own context rather than the default. ?The problem is that if I call
>> >> > > > > > someone else and have a 3-way between the other party and the
>> >> > > > > > conference, its fine till I hang up and then fs tries to transfer
>> >> > to
>> >> > > > the
>> >> > > > > > correct conference name, but uses default for the context instead.
>> >> > ?Is
>> >> > > > > > this expected behavior or should I file a bug?
>> >> > > > > >
>> >> > > > > > Thanks.
>> >> > > > > >
>> >> > > > > > --
>> >> > > > > > Your life is like a penny. ?You're going to lose it. ?The question
>> >> > is:
>> >> > > > > > How do
>> >> > > > > > you spend it?
>> >> > > > > >
>> >> > > > > > ? ? ? ? John Covici
>> >> > > > > > ? ? ? ? covici at ccs.covici.com
>> >> > > > > >
>> >> > > > > > _______________________________________________
>> >> > > > > > FreeSWITCH-users mailing list
>> >> > > > > > FreeSWITCH-users at lists.freeswitch.org
>> >> > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > > > > > UNSUBSCRIBE:
>> >> > > > http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > > > > > http://www.freeswitch.org
>> >> > > > > >
>> >> > > > >
>> >> > > > > ----------------------------------------------------
>> >> > > > > Alternatives:
>> >> > > > >
>> >> > > > > ----------------------------------------------------
>> >> > > > > _______________________________________________
>> >> > > > > FreeSWITCH-users mailing list
>> >> > > > > FreeSWITCH-users at lists.freeswitch.org
>> >> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > > > > UNSUBSCRIBE:
>> >> > http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > > > > http://www.freeswitch.org
>> >> > > >
>> >> > > > --
>> >> > > > Your life is like a penny. ?You're going to lose it. ?The question is:
>> >> > > > How do
>> >> > > > you spend it?
>> >> > > >
>> >> > > > ? ? ? ? John Covici
>> >> > > > ? ? ? ? covici at ccs.covici.com
>> >> > > >
>> >> > > > _______________________________________________
>> >> > > > FreeSWITCH-users mailing list
>> >> > > > FreeSWITCH-users at lists.freeswitch.org
>> >> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > > > UNSUBSCRIBE:
>> >> > http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > > > http://www.freeswitch.org
>> >> > > >
>> >> > >
>> >> > > ----------------------------------------------------
>> >> > > Alternatives:
>> >> > >
>> >> > > ----------------------------------------------------
>> >> > > _______________________________________________
>> >> > > FreeSWITCH-users mailing list
>> >> > > FreeSWITCH-users at lists.freeswitch.org
>> >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > > http://www.freeswitch.org
>> >> >
>> >> > --
>> >> > Your life is like a penny. ?You're going to lose it. ?The question is:
>> >> > How do
>> >> > you spend it?
>> >> >
>> >> > ? ? ? ? John Covici
>> >> > ? ? ? ? covici at ccs.covici.com
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >>
>> >> ----------------------------------------------------
>> >> Alternatives:
>> >>
>> >> ----------------------------------------------------
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> > --
>> > Your life is like a penny. ?You're going to lose it. ?The question is:
>> > How do
>> > you spend it?
>> >
>> > ? ? ? ? John Covici
>> > ? ? ? ? covici at ccs.covici.com
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> --
> Your life is like a penny. ?You're going to lose it. ?The question is:
> How do
> you spend it?
>
> ? ? ? ? John Covici
> ? ? ? ? covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From mustafa.pk at gmail.com  Tue Oct 26 13:44:17 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Wed, 27 Oct 2010 01:44:17 +0500
Subject: [Freeswitch-users] Is anyone familiar with Telemaque?
In-Reply-To: <4CC6A214.1080601@bksys.co.in>
References: <AANLkTimib+3T7L9sP28HN6BREJF5ca2j2nou-84YU=Ln@mail.gmail.com>
	<4CAAE5DE.6080407@telemaque.fr> <4CABF9D5.7000501@bksys.co.in>
	<4CAC4EB4.6080901@telemaque.fr> <4CC6A214.1080601@bksys.co.in>
Message-ID: <AANLkTimg-wDSH4U-XLV1up7Bp1t_UrbpRo9XdG8LqdrY@mail.gmail.com>

Hi,

I am not sure about how mod_fifo will behave if there is a high call
volume lets's say 300 calls in the queue, i am planning to use
mod_fifo with mod_conference for inbound call center scenario.

As you said freeswitch hangs if there are ~25 calls in the fifo, i
would like someone to advice me how scalable mod_fifo is? is it
capable of handling 300 calls (on quadcore xeon servers with 8g ram),
if not then what are other options to queue incoming calls in FS.

Currently we are using 2 vicidial (asterisk+astguiclient) server + 2
web and 3 node mysql-cluster (all Dell R710 series) servers setup to
handle 250+ incoming concurrent calls. though this setup works but i
am not happy with the way vicidial abuses system and database
resources. (with excuse to matt florel and other vicidial authors,
infact what they developed is competing big boys in the market and
works like a charm), i am sure they will improve and fix their
technical design with the time.

Best Regards,


On Tue, Oct 26, 2010 at 2:40 PM, abubacker <abubacker at bksystems.co.in> wrote:
> On Wednesday 06 October 2010 03:55 PM, Tristan Mah? wrote:
>
> Hi,
>
> The telco cluster is powered by IBM servers ( from dual Xeon/8gb ram to dual
> quad core/16gb ram depending on the node ) with FC13 and latest FS git (
> well almost always, as of today I'm a week late, but will upgrade soon again
> to benefit of the amazing work the dev do ).
>
> If you have more precise questions, don't hesitate, I'm sorry if I can't
> reveal all the details on how we're doing things,but I'm sure you can
> understand that...
>
> Regards,
>
> Tristan.
> Le 10/06/2010 06:23 AM, abubacker a ?crit?:
>
> On Tuesday 05 October 2010 02:16 PM, Tristan Mah? wrote:
>
> There's someone working for them on list :)
>
> Feel free to ask me questions Michael.
>
> Regards,
>
> Gled.
>
> Le 10/04/2010 08:39 PM, Michael Collins a ?crit?:
>
> Dear FreeSWITCH Community,
>
> Some of you may have stumbled upon this news item:
>
> http://blogs.oracle.com/mysql/2010/10/innovating_with_open_source_call_center_services.html
>
> Evidently there is a company in France (Telemaque) who is using MySQL +
> Kamailio + FreeSWITCH for some heavy duty call processing. If you are at all
> familiar with this company please let me know. I'd like to learn more about
> what they are doing. (Je ne parle pas boucoup de Francais :( )
>
> Thanks for your help,
> MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> "Feel free to ask me questions Michael"
>
> I think I must use this opportunity ,
> please specify the software ( OS ) and the hardwares required to handle the
> heavy duty call
> processing and also specify the FreeSWITCH version.
> I guess you could answer this very precisely.
>
> Thanks in Advance !
>
> --
> Best regards,
> N.Abubacker ,
> Associate system engineer ,
> bk systems pvt ltd ,
> Ph : 9144-43902701
>
> Disclaimer: http://www.bksystems.co.in/email-policy
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> Dear Tristan ,
> Knew this is an old post , but even though I wanted this to continue to get
> some more details ,
> what mod you are using? to handle more calls ? ( hope its mod_fifo )
> I am working in mod_fifo , some times it throws socketerror and makes the
> entire freeswitch hang
> when more than 25 calls in a queue.
>
> My question is can u give me some details about the queue and member
> configuration ( if possible )
>
> If you are not using mod_fifo , just tell me the general configuration
> required to fine tune the
> freeswitch.
>
> Thanks Again !
>
> --
> Best regards,
> N.Abubacker ,
> Associate system engineer ,
> bk systems pvt ltd ,
> Ph : 9144-43902701
>
> Disclaimer: http://www.bksystems.co.in/email-policy
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com


From mario_fs at mgtech.com  Tue Oct 26 14:16:35 2010
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 26 Oct 2010 14:16:35 -0700
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <AANLkTi=BiDiXLgEoj_MfwmZbGjvXoOuNJgUYdetciNCj@mail.gmail.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
	<407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>
	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
	<AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
	<6D9FD5E2-21FA-4630-8CF4-A87ADB30AC87@mgtech.com>
	<AANLkTi=BiDiXLgEoj_MfwmZbGjvXoOuNJgUYdetciNCj@mail.gmail.com>
Message-ID: <9BAFDAF8-1E66-444C-A024-A88FF65D8BCA@mgtech.com>

I would love to get static mapping to work but I could not. I  have 1 static public address and 1 dynamic both in the same router, this is the root of the problem. I tried to get FS to use the static ip but the router upnp only tells FS about the dynamic external ip. I tried setting external sip/rtp to just the ip (no autonat:) and ran with -nonat but I could not get incoming calls. BTW, I did change the expire-seconds to 30 in one gateway to test. Will keep this thread up-to-date for anyone who may be in the same boat someday. Thanks a lot for looking at the trace and pointing me in the right direction!
Mario

On Oct 26, 2010, at 1:22 PM, Anthony Minessale wrote:

> You should be setting the req freq to a low number on the outbound gateways
> The examples you showed had a series of inbound reg
> 
> also set expire-seconds to 30 in your gateway xml
> 
> 
> The problem is if you are not constantly sending traffic to the box
> the nat mapping will go away.
> 
> If you are in production you should be using a static ip with a static
> mapping, any trouble you are having is your own fault for playing with
> fire.  The best we can do is tell you how to keep it contained.
> 
> 
> 
> 
> On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com> wrote:
>> I made the change. I had no idea the settings for the inside phones effected nat for the outside sip accounts. I was looking into aggressive-nat- detection since the internal profile status always shows the right external static IP but the nat_ap status always shows the dynamic ip. Crossing fingers/etc since this problem is 85% of time (weeks!) into FS changeover. Thanks!
>> Mario
>> 
>> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
>> 
>>> add
>>> 
>>> <variable name="sip-force-expires" value="30"/>
>>> 
>>> to the <variables> section of your <user>
>>> 
>>> you have it at 600 and the nat mapping is timing out while the 600
>>> seconds is ticking away
>>> 
>>> 
>>> 
>>> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>> From the TSP:
>>>> "I have enabled the SIP trace on your account. We are not currently seeing
>>>> any registration attempts to your account within the last 15 minutes. Please
>>>> restart FreeSwitch so that registration attempts begin again. Thank you. ".
>>>> So FS is not getting past router.
>>>> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>>>> 
>>>> I ran the global trace during the problem and it is
>>>> at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
>>>> "accttwo", "acct3". The trace includes phones since it was global. I am
>>>> using:
>>>>     <param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>>>>     <param name="ext-sip-ip" value="autonat:my-static.ip"/>
>>>> I tried dumping nat and removing the autonat: above and using -nonat but
>>>> that did not work, registration proceeded but no calls inbound.
>>>> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>>>> 
>>>> Whoops, I am using an IP address for at least one gateway so that is not the
>>>> problem:
>>>> They look outbound to me and I am using dns for 2 and an IP for one so that
>>>> is not the issue. I was able to get FS to clear this up by doing "nat_map
>>>> reinit" which is why I think this is a nat problem. I will do the trace you
>>>> mentioned. I will plug an ip address into one of the gateways to see what
>>>> happens, they all fail at once. Thanks for responding!
>>>> Mario
>>>> 
>>>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>>>> 
>>>> I really need help on this as I have weeks into this problem. I thought I
>>>> had it nailed but I guess not. After 5.5 hours I get:
>>>> 
>>>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>>>> Registration, setting retry to 15 seconds.
>>>> 
>>>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>> 
>>>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>> 
>>>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
>>>> Registration, setting retry to 30 seconds.
>>>> 
>>>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>>>> Registration, setting retry to 15 seconds.
>>>> 
>>>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>>>> Registration, setting retry to 15 seconds.
>>>> 
>>>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>> 
>>>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>> 
>>>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>>>> 
>>>> and no way to make/get calls until I restart FS. I did this:
>>>> 
>>>> 1. log 7
>>>> 
>>>> 2. sofia profile xxxx siptrace on   for each profile/gateway
>>>> 
>>>> 3. restarted router
>>>> 
>>>> All three did not solve the problem. The trace and log produced no
>>>> additional lines which is why I am wondering if FS has a problem since the
>>>> trace shows no SIP activity.
>>>> 
>>>> 3 gateways with 2 ITSPs
>>>> 
>>>> 2 DSL/WAN lines, 1 static and 1 dynamic
>>>> 
>>>> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
>>>> external static ip.
>>>> 
>>>> sofia status profile ... has the right ext ip
>>>> 
>>>> nat_map status shows the dynamic (wrong) IP
>>>> 
>>>> I tried starting with -nonat but that was worse
>>>> 
>>>> the only way to fix is restart FS.
>>>> 
>>>> I read the wiki on external nat, auto_nat and everything else many times.
>>>> 
>>>> Thanks Mario
>>>> 
>>>> 
>>>> _______________________________________________
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> 
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> 
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> 
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> 
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> 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
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From covici at ccs.covici.com  Tue Oct 26 14:41:21 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 26 Oct 2010 17:41:21 -0400
Subject: [Freeswitch-users] transfer problem
In-Reply-To: <AANLkTimmgeivgE-+JbCQRcNvRGKhAE1mf9PUAWaJBb98@mail.gmail.com>
References: <13464.1287506974@ccs.covici.com>
	<AANLkTikh67cGuMn3v4D3QZ+AuQ0vWrLE_fRhL5+g1fg7@mail.gmail.com>
	<19713.1288004756@ccs.covici.com>
	<AANLkTinA4Egvfr1N-EvLun6rh-BR=WxwutHdOryACnYp@mail.gmail.com>
	<28001.1288041100@ccs.covici.com>
	<AANLkTinVAP-5A4KA7gz6hbRUVw7qh6Kc+ty48gs4eucZ@mail.gmail.com>
	<29557.1288051598@ccs.covici.com>
	<AANLkTi==CoAgm9_foaPEWDpBmME2HWq5hoskDNjE4MA6@mail.gmail.com>
	<12571.1288120881@ccs.covici.com>
	<AANLkTimmgeivgE-+JbCQRcNvRGKhAE1mf9PUAWaJBb98@mail.gmail.com>
Message-ID: <14013.1288129281@ccs.covici.com>

OK, here is the log to correspond with putting the export of the
variable http://pastebin.freeswitch.org/14330 .

Anthony Minessale <anthony.minessale at gmail.com> wrote:

> you need to learn to post logs to go with "it still doesn't work"
> every new change you make needs a new log.
> 
> 
> On Tue, Oct 26, 2010 at 2:21 PM,  <covici at ccs.covici.com> wrote:
> > HI. ?Well, Here is the extension in conferences context, where I did an
> > export of the variable, but it is still not working -- how do I get the
> > variable to be on the original channel in this case 7034754612 as well
> > as the 202?
> >
> > <extension name="utah">
> > <condition field="destination_number" expression="^utah$">
> > <action application="export" data="force_transfer_context=conferences"/>
> > <action application="system" data="sayconf -a local -a utah -a
> > ${caller_id_number} ?&amp;"/>
> > <action application="playback" data="conference/conf-muted.wav"/>
> > <action application="playback" data="conference/press_0_to_unmute.wav"/>
> > <action application="conference" data="utah at all"/>
> > </condition>
> > </extension>
> >
> > Anthony Minessale <anthony.minessale at gmail.com> wrote:
> >
> >> set the variable force_transfer_context to the desired context on all
> >> channels involved
> >>
> >> On Mon, Oct 25, 2010 at 7:06 PM, ?<covici at ccs.covici.com> wrote:
> >> > I can capture the log, but let me tell you the steps. ?202 does call
> >> > 7034754612, then it calls 7000, which is an ivr which transfers to an
> >> > extensionn in the context conferences, which in turn puts the caller in
> >> > a conference. ?So 202 calls 7034754612 and then is in this conference
> >> > whose context is not default. ?Now I flash the hook on 202, so
> >> > 7034754612 can hear the conference and also can hear 202. ?Now, what I
> >> > was hoping would happen is that I could hang up 202 and 4754612 would be
> >> > in the conference, but instead 4754612 hangs up and ?this seems to be
> >> > because the system tries to transfer 4754612 to theconference name but
> >> > in the default context and I get
> >> > 2010-10-25 17:09:46.137372 [DEBUG] mod_enum.c:204 Unable to lookup NAPTR
> >> > record for e164.org: valid domain but no data of requested type
> >> > and the 4754612 hangs up.
> >> >
> >> > I hope this clarifies what the problem is.
> >> >
> >> > Michael Collins <msc at freeswitch.org> wrote:
> >> >
> >> >> John,
> >> >>
> >> >> Perhaps I've missed something. Can you confirm exactly what steps you are
> >> >> taking? I don't want to make any assumptions. I can see that ext 202 calls
> >> >> out to 7034754612 and then 202 also makes a call to ext 7000. What I don't
> >> >> know is if that's a transfer or 3-way or what. Also, can you capture the log
> >> >> again, this time with a sip trace? Use "sofia global siptrace on".
> >> >>
> >> >> Thanks,
> >> >> MC
> >> >>
> >> >> On Mon, Oct 25, 2010 at 2:11 PM, <covici at ccs.covici.com> wrote:
> >> >>
> >> >> > Thanks for the correction -- however still same result. ?I did not think
> >> >> > it would matter in this case because the call did reach the phone and
> >> >> > everything after that is local. ?Question is why does it try to transfer
> >> >> > to the wrong context after the local extension hangs up?
> >> >> >
> >> >> > Michael Collins <msc at freeswitch.org> wrote:
> >> >> >
> >> >> > > John,
> >> >> > >
> >> >> > > I see two issues. At line #174 of your pastebin there is a message about
> >> >> > > "invalid gateway". I suspect that the gateway name should be "flowroute"
> >> >> > and
> >> >> > > not "fflowroute".
> >> >> > >
> >> >> > > At line #180 you have an invalid number format error. I would fix the
> >> >> > > invalid gateway first and retest as this second error may be a result of
> >> >> > the
> >> >> > > bad gateway.
> >> >> > >
> >> >> > > -MC
> >> >> > >
> >> >> > > On Mon, Oct 25, 2010 at 4:05 AM, <covici at ccs.covici.com> wrote:
> >> >> > >
> >> >> > > > OK, the debug log is here http://pastebin.freeswitch.org/14307. ?The
> >> >> > > > relevant part of the configs is below.
> >> >> > > > ? ?<extension name="conference_ivr">
> >> >> > > > ? ? ?<condition field="destination_number" expression="^7000$">
> >> >> > > > ? ? ? ?<action application="answer"/>
> >> >> > > > <action application="sleep" data="500"/>
> >> >> > > > <action application="set" data="api_hangup_hook=system sayconf -a local
> >> >> > ?-a
> >> >> > > > ${caller_id_number} -a hung -a up"/>
> >> >> > > > ? ? ? ?<action application="ivr" data="conference_ivr"/>
> >> >> > > > ? ? ?</condition>
> >> >> > > > ? ?</extension>
> >> >> > > > and the ivr just asks for passwords and send the call to an extension
> >> >> > in
> >> >> > > > context conferences.
> >> >> > > >
> >> >> > > > Hope this helps.
> >> >> > > >
> >> >> > > > Thanks.
> >> >> > > >
> >> >> > > > Michael Collins <msc at freeswitch.org> wrote:
> >> >> > > >
> >> >> > > > > Please supply the configs and a debug trace. My guess is that there's
> >> >> > > > > probably just a simple config element that is wrong or missing.
> >> >> > > > >
> >> >> > > > > -MC
> >> >> > > > >
> >> >> > > > > On Tue, Oct 19, 2010 at 9:49 AM, <covici at ccs.covici.com> wrote:
> >> >> > > > >
> >> >> > > > > > Hi. ?I have an ivr which connects to conferences which I have in
> >> >> > their
> >> >> > > > > > own context rather than the default. ?The problem is that if I call
> >> >> > > > > > someone else and have a 3-way between the other party and the
> >> >> > > > > > conference, its fine till I hang up and then fs tries to transfer
> >> >> > to
> >> >> > > > the
> >> >> > > > > > correct conference name, but uses default for the context instead.
> >> >> > ?Is
> >> >> > > > > > this expected behavior or should I file a bug?
> >> >> > > > > >
> >> >> > > > > > Thanks.
> >> >> > > > > >
> >> >> > > > > > --
> >> >> > > > > > Your life is like a penny. ?You're going to lose it. ?The question
> >> >> > is:
> >> >> > > > > > How do
> >> >> > > > > > you spend it?
> >> >> > > > > >
> >> >> > > > > > ? ? ? ? John Covici
> >> >> > > > > > ? ? ? ? covici at ccs.covici.com
> >> >> > > > > >
> >> >> > > > > > _______________________________________________
> >> >> > > > > > FreeSWITCH-users mailing list
> >> >> > > > > > FreeSWITCH-users at lists.freeswitch.org
> >> >> > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> > > > > > UNSUBSCRIBE:
> >> >> > > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > > > > > http://www.freeswitch.org
> >> >> > > > > >
> >> >> > > > >
> >> >> > > > > ----------------------------------------------------
> >> >> > > > > Alternatives:
> >> >> > > > >
> >> >> > > > > ----------------------------------------------------
> >> >> > > > > _______________________________________________
> >> >> > > > > FreeSWITCH-users mailing list
> >> >> > > > > FreeSWITCH-users at lists.freeswitch.org
> >> >> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> > > > > UNSUBSCRIBE:
> >> >> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > > > > http://www.freeswitch.org
> >> >> > > >
> >> >> > > > --
> >> >> > > > Your life is like a penny. ?You're going to lose it. ?The question is:
> >> >> > > > How do
> >> >> > > > you spend it?
> >> >> > > >
> >> >> > > > ? ? ? ? John Covici
> >> >> > > > ? ? ? ? covici at ccs.covici.com
> >> >> > > >
> >> >> > > > _______________________________________________
> >> >> > > > FreeSWITCH-users mailing list
> >> >> > > > FreeSWITCH-users at lists.freeswitch.org
> >> >> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> > > > UNSUBSCRIBE:
> >> >> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > > > http://www.freeswitch.org
> >> >> > > >
> >> >> > >
> >> >> > > ----------------------------------------------------
> >> >> > > Alternatives:
> >> >> > >
> >> >> > > ----------------------------------------------------
> >> >> > > _______________________________________________
> >> >> > > FreeSWITCH-users mailing list
> >> >> > > FreeSWITCH-users at lists.freeswitch.org
> >> >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > > http://www.freeswitch.org
> >> >> >
> >> >> > --
> >> >> > Your life is like a penny. ?You're going to lose it. ?The question is:
> >> >> > How do
> >> >> > you spend it?
> >> >> >
> >> >> > ? ? ? ? John Covici
> >> >> > ? ? ? ? covici at ccs.covici.com
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >>
> >> >> ----------------------------------------------------
> >> >> Alternatives:
> >> >>
> >> >> ----------------------------------------------------
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> > --
> >> > Your life is like a penny. ?You're going to lose it. ?The question is:
> >> > How do
> >> > you spend it?
> >> >
> >> > ? ? ? ? John Covici
> >> > ? ? ? ? covici at ccs.covici.com
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > 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
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > --
> > Your life is like a penny. ?You're going to lose it. ?The question is:
> > How do
> > you spend it?
> >
> > ? ? ? ? John Covici
> > ? ? ? ? covici at ccs.covici.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From bmcdaniel at teradon.com  Tue Oct 26 13:41:51 2010
From: bmcdaniel at teradon.com (Brian McDaniel)
Date: Tue, 26 Oct 2010 14:41:51 -0600
Subject: [Freeswitch-users] Caller-Id Gateways
Message-ID: <C4B187DE-AE5D-4BCE-A248-65A1C54C7F87@teradon.com>

Freeswtich rejects calls based on the format of the caller-id.  I've tried using both an SPA3102 and a Grandstream HT503 gateways and have seen the same issue on both.  Basically freeswitch rejects certain calls if the caller-id is not of the form Name, number, i.e. if it's ANNE, MARIEORTIZ, xxxxx it rejects the call but if the caller-id is ANNE MARIEORTIZ, xxxxxx it's accepts the call and passes the caller id accordingly.  It appears as though freeswitch has issue with the extra comma or that it expects a number instead of text, either way it seems that the call should not be rejected if anything leave the caller-id field blank or put unknown but don't reject the call.  I've seen numerous complaints about this on the web and was wondering if there is a fix to this issue or a workaround.  

From mcampbellsmith at gmail.com  Tue Oct 26 14:21:42 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 27 Oct 2010 08:21:42 +1100
Subject: [Freeswitch-users] FS not detecting public IP address change
In-Reply-To: <AF5D2E2E-F8A7-44AB-8EA0-D240A28A51F6@mgtech.com>
References: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>
	<AANLkTinEoTcLmUVcpras8-m8pdkfpE=_r0spz1S50iMF@mail.gmail.com>
	<AANLkTi=MoV27FW0Ukd6K+mqmgnkyNgb9hjFx-vBYV1ak@mail.gmail.com>
	<AANLkTimCRfKSWG+5OdQY0=HTt2kOqxwk+0vKvaBNdmAd@mail.gmail.com>
	<AANLkTikfpRi9Shxiqd2w5nz3ZHsBnrHTt+=rHHnvoDJn@mail.gmail.com>
	<AANLkTikH3JJCCpL5s_eF2qjjttnEetE0i=nPn87zhBUL@mail.gmail.com>
	<AANLkTimzkb7qY8nW3M2afTPn18=OxY6jQye8D9TKr9Ts@mail.gmail.com>
	<AANLkTimR2x5AQPNiDDt4fZ8F=z_Wep3TBVR9Lob_72OA@mail.gmail.com>
	<AANLkTi=JgCME8u=27kUyLGU3cMP2nZpFpu7Zd5-4tpj0@mail.gmail.com>
	<AANLkTikL=kvbfLQx3E-vCaErRM-ARnU3_YSAvKo67Uvy@mail.gmail.com>
	<AF5D2E2E-F8A7-44AB-8EA0-D240A28A51F6@mgtech.com>
Message-ID: <AANLkTikkFaJ-EWipGij2UAzMpmH8JGyhj+bUNKSDS8T5@mail.gmail.com>

I have it set to debug in switch.conf.xml and enable fsctl loglevel 7 in
fs_cli....

:~$ cat /usr/local/freeswitch/conf/autoload_configs/switch.conf.xml | egrep
"param.*loglevel"
    <param name="loglevel" value="debug"/>

With regards to sofia: how does it determine if the external IP has changed?
 And when it does, does sofia initiate a reregister of all external sip
providers?

Thanks!

On Wed, Oct 27, 2010 at 3:14 AM, Mario G <mario_fs at mgtech.com> wrote:

> What you said: "And nat_map status shows me which ports have been forwarded
> and the external ip address (which is sometimes wrong).  I also see this in
> the router gui." is also happening to me on a Linksys/Cisco RV042 router.
> Can't keep FS working for more than 2-5 hours without doing a restart or
> nat_map reinit. I think the routers upnp has a problem with dual wans, I
> have one static and one dynamic DSL.
>
> On Oct 26, 2010, at 6:43 AM, Rupa Schomaker wrote:
>
> What do you have set for loglevel in switch.conf.xml?  My guess is that you
> have it set to something other than debug.  If that is set to (say) info,
> then info is as low as you can go.  Setting the console log level to debug
> will not give you debug output.
>
> sofia is unaware of upnp so a keepalive packet should have no impact on
> it's operation.  I'm not familiar enough with sofia to tell what triggers
> that event.
>
> On Tue, Oct 26, 2010 at 6:24 AM, Mark Campbell-Smith <
> mcampbellsmith at gmail.com> wrote:
>
>> OK..  tcpdump gave me this and I still do not see anything on FS except
>> for this line:
>>
>> 2010-10-26 22:02:51.235608 [DEBUG] sofia.c:957 nua_i_outbound: unknown
>> event 8: 102 NAT binding changed
>>
>> Hypertext Transfer Protocol
>>     NOTIFY * HTTP/1.1\r\n
>>     HOST:239.255.255.250:1900\r\n
>>     Cache-Control:max-age=120\r\n
>>     Location:http://192.168.1.1:1278/rootDesc.xml\r\n<http://192.168.1.1:1278/rootDesc.xml%5Cr%5Cn>
>>     Server: Tomato UPnP/1.0 MiniUPnPd/1.4\r\n
>>     NT:urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>>
>>  USN:uuid:882dbe81-c28a-4992-baf7-410c05f1caf4::urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>>     NTS:ssdp:alive\r\n
>>     \r\n
>>
>> This looks the same as you posted before, so why do I get the unknown
>> event 8 (is this related)?
>>
>> Thanks!
>>
>> On Tue, Oct 26, 2010 at 3:32 PM, Mark Campbell-Smith <
>> mcampbellsmith at gmail.com> wrote:
>>
>>> :)  that was the first thing I did, but I get nothing; no keep alive
>>> packets.
>>>
>>> in the logfile at startup I see:
>>>
>>> 2010-10-26 10:19:45.484258 [INFO] switch_nat.c:410 Scanning for NAT
>>> 2010-10-26 10:19:45.485991 [DEBUG] switch_nat.c:167 Checking for PMP 1/5
>>> 2010-10-26 10:19:45.488575 [INFO] switch_nat.c:423 NAT detected type:
>>> pmp, ExtIP: '203.xxx.xxx.xxx'
>>> 2010-10-26 10:19:45.490683 [DEBUG] switch_nat.c:256 NAT thread configured
>>> 2010-10-26 10:19:45.492180 [DEBUG] switch_nat.c:267 NAT thread started
>>>
>>> And nat_map status shows me which ports have been forwarded and the
>>> external ip address (which is sometimes wrong).  I also see this in the
>>> router gui.
>>>
>>> But I never see the keep alive packets.
>>>
>>> On Tue, Oct 26, 2010 at 3:13 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>>> Just turn up console logging to level 7.
>>>>
>>>> /log 7 if using fs_cli
>>>>
>>>>
>>>> On Mon, Oct 25, 2010 at 6:53 PM, Mark Campbell-Smith <
>>>> mcampbellsmith at gmail.com> wrote:
>>>>
>>>>> Hi!
>>>>>
>>>>> How do I enable debugging to get the UPnP keep alive packets?
>>>>>
>>>>> Thanks!
>>>>>
>>>>> On Tue, Oct 26, 2010 at 1:15 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>>>
>>>>>> I need the debug logging from the upnp stuff.  It should look
>>>>>> something like:
>>>>>>
>>>>>> +OK log level 7 [7]
>>>>>> freeswitch at internal> 2010-10-25 09:13:59.926861 [DEBUG]
>>>>>> switch_nat.c:299 got UPnP keep alive packet:
>>>>>> NOTIFY * HTTP/1.1
>>>>>> HOST:239.255.255.250:1900
>>>>>> Cache-Control:max-age=60
>>>>>> Location:http://192.168.1.1:5000/rootDesc.xml
>>>>>> Server: Tomato UPnP/1.0 MiniUPnPd/1.4
>>>>>> NT:urn:schemas-upnp-org:service:WANIPConnection:1
>>>>>>
>>>>>> USN:uuid:00000000-0000-0000-0000-000000000000::urn:schemas-upnp-org:service:WANIPConnection:1
>>>>>> NTS:ssdp:alive
>>>>>>
>>>>>> I also run tomato and haven't had trouble with the upnp support so at
>>>>>> least we have that part working.
>>>>>>
>>>>>> You should see the above keepalive appear periodically every 30s or
>>>>>> so.  You should see another set of messages when you terminate the DSL
>>>>>> connection and when that dsl connection comes back online.
>>>>>>
>>>>>> On Sat, Oct 23, 2010 at 5:36 PM, Mark Campbell-Smith <
>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>
>>>>>>> Hi!
>>>>>>>
>>>>>>> The router is a ASUS router but it is running the tomato firmware
>>>>>>> which runs miniupnd.
>>>>>>>
>>>>>>> In the debug I see the message: 2010-10-24 08:02:59.178918 [DEBUG]
>>>>>>> sofia.c:956 nua_i_outbound: unknown event 8: 101 NAT detected
>>>>>>>
>>>>>>> This is what I see below. I hope I enabled all debug messages.  I
>>>>>>> thought I saw some xml type messages earlier, but not sure how I enabled
>>>>>>> them.
>>>>>>>
>>>>>>> nta_outgoing: RTT is 67.958 ms
>>>>>>> outbound(0xb6e07c00): NAT binding changed: [210.xx.xxx.xx]:5080 !=
>>>>>>> [58.xxx.xx.xx]:5080
>>>>>>> nua(0xb6e07c00): event i_outbound 102 NAT binding changed
>>>>>>> nua: nua_application_event: entering
>>>>>>> 2010-10-24 09:05:08.849525 [DEBUG] sofia.c:956 nua_i_outbound:
>>>>>>> unknown event 8: 102 NAT binding changed
>>>>>>>
>>>>>>>
>>>>>>> On Sun, Oct 24, 2010 at 1:08 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>
>>>>>>>> btw: what router are you using?
>>>>>>>>
>>>>>>>>
>>>>>>>> On Sat, Oct 23, 2010 at 9:06 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>
>>>>>>>>> In a upnp config, FS is depending on the router to notify it of the
>>>>>>>>> ip address change.  It is possible the upnp message(s) aren't formatted as
>>>>>>>>> expected.  If you turn on debug logging you'll get the upnp messages on the
>>>>>>>>> console.  If you can pastebin the capture of those while dropping and
>>>>>>>>> restarting the dsl connection I can maybe see if there is something obvious
>>>>>>>>> going on.  The debug should also have messages related to the processing of
>>>>>>>>> those upnp messages.
>>>>>>>>>
>>>>>>>>> On Sat, Oct 23, 2010 at 7:04 AM, Mark Campbell-Smith <
>>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> HI!
>>>>>>>>>>
>>>>>>>>>> I know this question must have been answered 100's of times....
>>>>>>>>>>
>>>>>>>>>> My adsl is a bit dodgy at the moment and tends to go up and down;
>>>>>>>>>> which means I get assigned a new ip address from my ISP.  FS is nat'd behind
>>>>>>>>>> a upnp capable router.
>>>>>>>>>>
>>>>>>>>>> The problem is that FS does not seem to be detecting the change.
>>>>>>>>>> For both the internal and external profile, I have auto-nat setup, and I
>>>>>>>>>> have tried stun and host settings in vars.conf.  When I issue a nat_map
>>>>>>>>>> status I see my old IP address, and if I then issue a nat_map reinit, I see
>>>>>>>>>> the new public IP address.  How can I get this to be automatic?
>>>>>>>>>>
>>>>>>>>>> Also the public IP address shown in the nat_map status is not the
>>>>>>>>>> same as shown in the sofia profile internal/external printouts.  Why is
>>>>>>>>>> this?
>>>>>>>>>>
>>>>>>>>>> I'm sure this is configuration, just not sure what to change.
>>>>>>>>>>
>>>>>>>>>> Thanks!
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>> -Rupa
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> -Rupa
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> -Rupa
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rupa at rupa.com  Tue Oct 26 17:41:49 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 26 Oct 2010 19:41:49 -0500
Subject: [Freeswitch-users] FS not detecting public IP address change
In-Reply-To: <AANLkTikkFaJ-EWipGij2UAzMpmH8JGyhj+bUNKSDS8T5@mail.gmail.com>
References: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>
	<AANLkTinEoTcLmUVcpras8-m8pdkfpE=_r0spz1S50iMF@mail.gmail.com>
	<AANLkTi=MoV27FW0Ukd6K+mqmgnkyNgb9hjFx-vBYV1ak@mail.gmail.com>
	<AANLkTimCRfKSWG+5OdQY0=HTt2kOqxwk+0vKvaBNdmAd@mail.gmail.com>
	<AANLkTikfpRi9Shxiqd2w5nz3ZHsBnrHTt+=rHHnvoDJn@mail.gmail.com>
	<AANLkTikH3JJCCpL5s_eF2qjjttnEetE0i=nPn87zhBUL@mail.gmail.com>
	<AANLkTimzkb7qY8nW3M2afTPn18=OxY6jQye8D9TKr9Ts@mail.gmail.com>
	<AANLkTimR2x5AQPNiDDt4fZ8F=z_Wep3TBVR9Lob_72OA@mail.gmail.com>
	<AANLkTi=JgCME8u=27kUyLGU3cMP2nZpFpu7Zd5-4tpj0@mail.gmail.com>
	<AANLkTikL=kvbfLQx3E-vCaErRM-ARnU3_YSAvKo67Uvy@mail.gmail.com>
	<AF5D2E2E-F8A7-44AB-8EA0-D240A28A51F6@mgtech.com>
	<AANLkTikkFaJ-EWipGij2UAzMpmH8JGyhj+bUNKSDS8T5@mail.gmail.com>
Message-ID: <AANLkTin=hRwNSBLxTRdJ6SqcPedbHiDgAJPCFESM9nCT@mail.gmail.com>

Do you see ANY debug messages on your console from any other FS module?
 fsctl loglevel 7 sets the base log level, but not necessarily the log level
on your screen.

/log 7

sets the log level in fs_cli.

Your screen should get VERY busy especially when processing a call.

On Tue, Oct 26, 2010 at 4:21 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> I have it set to debug in switch.conf.xml and enable fsctl loglevel 7 in
> fs_cli....
>
> :~$ cat /usr/local/freeswitch/conf/autoload_configs/switch.conf.xml | egrep
> "param.*loglevel"
>     <param name="loglevel" value="debug"/>
>
> With regards to sofia: how does it determine if the external IP has
> changed?  And when it does, does sofia initiate a reregister of all external
> sip providers?
>
> Thanks!
>
> On Wed, Oct 27, 2010 at 3:14 AM, Mario G <mario_fs at mgtech.com> wrote:
>
>> What you said: "And nat_map status shows me which ports have been
>> forwarded and the external ip address (which is sometimes wrong).  I also
>> see this in the router gui." is also happening to me on a Linksys/Cisco
>> RV042 router. Can't keep FS working for more than 2-5 hours without doing a
>> restart or nat_map reinit. I think the routers upnp has a problem with dual
>> wans, I have one static and one dynamic DSL.
>>
>> On Oct 26, 2010, at 6:43 AM, Rupa Schomaker wrote:
>>
>> What do you have set for loglevel in switch.conf.xml?  My guess is that
>> you have it set to something other than debug.  If that is set to (say)
>> info, then info is as low as you can go.  Setting the console log level to
>> debug will not give you debug output.
>>
>> sofia is unaware of upnp so a keepalive packet should have no impact on
>> it's operation.  I'm not familiar enough with sofia to tell what triggers
>> that event.
>>
>> On Tue, Oct 26, 2010 at 6:24 AM, Mark Campbell-Smith <
>> mcampbellsmith at gmail.com> wrote:
>>
>>> OK..  tcpdump gave me this and I still do not see anything on FS except
>>> for this line:
>>>
>>> 2010-10-26 22:02:51.235608 [DEBUG] sofia.c:957 nua_i_outbound: unknown
>>> event 8: 102 NAT binding changed
>>>
>>> Hypertext Transfer Protocol
>>>     NOTIFY * HTTP/1.1\r\n
>>>     HOST:239.255.255.250:1900\r\n
>>>     Cache-Control:max-age=120\r\n
>>>     Location:http://192.168.1.1:1278/rootDesc.xml\r\n<http://192.168.1.1:1278/rootDesc.xml%5Cr%5Cn>
>>>     Server: Tomato UPnP/1.0 MiniUPnPd/1.4\r\n
>>>     NT:urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>>>
>>>  USN:uuid:882dbe81-c28a-4992-baf7-410c05f1caf4::urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>>>     NTS:ssdp:alive\r\n
>>>     \r\n
>>>
>>> This looks the same as you posted before, so why do I get the unknown
>>> event 8 (is this related)?
>>>
>>> Thanks!
>>>
>>> On Tue, Oct 26, 2010 at 3:32 PM, Mark Campbell-Smith <
>>> mcampbellsmith at gmail.com> wrote:
>>>
>>>> :)  that was the first thing I did, but I get nothing; no keep alive
>>>> packets.
>>>>
>>>> in the logfile at startup I see:
>>>>
>>>> 2010-10-26 10:19:45.484258 [INFO] switch_nat.c:410 Scanning for NAT
>>>> 2010-10-26 10:19:45.485991 [DEBUG] switch_nat.c:167 Checking for PMP 1/5
>>>> 2010-10-26 10:19:45.488575 [INFO] switch_nat.c:423 NAT detected type:
>>>> pmp, ExtIP: '203.xxx.xxx.xxx'
>>>> 2010-10-26 10:19:45.490683 [DEBUG] switch_nat.c:256 NAT thread
>>>> configured
>>>> 2010-10-26 10:19:45.492180 [DEBUG] switch_nat.c:267 NAT thread started
>>>>
>>>> And nat_map status shows me which ports have been forwarded and the
>>>> external ip address (which is sometimes wrong).  I also see this in the
>>>> router gui.
>>>>
>>>> But I never see the keep alive packets.
>>>>
>>>> On Tue, Oct 26, 2010 at 3:13 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>>
>>>>> Just turn up console logging to level 7.
>>>>>
>>>>> /log 7 if using fs_cli
>>>>>
>>>>>
>>>>> On Mon, Oct 25, 2010 at 6:53 PM, Mark Campbell-Smith <
>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>
>>>>>> Hi!
>>>>>>
>>>>>> How do I enable debugging to get the UPnP keep alive packets?
>>>>>>
>>>>>> Thanks!
>>>>>>
>>>>>> On Tue, Oct 26, 2010 at 1:15 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>
>>>>>>> I need the debug logging from the upnp stuff.  It should look
>>>>>>> something like:
>>>>>>>
>>>>>>> +OK log level 7 [7]
>>>>>>> freeswitch at internal> 2010-10-25 09:13:59.926861 [DEBUG]
>>>>>>> switch_nat.c:299 got UPnP keep alive packet:
>>>>>>> NOTIFY * HTTP/1.1
>>>>>>> HOST:239.255.255.250:1900
>>>>>>> Cache-Control:max-age=60
>>>>>>> Location:http://192.168.1.1:5000/rootDesc.xml
>>>>>>> Server: Tomato UPnP/1.0 MiniUPnPd/1.4
>>>>>>> NT:urn:schemas-upnp-org:service:WANIPConnection:1
>>>>>>>
>>>>>>> USN:uuid:00000000-0000-0000-0000-000000000000::urn:schemas-upnp-org:service:WANIPConnection:1
>>>>>>> NTS:ssdp:alive
>>>>>>>
>>>>>>> I also run tomato and haven't had trouble with the upnp support so at
>>>>>>> least we have that part working.
>>>>>>>
>>>>>>> You should see the above keepalive appear periodically every 30s or
>>>>>>> so.  You should see another set of messages when you terminate the DSL
>>>>>>> connection and when that dsl connection comes back online.
>>>>>>>
>>>>>>> On Sat, Oct 23, 2010 at 5:36 PM, Mark Campbell-Smith <
>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>
>>>>>>>> Hi!
>>>>>>>>
>>>>>>>> The router is a ASUS router but it is running the tomato firmware
>>>>>>>> which runs miniupnd.
>>>>>>>>
>>>>>>>> In the debug I see the message: 2010-10-24 08:02:59.178918 [DEBUG]
>>>>>>>> sofia.c:956 nua_i_outbound: unknown event 8: 101 NAT detected
>>>>>>>>
>>>>>>>> This is what I see below. I hope I enabled all debug messages.  I
>>>>>>>> thought I saw some xml type messages earlier, but not sure how I enabled
>>>>>>>> them.
>>>>>>>>
>>>>>>>> nta_outgoing: RTT is 67.958 ms
>>>>>>>> outbound(0xb6e07c00): NAT binding changed: [210.xx.xxx.xx]:5080 !=
>>>>>>>> [58.xxx.xx.xx]:5080
>>>>>>>> nua(0xb6e07c00): event i_outbound 102 NAT binding changed
>>>>>>>> nua: nua_application_event: entering
>>>>>>>> 2010-10-24 09:05:08.849525 [DEBUG] sofia.c:956 nua_i_outbound:
>>>>>>>> unknown event 8: 102 NAT binding changed
>>>>>>>>
>>>>>>>>
>>>>>>>> On Sun, Oct 24, 2010 at 1:08 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>
>>>>>>>>> btw: what router are you using?
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Sat, Oct 23, 2010 at 9:06 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>>
>>>>>>>>>> In a upnp config, FS is depending on the router to notify it of
>>>>>>>>>> the ip address change.  It is possible the upnp message(s) aren't formatted
>>>>>>>>>> as expected.  If you turn on debug logging you'll get the upnp messages on
>>>>>>>>>> the console.  If you can pastebin the capture of those while dropping and
>>>>>>>>>> restarting the dsl connection I can maybe see if there is something obvious
>>>>>>>>>> going on.  The debug should also have messages related to the processing of
>>>>>>>>>> those upnp messages.
>>>>>>>>>>
>>>>>>>>>> On Sat, Oct 23, 2010 at 7:04 AM, Mark Campbell-Smith <
>>>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> HI!
>>>>>>>>>>>
>>>>>>>>>>> I know this question must have been answered 100's of times....
>>>>>>>>>>>
>>>>>>>>>>> My adsl is a bit dodgy at the moment and tends to go up and down;
>>>>>>>>>>> which means I get assigned a new ip address from my ISP.  FS is nat'd behind
>>>>>>>>>>> a upnp capable router.
>>>>>>>>>>>
>>>>>>>>>>> The problem is that FS does not seem to be detecting the change.
>>>>>>>>>>>   For both the internal and external profile, I have auto-nat setup, and I
>>>>>>>>>>> have tried stun and host settings in vars.conf.  When I issue a nat_map
>>>>>>>>>>> status I see my old IP address, and if I then issue a nat_map reinit, I see
>>>>>>>>>>> the new public IP address.  How can I get this to be automatic?
>>>>>>>>>>>
>>>>>>>>>>> Also the public IP address shown in the nat_map status is not the
>>>>>>>>>>> same as shown in the sofia profile internal/external printouts.  Why is
>>>>>>>>>>> this?
>>>>>>>>>>>
>>>>>>>>>>> I'm sure this is configuration, just not sure what to change.
>>>>>>>>>>>
>>>>>>>>>>> Thanks!
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> --
>>>>>>>>>> -Rupa
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>> -Rupa
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> -Rupa
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> -Rupa
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>  _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From mcampbellsmith at gmail.com  Tue Oct 26 18:09:19 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 27 Oct 2010 12:09:19 +1100
Subject: [Freeswitch-users] FS not detecting public IP address change
In-Reply-To: <AANLkTin=hRwNSBLxTRdJ6SqcPedbHiDgAJPCFESM9nCT@mail.gmail.com>
References: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>
	<AANLkTinEoTcLmUVcpras8-m8pdkfpE=_r0spz1S50iMF@mail.gmail.com>
	<AANLkTi=MoV27FW0Ukd6K+mqmgnkyNgb9hjFx-vBYV1ak@mail.gmail.com>
	<AANLkTimCRfKSWG+5OdQY0=HTt2kOqxwk+0vKvaBNdmAd@mail.gmail.com>
	<AANLkTikfpRi9Shxiqd2w5nz3ZHsBnrHTt+=rHHnvoDJn@mail.gmail.com>
	<AANLkTikH3JJCCpL5s_eF2qjjttnEetE0i=nPn87zhBUL@mail.gmail.com>
	<AANLkTimzkb7qY8nW3M2afTPn18=OxY6jQye8D9TKr9Ts@mail.gmail.com>
	<AANLkTimR2x5AQPNiDDt4fZ8F=z_Wep3TBVR9Lob_72OA@mail.gmail.com>
	<AANLkTi=JgCME8u=27kUyLGU3cMP2nZpFpu7Zd5-4tpj0@mail.gmail.com>
	<AANLkTikL=kvbfLQx3E-vCaErRM-ARnU3_YSAvKo67Uvy@mail.gmail.com>
	<AF5D2E2E-F8A7-44AB-8EA0-D240A28A51F6@mgtech.com>
	<AANLkTikkFaJ-EWipGij2UAzMpmH8JGyhj+bUNKSDS8T5@mail.gmail.com>
	<AANLkTin=hRwNSBLxTRdJ6SqcPedbHiDgAJPCFESM9nCT@mail.gmail.com>
Message-ID: <AANLkTi=+ggi1L7oAhKGFFR8f=QaXhoj_fMGPJ1pUFk+v@mail.gmail.com>

I see debug from sofia.c, switch_core_state_machine, mod_sofia, diaplan
parsing, sofia_glue, switch_rtp, switch_ivr_play_say etc.



On Wed, Oct 27, 2010 at 11:41 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> Do you see ANY debug messages on your console from any other FS module?
>  fsctl loglevel 7 sets the base log level, but not necessarily the log level
> on your screen.
>
> /log 7
>
> sets the log level in fs_cli.
>
> Your screen should get VERY busy especially when processing a call.
>
> On Tue, Oct 26, 2010 at 4:21 PM, Mark Campbell-Smith <
> mcampbellsmith at gmail.com> wrote:
>
>> I have it set to debug in switch.conf.xml and enable fsctl loglevel 7 in
>> fs_cli....
>>
>> :~$ cat /usr/local/freeswitch/conf/autoload_configs/switch.conf.xml |
>> egrep "param.*loglevel"
>>     <param name="loglevel" value="debug"/>
>>
>> With regards to sofia: how does it determine if the external IP has
>> changed?  And when it does, does sofia initiate a reregister of all external
>> sip providers?
>>
>> Thanks!
>>
>> On Wed, Oct 27, 2010 at 3:14 AM, Mario G <mario_fs at mgtech.com> wrote:
>>
>>> What you said: "And nat_map status shows me which ports have been
>>> forwarded and the external ip address (which is sometimes wrong).  I also
>>> see this in the router gui." is also happening to me on a Linksys/Cisco
>>> RV042 router. Can't keep FS working for more than 2-5 hours without doing a
>>> restart or nat_map reinit. I think the routers upnp has a problem with dual
>>> wans, I have one static and one dynamic DSL.
>>>
>>> On Oct 26, 2010, at 6:43 AM, Rupa Schomaker wrote:
>>>
>>> What do you have set for loglevel in switch.conf.xml?  My guess is that
>>> you have it set to something other than debug.  If that is set to (say)
>>> info, then info is as low as you can go.  Setting the console log level to
>>> debug will not give you debug output.
>>>
>>> sofia is unaware of upnp so a keepalive packet should have no impact on
>>> it's operation.  I'm not familiar enough with sofia to tell what triggers
>>> that event.
>>>
>>> On Tue, Oct 26, 2010 at 6:24 AM, Mark Campbell-Smith <
>>> mcampbellsmith at gmail.com> wrote:
>>>
>>>> OK..  tcpdump gave me this and I still do not see anything on FS except
>>>> for this line:
>>>>
>>>> 2010-10-26 22:02:51.235608 [DEBUG] sofia.c:957 nua_i_outbound: unknown
>>>> event 8: 102 NAT binding changed
>>>>
>>>> Hypertext Transfer Protocol
>>>>     NOTIFY * HTTP/1.1\r\n
>>>>     HOST:239.255.255.250:1900\r\n
>>>>     Cache-Control:max-age=120\r\n
>>>>     Location:http://192.168.1.1:1278/rootDesc.xml\r\n<http://192.168.1.1:1278/rootDesc.xml%5Cr%5Cn>
>>>>     Server: Tomato UPnP/1.0 MiniUPnPd/1.4\r\n
>>>>     NT:urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>>>>
>>>>  USN:uuid:882dbe81-c28a-4992-baf7-410c05f1caf4::urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>>>>     NTS:ssdp:alive\r\n
>>>>     \r\n
>>>>
>>>> This looks the same as you posted before, so why do I get the unknown
>>>> event 8 (is this related)?
>>>>
>>>> Thanks!
>>>>
>>>> On Tue, Oct 26, 2010 at 3:32 PM, Mark Campbell-Smith <
>>>> mcampbellsmith at gmail.com> wrote:
>>>>
>>>>> :)  that was the first thing I did, but I get nothing; no keep alive
>>>>> packets.
>>>>>
>>>>> in the logfile at startup I see:
>>>>>
>>>>> 2010-10-26 10:19:45.484258 [INFO] switch_nat.c:410 Scanning for NAT
>>>>> 2010-10-26 10:19:45.485991 [DEBUG] switch_nat.c:167 Checking for PMP
>>>>> 1/5
>>>>> 2010-10-26 10:19:45.488575 [INFO] switch_nat.c:423 NAT detected type:
>>>>> pmp, ExtIP: '203.xxx.xxx.xxx'
>>>>> 2010-10-26 10:19:45.490683 [DEBUG] switch_nat.c:256 NAT thread
>>>>> configured
>>>>> 2010-10-26 10:19:45.492180 [DEBUG] switch_nat.c:267 NAT thread started
>>>>>
>>>>> And nat_map status shows me which ports have been forwarded and the
>>>>> external ip address (which is sometimes wrong).  I also see this in the
>>>>> router gui.
>>>>>
>>>>> But I never see the keep alive packets.
>>>>>
>>>>> On Tue, Oct 26, 2010 at 3:13 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>>>
>>>>>> Just turn up console logging to level 7.
>>>>>>
>>>>>> /log 7 if using fs_cli
>>>>>>
>>>>>>
>>>>>> On Mon, Oct 25, 2010 at 6:53 PM, Mark Campbell-Smith <
>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>
>>>>>>> Hi!
>>>>>>>
>>>>>>> How do I enable debugging to get the UPnP keep alive packets?
>>>>>>>
>>>>>>> Thanks!
>>>>>>>
>>>>>>> On Tue, Oct 26, 2010 at 1:15 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>
>>>>>>>> I need the debug logging from the upnp stuff.  It should look
>>>>>>>> something like:
>>>>>>>>
>>>>>>>> +OK log level 7 [7]
>>>>>>>> freeswitch at internal> 2010-10-25 09:13:59.926861 [DEBUG]
>>>>>>>> switch_nat.c:299 got UPnP keep alive packet:
>>>>>>>> NOTIFY * HTTP/1.1
>>>>>>>> HOST:239.255.255.250:1900
>>>>>>>> Cache-Control:max-age=60
>>>>>>>> Location:http://192.168.1.1:5000/rootDesc.xml
>>>>>>>> Server: Tomato UPnP/1.0 MiniUPnPd/1.4
>>>>>>>> NT:urn:schemas-upnp-org:service:WANIPConnection:1
>>>>>>>>
>>>>>>>> USN:uuid:00000000-0000-0000-0000-000000000000::urn:schemas-upnp-org:service:WANIPConnection:1
>>>>>>>> NTS:ssdp:alive
>>>>>>>>
>>>>>>>> I also run tomato and haven't had trouble with the upnp support so
>>>>>>>> at least we have that part working.
>>>>>>>>
>>>>>>>> You should see the above keepalive appear periodically every 30s or
>>>>>>>> so.  You should see another set of messages when you terminate the DSL
>>>>>>>> connection and when that dsl connection comes back online.
>>>>>>>>
>>>>>>>> On Sat, Oct 23, 2010 at 5:36 PM, Mark Campbell-Smith <
>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Hi!
>>>>>>>>>
>>>>>>>>> The router is a ASUS router but it is running the tomato firmware
>>>>>>>>> which runs miniupnd.
>>>>>>>>>
>>>>>>>>> In the debug I see the message: 2010-10-24 08:02:59.178918 [DEBUG]
>>>>>>>>> sofia.c:956 nua_i_outbound: unknown event 8: 101 NAT detected
>>>>>>>>>
>>>>>>>>> This is what I see below. I hope I enabled all debug messages.  I
>>>>>>>>> thought I saw some xml type messages earlier, but not sure how I enabled
>>>>>>>>> them.
>>>>>>>>>
>>>>>>>>> nta_outgoing: RTT is 67.958 ms
>>>>>>>>> outbound(0xb6e07c00): NAT binding changed: [210.xx.xxx.xx]:5080 !=
>>>>>>>>> [58.xxx.xx.xx]:5080
>>>>>>>>> nua(0xb6e07c00): event i_outbound 102 NAT binding changed
>>>>>>>>> nua: nua_application_event: entering
>>>>>>>>> 2010-10-24 09:05:08.849525 [DEBUG] sofia.c:956 nua_i_outbound:
>>>>>>>>> unknown event 8: 102 NAT binding changed
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Sun, Oct 24, 2010 at 1:08 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>>
>>>>>>>>>> btw: what router are you using?
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Sat, Oct 23, 2010 at 9:06 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>>>
>>>>>>>>>>> In a upnp config, FS is depending on the router to notify it of
>>>>>>>>>>> the ip address change.  It is possible the upnp message(s) aren't formatted
>>>>>>>>>>> as expected.  If you turn on debug logging you'll get the upnp messages on
>>>>>>>>>>> the console.  If you can pastebin the capture of those while dropping and
>>>>>>>>>>> restarting the dsl connection I can maybe see if there is something obvious
>>>>>>>>>>> going on.  The debug should also have messages related to the processing of
>>>>>>>>>>> those upnp messages.
>>>>>>>>>>>
>>>>>>>>>>> On Sat, Oct 23, 2010 at 7:04 AM, Mark Campbell-Smith <
>>>>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> HI!
>>>>>>>>>>>>
>>>>>>>>>>>> I know this question must have been answered 100's of times....
>>>>>>>>>>>>
>>>>>>>>>>>> My adsl is a bit dodgy at the moment and tends to go up and
>>>>>>>>>>>> down; which means I get assigned a new ip address from my ISP.  FS is nat'd
>>>>>>>>>>>> behind a upnp capable router.
>>>>>>>>>>>>
>>>>>>>>>>>> The problem is that FS does not seem to be detecting the change.
>>>>>>>>>>>>   For both the internal and external profile, I have auto-nat setup, and I
>>>>>>>>>>>> have tried stun and host settings in vars.conf.  When I issue a nat_map
>>>>>>>>>>>> status I see my old IP address, and if I then issue a nat_map reinit, I see
>>>>>>>>>>>> the new public IP address.  How can I get this to be automatic?
>>>>>>>>>>>>
>>>>>>>>>>>> Also the public IP address shown in the nat_map status is not
>>>>>>>>>>>> the same as shown in the sofia profile internal/external printouts.  Why is
>>>>>>>>>>>> this?
>>>>>>>>>>>>
>>>>>>>>>>>> I'm sure this is configuration, just not sure what to change.
>>>>>>>>>>>>
>>>>>>>>>>>> Thanks!
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> --
>>>>>>>>>>> -Rupa
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> --
>>>>>>>>>> -Rupa
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> -Rupa
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> -Rupa
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>  _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From fs-list at communicatefreely.net  Tue Oct 26 20:03:31 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Tue, 26 Oct 2010 23:03:31 -0400
Subject: [Freeswitch-users] caller-hunt-destination in lua
Message-ID: <4CC79683.7030301@communicatefreely.net>

Hello,

I am making extensive use of lua as a dialplan, and for the most part, it works really well.

The one problem I'm having is where I use execute_extension in an XML dialplan to execute the lua
dialplan and return.

In this situation, I would like to know what the hunt destination is, as opposed to the
dialed_destination.  They are not always the same.

For example:
<condition field="destination_number" expression="^(5\d{3})$">
	<action="execute_extension" data="$1 lua extension.lua"/>
</condition>

How do I know what the value of $1 was in the lua script?

Right now, I am using session:getVariable("caller_destination"), and it works for the most part, but
problems arise if there are nested executes.

If I execute an extension in a dialplan, that then executes another dialplan, the above variable
returns the destination in the first dialplan, whereas I want the destination in the most recent
dialplan (the one we are actually hunting in).

I tried passing it to the lua script as an argument, but it didn't show up in the argv[] table.

Any suggestions?




From moises.silva at gmail.com  Tue Oct 26 20:04:58 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 26 Oct 2010 23:04:58 -0400
Subject: [Freeswitch-users] FreeTDM - Screening Indicator Field
In-Reply-To: <AANLkTin-5ncn2NvGJo1Qmd5EkjUqf=xAaSr1_c_Kd79J@mail.gmail.com>
References: <AANLkTin-5ncn2NvGJo1Qmd5EkjUqf=xAaSr1_c_Kd79J@mail.gmail.com>
Message-ID: <AANLkTimNdWANeBvzTLpCUbmLxApo47VMu3HZNkotiDoi@mail.gmail.com>

On Tue, Oct 26, 2010 at 11:37 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> Hi,
> I have a server with Freeswitch (latest GIT revision) + FreeTDM with some
> Sangoma A108 boards.
> I have the necessity to set, for outgoing calls, the ISDN information
> element "Screening Indicator" of calling party number.
> I have see that some information element can be set in the
> "freetdm.conf.xml" file (i.e. calling and called typer of number or
> numbering plan) but I have not found any reference to the screening
> indicator.
> If there is no way to specify this information element for each trunk, it's
> possible to change its default value (that is 01=user provided, verified and
> passed)?

Not yet. But we can take a look at adding that. I am assuming you're
using ftmod_sangoma_isdn, is that the case?

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From jmesquita at freeswitch.org  Tue Oct 26 21:14:32 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 27 Oct 2010 01:14:32 -0300
Subject: [Freeswitch-users] encrypt gateway password in FSComm
In-Reply-To: <AANLkTim6eEZb89uWKh=nsGXLUUrhgAbY6wiU_EZoV+0r@mail.gmail.com>
References: <AANLkTim6eEZb89uWKh=nsGXLUUrhgAbY6wiU_EZoV+0r@mail.gmail.com>
Message-ID: <AANLkTimPxXemTyGbhym_=hFC0txzsJxy3MoJuihfbfWu@mail.gmail.com>

No you shouldn't get mod_qsettings back. It was basically to translate
QSettings to XML. Now we use XML directly so there is nothing in between the
config and the UI, making it safer and less error prone.

As for the A1-hash, I would like to ask the help of the core developers,
because I don't know the answer for that one.

Regards,

Jo?o Mesquita


On Sun, Oct 24, 2010 at 8:44 AM, Seven Du <dujinfang at gmail.com> wrote:

> Hi,
>
> Current FSComm using XML to store gateway information, is there a way
> to encrypt password?  It would be  great if it can do A1 hash like
> user directory.
>
> I remember there's a mod (mod_qsettings ? I forgot the name) in FSComm
> which been removed, should we get it back to support encrypted
> password?
>
> Thanks.
>
> --
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rupa at rupa.com  Tue Oct 26 22:01:39 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 27 Oct 2010 00:01:39 -0500
Subject: [Freeswitch-users] FS not detecting public IP address change
In-Reply-To: <AANLkTi=+ggi1L7oAhKGFFR8f=QaXhoj_fMGPJ1pUFk+v@mail.gmail.com>
References: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>
	<AANLkTinEoTcLmUVcpras8-m8pdkfpE=_r0spz1S50iMF@mail.gmail.com>
	<AANLkTi=MoV27FW0Ukd6K+mqmgnkyNgb9hjFx-vBYV1ak@mail.gmail.com>
	<AANLkTimCRfKSWG+5OdQY0=HTt2kOqxwk+0vKvaBNdmAd@mail.gmail.com>
	<AANLkTikfpRi9Shxiqd2w5nz3ZHsBnrHTt+=rHHnvoDJn@mail.gmail.com>
	<AANLkTikH3JJCCpL5s_eF2qjjttnEetE0i=nPn87zhBUL@mail.gmail.com>
	<AANLkTimzkb7qY8nW3M2afTPn18=OxY6jQye8D9TKr9Ts@mail.gmail.com>
	<AANLkTimR2x5AQPNiDDt4fZ8F=z_Wep3TBVR9Lob_72OA@mail.gmail.com>
	<AANLkTi=JgCME8u=27kUyLGU3cMP2nZpFpu7Zd5-4tpj0@mail.gmail.com>
	<AANLkTikL=kvbfLQx3E-vCaErRM-ARnU3_YSAvKo67Uvy@mail.gmail.com>
	<AF5D2E2E-F8A7-44AB-8EA0-D240A28A51F6@mgtech.com>
	<AANLkTikkFaJ-EWipGij2UAzMpmH8JGyhj+bUNKSDS8T5@mail.gmail.com>
	<AANLkTin=hRwNSBLxTRdJ6SqcPedbHiDgAJPCFESM9nCT@mail.gmail.com>
	<AANLkTi=+ggi1L7oAhKGFFR8f=QaXhoj_fMGPJ1pUFk+v@mail.gmail.com>
Message-ID: <AANLkTin9X9vXL0Sg8BNPO9opuEzSq=iLChwVypuSec2x@mail.gmail.com>

Then I have no idea why you aren't seeing upnp messages on the console.
 What do you see when you do a "nat_map reinit" ?

On Tue, Oct 26, 2010 at 8:09 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> I see debug from sofia.c, switch_core_state_machine, mod_sofia, diaplan
> parsing, sofia_glue, switch_rtp, switch_ivr_play_say etc.
>
>
>
> On Wed, Oct 27, 2010 at 11:41 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> Do you see ANY debug messages on your console from any other FS module?
>>  fsctl loglevel 7 sets the base log level, but not necessarily the log level
>> on your screen.
>>
>> /log 7
>>
>> sets the log level in fs_cli.
>>
>> Your screen should get VERY busy especially when processing a call.
>>
>> On Tue, Oct 26, 2010 at 4:21 PM, Mark Campbell-Smith <
>> mcampbellsmith at gmail.com> wrote:
>>
>>> I have it set to debug in switch.conf.xml and enable fsctl loglevel 7 in
>>> fs_cli....
>>>
>>> :~$ cat /usr/local/freeswitch/conf/autoload_configs/switch.conf.xml |
>>> egrep "param.*loglevel"
>>>     <param name="loglevel" value="debug"/>
>>>
>>> With regards to sofia: how does it determine if the external IP has
>>> changed?  And when it does, does sofia initiate a reregister of all external
>>> sip providers?
>>>
>>> Thanks!
>>>
>>> On Wed, Oct 27, 2010 at 3:14 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>
>>>> What you said: "And nat_map status shows me which ports have been
>>>> forwarded and the external ip address (which is sometimes wrong).  I also
>>>> see this in the router gui." is also happening to me on a Linksys/Cisco
>>>> RV042 router. Can't keep FS working for more than 2-5 hours without doing a
>>>> restart or nat_map reinit. I think the routers upnp has a problem with dual
>>>> wans, I have one static and one dynamic DSL.
>>>>
>>>> On Oct 26, 2010, at 6:43 AM, Rupa Schomaker wrote:
>>>>
>>>> What do you have set for loglevel in switch.conf.xml?  My guess is that
>>>> you have it set to something other than debug.  If that is set to (say)
>>>> info, then info is as low as you can go.  Setting the console log level to
>>>> debug will not give you debug output.
>>>>
>>>> sofia is unaware of upnp so a keepalive packet should have no impact on
>>>> it's operation.  I'm not familiar enough with sofia to tell what triggers
>>>> that event.
>>>>
>>>> On Tue, Oct 26, 2010 at 6:24 AM, Mark Campbell-Smith <
>>>> mcampbellsmith at gmail.com> wrote:
>>>>
>>>>> OK..  tcpdump gave me this and I still do not see anything on FS except
>>>>> for this line:
>>>>>
>>>>> 2010-10-26 22:02:51.235608 [DEBUG] sofia.c:957 nua_i_outbound: unknown
>>>>> event 8: 102 NAT binding changed
>>>>>
>>>>> Hypertext Transfer Protocol
>>>>>     NOTIFY * HTTP/1.1\r\n
>>>>>     HOST:239.255.255.250:1900\r\n
>>>>>     Cache-Control:max-age=120\r\n
>>>>>     Location:http://192.168.1.1:1278/rootDesc.xml\r\n<http://192.168.1.1:1278/rootDesc.xml%5Cr%5Cn>
>>>>>     Server: Tomato UPnP/1.0 MiniUPnPd/1.4\r\n
>>>>>     NT:urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>>>>>
>>>>>  USN:uuid:882dbe81-c28a-4992-baf7-410c05f1caf4::urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>>>>>     NTS:ssdp:alive\r\n
>>>>>     \r\n
>>>>>
>>>>> This looks the same as you posted before, so why do I get the unknown
>>>>> event 8 (is this related)?
>>>>>
>>>>> Thanks!
>>>>>
>>>>> On Tue, Oct 26, 2010 at 3:32 PM, Mark Campbell-Smith <
>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>
>>>>>> :)  that was the first thing I did, but I get nothing; no keep alive
>>>>>> packets.
>>>>>>
>>>>>> in the logfile at startup I see:
>>>>>>
>>>>>> 2010-10-26 10:19:45.484258 [INFO] switch_nat.c:410 Scanning for NAT
>>>>>> 2010-10-26 10:19:45.485991 [DEBUG] switch_nat.c:167 Checking for PMP
>>>>>> 1/5
>>>>>> 2010-10-26 10:19:45.488575 [INFO] switch_nat.c:423 NAT detected type:
>>>>>> pmp, ExtIP: '203.xxx.xxx.xxx'
>>>>>> 2010-10-26 10:19:45.490683 [DEBUG] switch_nat.c:256 NAT thread
>>>>>> configured
>>>>>> 2010-10-26 10:19:45.492180 [DEBUG] switch_nat.c:267 NAT thread started
>>>>>>
>>>>>> And nat_map status shows me which ports have been forwarded and the
>>>>>> external ip address (which is sometimes wrong).  I also see this in the
>>>>>> router gui.
>>>>>>
>>>>>> But I never see the keep alive packets.
>>>>>>
>>>>>> On Tue, Oct 26, 2010 at 3:13 PM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>
>>>>>>> Just turn up console logging to level 7.
>>>>>>>
>>>>>>> /log 7 if using fs_cli
>>>>>>>
>>>>>>>
>>>>>>> On Mon, Oct 25, 2010 at 6:53 PM, Mark Campbell-Smith <
>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>
>>>>>>>> Hi!
>>>>>>>>
>>>>>>>> How do I enable debugging to get the UPnP keep alive packets?
>>>>>>>>
>>>>>>>> Thanks!
>>>>>>>>
>>>>>>>> On Tue, Oct 26, 2010 at 1:15 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>
>>>>>>>>> I need the debug logging from the upnp stuff.  It should look
>>>>>>>>> something like:
>>>>>>>>>
>>>>>>>>> +OK log level 7 [7]
>>>>>>>>> freeswitch at internal> 2010-10-25 09:13:59.926861 [DEBUG]
>>>>>>>>> switch_nat.c:299 got UPnP keep alive packet:
>>>>>>>>> NOTIFY * HTTP/1.1
>>>>>>>>> HOST:239.255.255.250:1900
>>>>>>>>> Cache-Control:max-age=60
>>>>>>>>> Location:http://192.168.1.1:5000/rootDesc.xml
>>>>>>>>> Server: Tomato UPnP/1.0 MiniUPnPd/1.4
>>>>>>>>> NT:urn:schemas-upnp-org:service:WANIPConnection:1
>>>>>>>>>
>>>>>>>>> USN:uuid:00000000-0000-0000-0000-000000000000::urn:schemas-upnp-org:service:WANIPConnection:1
>>>>>>>>> NTS:ssdp:alive
>>>>>>>>>
>>>>>>>>> I also run tomato and haven't had trouble with the upnp support so
>>>>>>>>> at least we have that part working.
>>>>>>>>>
>>>>>>>>> You should see the above keepalive appear periodically every 30s or
>>>>>>>>> so.  You should see another set of messages when you terminate the DSL
>>>>>>>>> connection and when that dsl connection comes back online.
>>>>>>>>>
>>>>>>>>> On Sat, Oct 23, 2010 at 5:36 PM, Mark Campbell-Smith <
>>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> Hi!
>>>>>>>>>>
>>>>>>>>>> The router is a ASUS router but it is running the tomato firmware
>>>>>>>>>> which runs miniupnd.
>>>>>>>>>>
>>>>>>>>>> In the debug I see the message: 2010-10-24 08:02:59.178918 [DEBUG]
>>>>>>>>>> sofia.c:956 nua_i_outbound: unknown event 8: 101 NAT detected
>>>>>>>>>>
>>>>>>>>>> This is what I see below. I hope I enabled all debug messages.  I
>>>>>>>>>> thought I saw some xml type messages earlier, but not sure how I enabled
>>>>>>>>>> them.
>>>>>>>>>>
>>>>>>>>>> nta_outgoing: RTT is 67.958 ms
>>>>>>>>>> outbound(0xb6e07c00): NAT binding changed: [210.xx.xxx.xx]:5080 !=
>>>>>>>>>> [58.xxx.xx.xx]:5080
>>>>>>>>>> nua(0xb6e07c00): event i_outbound 102 NAT binding changed
>>>>>>>>>> nua: nua_application_event: entering
>>>>>>>>>> 2010-10-24 09:05:08.849525 [DEBUG] sofia.c:956 nua_i_outbound:
>>>>>>>>>> unknown event 8: 102 NAT binding changed
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Sun, Oct 24, 2010 at 1:08 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>>>
>>>>>>>>>>> btw: what router are you using?
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> On Sat, Oct 23, 2010 at 9:06 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>>>>
>>>>>>>>>>>> In a upnp config, FS is depending on the router to notify it of
>>>>>>>>>>>> the ip address change.  It is possible the upnp message(s) aren't formatted
>>>>>>>>>>>> as expected.  If you turn on debug logging you'll get the upnp messages on
>>>>>>>>>>>> the console.  If you can pastebin the capture of those while dropping and
>>>>>>>>>>>> restarting the dsl connection I can maybe see if there is something obvious
>>>>>>>>>>>> going on.  The debug should also have messages related to the processing of
>>>>>>>>>>>> those upnp messages.
>>>>>>>>>>>>
>>>>>>>>>>>> On Sat, Oct 23, 2010 at 7:04 AM, Mark Campbell-Smith <
>>>>>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> HI!
>>>>>>>>>>>>>
>>>>>>>>>>>>> I know this question must have been answered 100's of times....
>>>>>>>>>>>>>
>>>>>>>>>>>>> My adsl is a bit dodgy at the moment and tends to go up and
>>>>>>>>>>>>> down; which means I get assigned a new ip address from my ISP.  FS is nat'd
>>>>>>>>>>>>> behind a upnp capable router.
>>>>>>>>>>>>>
>>>>>>>>>>>>> The problem is that FS does not seem to be detecting the
>>>>>>>>>>>>> change.   For both the internal and external profile, I have auto-nat setup,
>>>>>>>>>>>>> and I have tried stun and host settings in vars.conf.  When I issue
>>>>>>>>>>>>> a nat_map status I see my old IP address, and if I then issue a nat_map
>>>>>>>>>>>>> reinit, I see the new public IP address.  How can I get this to be
>>>>>>>>>>>>> automatic?
>>>>>>>>>>>>>
>>>>>>>>>>>>> Also the public IP address shown in the nat_map status is not
>>>>>>>>>>>>> the same as shown in the sofia profile internal/external printouts.  Why is
>>>>>>>>>>>>> this?
>>>>>>>>>>>>>
>>>>>>>>>>>>> I'm sure this is configuration, just not sure what to change.
>>>>>>>>>>>>>
>>>>>>>>>>>>> Thanks!
>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> --
>>>>>>>>>>>> -Rupa
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> --
>>>>>>>>>>> -Rupa
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>> -Rupa
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> -Rupa
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>  _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From wstephen80 at gmail.com  Tue Oct 26 23:55:26 2010
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Wed, 27 Oct 2010 08:55:26 +0200
Subject: [Freeswitch-users] FreeTDM - Screening Indicator Field
In-Reply-To: <AANLkTimNdWANeBvzTLpCUbmLxApo47VMu3HZNkotiDoi@mail.gmail.com>
References: <AANLkTin-5ncn2NvGJo1Qmd5EkjUqf=xAaSr1_c_Kd79J@mail.gmail.com>
	<AANLkTimNdWANeBvzTLpCUbmLxApo47VMu3HZNkotiDoi@mail.gmail.com>
Message-ID: <AANLkTi=qfumFZKopgP9hxEA0EGMWgG=6SXPjaa7E8cXe@mail.gmail.com>

Hi,
yes, I'm using ftmod_sangoma_isdn library.
Thanks,

Stephen

On Wed, Oct 27, 2010 at 5:04 AM, Moises Silva <moises.silva at gmail.com>wrote:

>
> Not yet. But we can take a look at adding that. I am assuming you're
> using ftmod_sangoma_isdn, is that the case?
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
> L3R 9R6 Canada
> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From shamun.toha at gmail.com  Tue Oct 26 23:58:53 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Wed, 27 Oct 2010 08:58:53 +0200
Subject: [Freeswitch-users] FreeSWITCH - how to enable channels like
 mod_skypopen for Yahoo, Google Talk?
Message-ID: <AANLkTikFHXgu-nZ5NBDQ4xON4c2D1WRHX33wOSRjPdSB@mail.gmail.com>

Hello,

How can i enable Yahoo channels or Google Talk channels. Using mod_skypopen?


Thanks & Best regards
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From steveayre at gmail.com  Wed Oct 27 00:15:58 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 27 Oct 2010 08:15:58 +0100
Subject: [Freeswitch-users] FreeSWITCH - how to enable channels like
 mod_skypopen for Yahoo, Google Talk?
In-Reply-To: <AANLkTikFHXgu-nZ5NBDQ4xON4c2D1WRHX33wOSRjPdSB@mail.gmail.com>
References: <AANLkTikFHXgu-nZ5NBDQ4xON4c2D1WRHX33wOSRjPdSB@mail.gmail.com>
Message-ID: <AANLkTimMfFohwpBhkyK+eVyF0qu86jUyVxUgOPcx-MgL@mail.gmail.com>

Mod_dingaling is used to connect to Google Talk. The rest is all routing.
http://wiki.freeswitch.org/wiki/Mod_dingaling

Yahoo appears to accept SIP.  But it also looks like they use their
own extensions on top of the SIP protocol for authentication (Y-Cookie
header) so I don't think it'll work currently.
http://wiki.freeswitch.org/wiki/Provider_Configuration:_Yahoo_Messenger
http://blog.motiwala.com/2007/08/18/yahoo-messenger-voip-service-with-sip-phone

-Steve


On 27 October 2010 07:58, Shamun toha md <shamun.toha at gmail.com> wrote:
> Hello,
>
> How can i enable Yahoo channels or Google Talk channels. Using mod_skypopen?
>
>
> Thanks & Best regards
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From khurram at breezecom.ae  Wed Oct 27 02:27:10 2010
From: khurram at breezecom.ae (Khurram Shafi)
Date: Wed, 27 Oct 2010 14:27:10 +0500
Subject: [Freeswitch-users] Sofia (internal) Profile Locks up after 2-3 hours
Message-ID: <05cf01cb75b9$242dc510$6c894f30$@ae>

Hello,

 

I am having strange behavior of FS Sofia profile being locked up after
running 1200+ calls for 2-3 hours.

There is nothing in logs. FS internal profile which is set on port 5060
simply stops listening on port 5060 and accepts no calls on it. 

 

I m running FreeSWITCH Version 1.0.head (git-b834a0e 2010-09-19 16-59-58
-0500)

 

Regards

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From t.mahe at telemaque.fr  Wed Oct 27 04:11:48 2010
From: t.mahe at telemaque.fr (=?UTF-8?B?VHJpc3RhbiBNYWjDqQ==?=)
Date: Wed, 27 Oct 2010 13:11:48 +0200
Subject: [Freeswitch-users] Is anyone familiar with Telemaque?
In-Reply-To: <AANLkTimg-wDSH4U-XLV1up7Bp1t_UrbpRo9XdG8LqdrY@mail.gmail.com>
References: <AANLkTimib+3T7L9sP28HN6BREJF5ca2j2nou-84YU=Ln@mail.gmail.com>	<4CAAE5DE.6080407@telemaque.fr>
	<4CABF9D5.7000501@bksys.co.in>	<4CAC4EB4.6080901@telemaque.fr>
	<4CC6A214.1080601@bksys.co.in>
	<AANLkTimg-wDSH4U-XLV1up7Bp1t_UrbpRo9XdG8LqdrY@mail.gmail.com>
Message-ID: <4CC808F4.3030703@telemaque.fr>

Hi guys,

Just to answer the first question, we're not using mod_fifo, but a c 
module that looks more like mod_callcenter (highly integrated in our 
internal tools )
with a bunch of lua scripts running on multiples fs boxes and multiple 
profiles with a bunch of kamailio boxes dispatching the load between fs...

mod_fifo was not the right tool for us, as we needed multiple criterias 
to select the good agent ( well a little more than that, but that's the 
first point ).

Regarding fine-tuning freeswitch, it depends on a lot of things, and I 
think the wiki is well documented on that case. If you have precise 
questions,
don't hesitate, but that's a little bit too 'broadband' to answer like 
that ;)

Concerning mod_fifo, I'm surprised of that 25 concurrent call limit, 
though i've not extensively tested it and I know Anthony did some tweaks 
recently on it... mod_fifo should handle more without troubles...

@Ghulam: with fs you don't need 2 servers to handle 250 incoming calls, 
and I won't recommend vicidial + ndb storage engine ( at least when we 
tested it some long time ago ), as ndb behaves totally differently than 
myisam/innodb..., queries must fit the ndb engine behaviour, and that 
was not the case, causing a lot of perf loss, maybe it has changed ?

Regards,

Tristan.

Le 26/10/2010 22:44, Ghulam Mustafa a ?crit :
> Hi,
>
> I am not sure about how mod_fifo will behave if there is a high call
> volume lets's say 300 calls in the queue, i am planning to use
> mod_fifo with mod_conference for inbound call center scenario.
>
> As you said freeswitch hangs if there are ~25 calls in the fifo, i
> would like someone to advice me how scalable mod_fifo is? is it
> capable of handling 300 calls (on quadcore xeon servers with 8g ram),
> if not then what are other options to queue incoming calls in FS.
>
> Currently we are using 2 vicidial (asterisk+astguiclient) server + 2
> web and 3 node mysql-cluster (all Dell R710 series) servers setup to
> handle 250+ incoming concurrent calls. though this setup works but i
> am not happy with the way vicidial abuses system and database
> resources. (with excuse to matt florel and other vicidial authors,
> infact what they developed is competing big boys in the market and
> works like a charm), i am sure they will improve and fix their
> technical design with the time.
>
> Best Regards,
>
>
> On Tue, Oct 26, 2010 at 2:40 PM, abubacker<abubacker at bksystems.co.in>  wrote:
>    
>> On Wednesday 06 October 2010 03:55 PM, Tristan Mah? wrote:
>>
>> Hi,
>>
>> The telco cluster is powered by IBM servers ( from dual Xeon/8gb ram to dual
>> quad core/16gb ram depending on the node ) with FC13 and latest FS git (
>> well almost always, as of today I'm a week late, but will upgrade soon again
>> to benefit of the amazing work the dev do ).
>>
>> If you have more precise questions, don't hesitate, I'm sorry if I can't
>> reveal all the details on how we're doing things,but I'm sure you can
>> understand that...
>>
>> Regards,
>>
>> Tristan.
>> Le 10/06/2010 06:23 AM, abubacker a ?crit :
>>
>> On Tuesday 05 October 2010 02:16 PM, Tristan Mah? wrote:
>>
>> There's someone working for them on list :)
>>
>> Feel free to ask me questions Michael.
>>
>> Regards,
>>
>> Gled.
>>
>> Le 10/04/2010 08:39 PM, Michael Collins a ?crit :
>>
>> Dear FreeSWITCH Community,
>>
>> Some of you may have stumbled upon this news item:
>>
>> http://blogs.oracle.com/mysql/2010/10/innovating_with_open_source_call_center_services.html
>>
>> Evidently there is a company in France (Telemaque) who is using MySQL +
>> Kamailio + FreeSWITCH for some heavy duty call processing. If you are at all
>> familiar with this company please let me know. I'd like to learn more about
>> what they are doing. (Je ne parle pas boucoup de Francais :( )
>>
>> Thanks for your help,
>> MC
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> "Feel free to ask me questions Michael"
>>
>> I think I must use this opportunity ,
>> please specify the software ( OS ) and the hardwares required to handle the
>> heavy duty call
>> processing and also specify the FreeSWITCH version.
>> I guess you could answer this very precisely.
>>
>> Thanks in Advance !
>>
>> --
>> Best regards,
>> N.Abubacker ,
>> Associate system engineer ,
>> bk systems pvt ltd ,
>> Ph : 9144-43902701
>>
>> Disclaimer: http://www.bksystems.co.in/email-policy
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> Dear Tristan ,
>> Knew this is an old post , but even though I wanted this to continue to get
>> some more details ,
>> what mod you are using  to handle more calls ? ( hope its mod_fifo )
>> I am working in mod_fifo , some times it throws socketerror and makes the
>> entire freeswitch hang
>> when more than 25 calls in a queue.
>>
>> My question is can u give me some details about the queue and member
>> configuration ( if possible )
>>
>> If you are not using mod_fifo , just tell me the general configuration
>> required to fine tune the
>> freeswitch.
>>
>> Thanks Again !
>>
>> --
>> Best regards,
>> N.Abubacker ,
>> Associate system engineer ,
>> bk systems pvt ltd ,
>> Ph : 9144-43902701
>>
>> Disclaimer: http://www.bksystems.co.in/email-policy
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>      
>
>
>    



From steveayre at gmail.com  Wed Oct 27 04:21:54 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 27 Oct 2010 12:21:54 +0100
Subject: [Freeswitch-users] Sofia (internal) Profile Locks up after 2-3
	hours
In-Reply-To: <05cf01cb75b9$242dc510$6c894f30$@ae>
References: <05cf01cb75b9$242dc510$6c894f30$@ae>
Message-ID: <AANLkTin243MsPeFcedegfAFAPh5iRotxX_o2ggp=uJLE@mail.gmail.com>

It probably means that there's a mutex that's been locked by something
but not rleaseed.

First, upgrade your version to the latest git. It's possible it's a
bug that's already been fixed.

If you can reproduce it on the new version, install gcore and collect
a corefile while FS is hung:
http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Simple_bash_script_to_make_debug_easy
and open a Jira ticket with the results at http://jira.freeswitch.org/browse/FS

-Steve



On 27 October 2010 10:27, Khurram Shafi <khurram at breezecom.ae> wrote:
> Hello,
>
>
>
> I am having strange behavior of FS Sofia profile being locked up after
> running 1200+ calls for 2-3 hours.
>
> There is nothing in logs. FS internal profile which is set on port 5060
> simply stops listening on port 5060 and accepts no calls on it.
>
>
>
> I m running FreeSWITCH Version 1.0.head (git-b834a0e 2010-09-19 16-59-58
> -0500)
>
>
>
> Regards
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From ovvenkatesan at gmail.com  Wed Oct 27 06:23:03 2010
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Wed, 27 Oct 2010 18:53:03 +0530
Subject: [Freeswitch-users] issue with freetdm
Message-ID: <AANLkTimwOY+TcjacxuXXWJ=Da1k3MptOj=sOAhwvaLDR@mail.gmail.com>

Hello,

I am using sangoma pri card A101 with freeSwitch.
Today, I moved from openzap to freetdm.
I have followed *
http://wiki.sangoma.com/wanpipe-freeswitch-ftdm-installation* page to
install freeSwitch and freedtm.

After installing  freeSwitch and Sangoma, I am getting following error in *
fs_cli

2010-10-27 17:57:03.228457 [ERR] ftmod_sangoma_isdn_stack_rcv.c:883
sng_isdn->s1:L1 Rx Error
2010-10-27 17:57:03.228457 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:871
sng_isdn->s1: Resetting L1 link

*
What is these mean?

I have followed wiki page and installed all the components.



-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From mcampbellsmith at gmail.com  Tue Oct 26 22:13:31 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 27 Oct 2010 16:13:31 +1100
Subject: [Freeswitch-users] FS not detecting public IP address change
In-Reply-To: <AANLkTin9X9vXL0Sg8BNPO9opuEzSq=iLChwVypuSec2x@mail.gmail.com>
References: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>
	<AANLkTinEoTcLmUVcpras8-m8pdkfpE=_r0spz1S50iMF@mail.gmail.com>
	<AANLkTi=MoV27FW0Ukd6K+mqmgnkyNgb9hjFx-vBYV1ak@mail.gmail.com>
	<AANLkTimCRfKSWG+5OdQY0=HTt2kOqxwk+0vKvaBNdmAd@mail.gmail.com>
	<AANLkTikfpRi9Shxiqd2w5nz3ZHsBnrHTt+=rHHnvoDJn@mail.gmail.com>
	<AANLkTikH3JJCCpL5s_eF2qjjttnEetE0i=nPn87zhBUL@mail.gmail.com>
	<AANLkTimzkb7qY8nW3M2afTPn18=OxY6jQye8D9TKr9Ts@mail.gmail.com>
	<AANLkTimR2x5AQPNiDDt4fZ8F=z_Wep3TBVR9Lob_72OA@mail.gmail.com>
	<AANLkTi=JgCME8u=27kUyLGU3cMP2nZpFpu7Zd5-4tpj0@mail.gmail.com>
	<AANLkTikL=kvbfLQx3E-vCaErRM-ARnU3_YSAvKo67Uvy@mail.gmail.com>
	<AF5D2E2E-F8A7-44AB-8EA0-D240A28A51F6@mgtech.com>
	<AANLkTikkFaJ-EWipGij2UAzMpmH8JGyhj+bUNKSDS8T5@mail.gmail.com>
	<AANLkTin=hRwNSBLxTRdJ6SqcPedbHiDgAJPCFESM9nCT@mail.gmail.com>
	<AANLkTi=+ggi1L7oAhKGFFR8f=QaXhoj_fMGPJ1pUFk+v@mail.gmail.com>
	<AANLkTin9X9vXL0Sg8BNPO9opuEzSq=iLChwVypuSec2x@mail.gmail.com>
Message-ID: <AANLkTi=KL7w=ZeudXyEPu3S78CadmM88HfFFrzkEkcO3@mail.gmail.com>

That gives me quite a bit....

Can there be the case that nat_map status has a different IP address to what
I would see in sofia status profile internal?  And does sofia reregister if
there is detected a public IP address change?

Thanks

freeswitch at internal> nat_map reinit
[DEBUG] libs/esl/src/esl.c:1141 esl_send() SEND
api nat_map reinit
console_execute: true


[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type] =
[api/response]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Length]
= [130]
Nat Type: NAT-PMP, ExtIP: 210.xx.xxx.xx
port,proto,proto_num,sticky
5060,udp,0,0
5060,tcp,1,0
5080,udp,0,0
5080,tcp,1,0

4 total.

freeswitch at internal> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV
HEADER [Content-Type] = [log/data]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Length]
= [68]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Level] =
[6]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Text-Channel] =
[0]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-File] =
[switch_nat.c]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Func] =
[switch_nat_init]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Line] =
[410]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [User-Data] = []
[DEBUG] libs/esl/src/esl.c:1115 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 68
Log-Level: 6
Text-Channel: 0
Log-File: switch_nat.c
Log-Func: switch_nat_init
Log-Line: 410
User-Data: _undef_

2010-10-27 16:10:07.003208 [INFO] switch_nat.c:410 Scanning for NAT

2010-10-27 16:10:07.003208 [INFO] switch_nat.c:410 Scanning for NAT
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type] =
[log/data]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Length]
= [73]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Level] =
[7]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Text-Channel] =
[0]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-File] =
[switch_nat.c]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Func] =
[get_pmp_pubaddr]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Line] =
[167]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [User-Data] = []
[DEBUG] libs/esl/src/esl.c:1115 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 73
Log-Level: 7
Text-Channel: 0
Log-File: switch_nat.c
Log-Func: get_pmp_pubaddr
Log-Line: 167
User-Data: _undef_

2010-10-27 16:10:07.003208 [DEBUG] switch_nat.c:167 Checking for PMP 1/5

2010-10-27 16:10:07.003208 [DEBUG] switch_nat.c:167 Checking for PMP 1/5
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type] =
[log/data]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Length]
= [98]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Level] =
[6]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Text-Channel] =
[0]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-File] =
[switch_nat.c]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Func] =
[switch_nat_init]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Line] =
[423]
[DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [User-Data] = []
[DEBUG] libs/esl/src/esl.c:1115 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: log/data
Content-Length: 98
Log-Level: 6
Text-Channel: 0
Log-File: switch_nat.c
Log-Func: switch_nat_init
Log-Line: 423
User-Data: _undef_

2010-10-27 16:10:07.013797 [INFO] switch_nat.c:423 NAT detected type: pmp,
ExtIP: '210.xx.xxx.xx'

2010-10-27 16:10:07.013797 [INFO] switch_nat.c:423 NAT detected type: pmp,
ExtIP: '210.xx.xxx.xx'


On Wed, Oct 27, 2010 at 4:01 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> Then I have no idea why you aren't seeing upnp messages on the console.
>  What do you see when you do a "nat_map reinit" ?
>
>
> On Tue, Oct 26, 2010 at 8:09 PM, Mark Campbell-Smith <
> mcampbellsmith at gmail.com> wrote:
>
>> I see debug from sofia.c, switch_core_state_machine, mod_sofia, diaplan
>> parsing, sofia_glue, switch_rtp, switch_ivr_play_say etc.
>>
>>
>>
>> On Wed, Oct 27, 2010 at 11:41 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>>> Do you see ANY debug messages on your console from any other FS module?
>>>  fsctl loglevel 7 sets the base log level, but not necessarily the log level
>>> on your screen.
>>>
>>> /log 7
>>>
>>> sets the log level in fs_cli.
>>>
>>> Your screen should get VERY busy especially when processing a call.
>>>
>>> On Tue, Oct 26, 2010 at 4:21 PM, Mark Campbell-Smith <
>>> mcampbellsmith at gmail.com> wrote:
>>>
>>>> I have it set to debug in switch.conf.xml and enable fsctl loglevel 7 in
>>>> fs_cli....
>>>>
>>>> :~$ cat /usr/local/freeswitch/conf/autoload_configs/switch.conf.xml |
>>>> egrep "param.*loglevel"
>>>>     <param name="loglevel" value="debug"/>
>>>>
>>>> With regards to sofia: how does it determine if the external IP has
>>>> changed?  And when it does, does sofia initiate a reregister of all external
>>>> sip providers?
>>>>
>>>> Thanks!
>>>>
>>>> On Wed, Oct 27, 2010 at 3:14 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>>
>>>>> What you said: "And nat_map status shows me which ports have been
>>>>> forwarded and the external ip address (which is sometimes wrong).  I also
>>>>> see this in the router gui." is also happening to me on a Linksys/Cisco
>>>>> RV042 router. Can't keep FS working for more than 2-5 hours without doing a
>>>>> restart or nat_map reinit. I think the routers upnp has a problem with dual
>>>>> wans, I have one static and one dynamic DSL.
>>>>>
>>>>> On Oct 26, 2010, at 6:43 AM, Rupa Schomaker wrote:
>>>>>
>>>>> What do you have set for loglevel in switch.conf.xml?  My guess is that
>>>>> you have it set to something other than debug.  If that is set to (say)
>>>>> info, then info is as low as you can go.  Setting the console log level to
>>>>> debug will not give you debug output.
>>>>>
>>>>> sofia is unaware of upnp so a keepalive packet should have no impact on
>>>>> it's operation.  I'm not familiar enough with sofia to tell what triggers
>>>>> that event.
>>>>>
>>>>> On Tue, Oct 26, 2010 at 6:24 AM, Mark Campbell-Smith <
>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>
>>>>>> OK..  tcpdump gave me this and I still do not see anything on FS
>>>>>> except for this line:
>>>>>>
>>>>>> 2010-10-26 22:02:51.235608 [DEBUG] sofia.c:957 nua_i_outbound: unknown
>>>>>> event 8: 102 NAT binding changed
>>>>>>
>>>>>> Hypertext Transfer Protocol
>>>>>>     NOTIFY * HTTP/1.1\r\n
>>>>>>     HOST:239.255.255.250:1900\r\n
>>>>>>     Cache-Control:max-age=120\r\n
>>>>>>     Location:http://192.168.1.1:1278/rootDesc.xml\r\n<http://192.168.1.1:1278/rootDesc.xml%5Cr%5Cn>
>>>>>>     Server: Tomato UPnP/1.0 MiniUPnPd/1.4\r\n
>>>>>>     NT:urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>>>>>>
>>>>>>  USN:uuid:882dbe81-c28a-4992-baf7-410c05f1caf4::urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>>>>>>     NTS:ssdp:alive\r\n
>>>>>>     \r\n
>>>>>>
>>>>>> This looks the same as you posted before, so why do I get the unknown
>>>>>> event 8 (is this related)?
>>>>>>
>>>>>> Thanks!
>>>>>>
>>>>>> On Tue, Oct 26, 2010 at 3:32 PM, Mark Campbell-Smith <
>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>
>>>>>>> :)  that was the first thing I did, but I get nothing; no keep alive
>>>>>>> packets.
>>>>>>>
>>>>>>> in the logfile at startup I see:
>>>>>>>
>>>>>>> 2010-10-26 10:19:45.484258 [INFO] switch_nat.c:410 Scanning for NAT
>>>>>>> 2010-10-26 10:19:45.485991 [DEBUG] switch_nat.c:167 Checking for PMP
>>>>>>> 1/5
>>>>>>> 2010-10-26 10:19:45.488575 [INFO] switch_nat.c:423 NAT detected type:
>>>>>>> pmp, ExtIP: '203.xxx.xxx.xxx'
>>>>>>> 2010-10-26 10:19:45.490683 [DEBUG] switch_nat.c:256 NAT thread
>>>>>>> configured
>>>>>>> 2010-10-26 10:19:45.492180 [DEBUG] switch_nat.c:267 NAT thread
>>>>>>> started
>>>>>>>
>>>>>>> And nat_map status shows me which ports have been forwarded and the
>>>>>>> external ip address (which is sometimes wrong).  I also see this in the
>>>>>>> router gui.
>>>>>>>
>>>>>>> But I never see the keep alive packets.
>>>>>>>
>>>>>>> On Tue, Oct 26, 2010 at 3:13 PM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>
>>>>>>>> Just turn up console logging to level 7.
>>>>>>>>
>>>>>>>> /log 7 if using fs_cli
>>>>>>>>
>>>>>>>>
>>>>>>>> On Mon, Oct 25, 2010 at 6:53 PM, Mark Campbell-Smith <
>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Hi!
>>>>>>>>>
>>>>>>>>> How do I enable debugging to get the UPnP keep alive packets?
>>>>>>>>>
>>>>>>>>> Thanks!
>>>>>>>>>
>>>>>>>>> On Tue, Oct 26, 2010 at 1:15 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>>
>>>>>>>>>> I need the debug logging from the upnp stuff.  It should look
>>>>>>>>>> something like:
>>>>>>>>>>
>>>>>>>>>> +OK log level 7 [7]
>>>>>>>>>> freeswitch at internal> 2010-10-25 09:13:59.926861 [DEBUG]
>>>>>>>>>> switch_nat.c:299 got UPnP keep alive packet:
>>>>>>>>>> NOTIFY * HTTP/1.1
>>>>>>>>>> HOST:239.255.255.250:1900
>>>>>>>>>> Cache-Control:max-age=60
>>>>>>>>>> Location:http://192.168.1.1:5000/rootDesc.xml
>>>>>>>>>> Server: Tomato UPnP/1.0 MiniUPnPd/1.4
>>>>>>>>>> NT:urn:schemas-upnp-org:service:WANIPConnection:1
>>>>>>>>>>
>>>>>>>>>> USN:uuid:00000000-0000-0000-0000-000000000000::urn:schemas-upnp-org:service:WANIPConnection:1
>>>>>>>>>> NTS:ssdp:alive
>>>>>>>>>>
>>>>>>>>>> I also run tomato and haven't had trouble with the upnp support so
>>>>>>>>>> at least we have that part working.
>>>>>>>>>>
>>>>>>>>>> You should see the above keepalive appear periodically every 30s
>>>>>>>>>> or so.  You should see another set of messages when you terminate the DSL
>>>>>>>>>> connection and when that dsl connection comes back online.
>>>>>>>>>>
>>>>>>>>>> On Sat, Oct 23, 2010 at 5:36 PM, Mark Campbell-Smith <
>>>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> Hi!
>>>>>>>>>>>
>>>>>>>>>>> The router is a ASUS router but it is running the tomato firmware
>>>>>>>>>>> which runs miniupnd.
>>>>>>>>>>>
>>>>>>>>>>> In the debug I see the message: 2010-10-24 08:02:59.178918
>>>>>>>>>>> [DEBUG] sofia.c:956 nua_i_outbound: unknown event 8: 101 NAT detected
>>>>>>>>>>>
>>>>>>>>>>> This is what I see below. I hope I enabled all debug messages.  I
>>>>>>>>>>> thought I saw some xml type messages earlier, but not sure how I enabled
>>>>>>>>>>> them.
>>>>>>>>>>>
>>>>>>>>>>> nta_outgoing: RTT is 67.958 ms
>>>>>>>>>>> outbound(0xb6e07c00): NAT binding changed: [210.xx.xxx.xx]:5080
>>>>>>>>>>> != [58.xxx.xx.xx]:5080
>>>>>>>>>>> nua(0xb6e07c00): event i_outbound 102 NAT binding changed
>>>>>>>>>>> nua: nua_application_event: entering
>>>>>>>>>>> 2010-10-24 09:05:08.849525 [DEBUG] sofia.c:956 nua_i_outbound:
>>>>>>>>>>> unknown event 8: 102 NAT binding changed
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> On Sun, Oct 24, 2010 at 1:08 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>>>>
>>>>>>>>>>>> btw: what router are you using?
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> On Sat, Oct 23, 2010 at 9:06 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> In a upnp config, FS is depending on the router to notify it of
>>>>>>>>>>>>> the ip address change.  It is possible the upnp message(s) aren't formatted
>>>>>>>>>>>>> as expected.  If you turn on debug logging you'll get the upnp messages on
>>>>>>>>>>>>> the console.  If you can pastebin the capture of those while dropping and
>>>>>>>>>>>>> restarting the dsl connection I can maybe see if there is something obvious
>>>>>>>>>>>>> going on.  The debug should also have messages related to the processing of
>>>>>>>>>>>>> those upnp messages.
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Sat, Oct 23, 2010 at 7:04 AM, Mark Campbell-Smith <
>>>>>>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>> HI!
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> I know this question must have been answered 100's of
>>>>>>>>>>>>>> times....
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> My adsl is a bit dodgy at the moment and tends to go up and
>>>>>>>>>>>>>> down; which means I get assigned a new ip address from my ISP.  FS is nat'd
>>>>>>>>>>>>>> behind a upnp capable router.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> The problem is that FS does not seem to be detecting the
>>>>>>>>>>>>>> change.   For both the internal and external profile, I have auto-nat setup,
>>>>>>>>>>>>>> and I have tried stun and host settings in vars.conf.  When I issue
>>>>>>>>>>>>>> a nat_map status I see my old IP address, and if I then issue a nat_map
>>>>>>>>>>>>>> reinit, I see the new public IP address.  How can I get this to be
>>>>>>>>>>>>>> automatic?
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Also the public IP address shown in the nat_map status is not
>>>>>>>>>>>>>> the same as shown in the sofia profile internal/external printouts.  Why is
>>>>>>>>>>>>>> this?
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> I'm sure this is configuration, just not sure what to change.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Thanks!
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> --
>>>>>>>>>>>>> -Rupa
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> --
>>>>>>>>>>>> -Rupa
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> --
>>>>>>>>>> -Rupa
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> -Rupa
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> -Rupa
>>>>>  _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mcampbellsmith at gmail.com  Wed Oct 27 05:12:57 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 27 Oct 2010 23:12:57 +1100
Subject: [Freeswitch-users] FS not detecting public IP address change
In-Reply-To: <AANLkTi=KL7w=ZeudXyEPu3S78CadmM88HfFFrzkEkcO3@mail.gmail.com>
References: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>
	<AANLkTinEoTcLmUVcpras8-m8pdkfpE=_r0spz1S50iMF@mail.gmail.com>
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	<AANLkTikfpRi9Shxiqd2w5nz3ZHsBnrHTt+=rHHnvoDJn@mail.gmail.com>
	<AANLkTikH3JJCCpL5s_eF2qjjttnEetE0i=nPn87zhBUL@mail.gmail.com>
	<AANLkTimzkb7qY8nW3M2afTPn18=OxY6jQye8D9TKr9Ts@mail.gmail.com>
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	<AANLkTi=JgCME8u=27kUyLGU3cMP2nZpFpu7Zd5-4tpj0@mail.gmail.com>
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	<AF5D2E2E-F8A7-44AB-8EA0-D240A28A51F6@mgtech.com>
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Message-ID: <AANLkTi==Kt7miDW4s6wa-5swpmfW5LTWWsSJ1uk30v-s@mail.gmail.com>

Isn't this going to cause problems where sofia has the wrong Ext-RTP-IP and
Ext-SIP-IP addresses?

>From sofia status profile internal

RTP-IP                  192.168.1.120
Ext-RTP-IP              21x.xx.xxx.254
SIP-IP                  192.168.1.120
Ext-SIP-IP              21x.xx.xxx.254

freeswitch at internal> nat_map status
Nat Type: NAT-PMP, ExtIP: 21x.xx.xxx.56
port,proto,proto_num,sticky


On Wed, Oct 27, 2010 at 4:13 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> That gives me quite a bit....
>
> Can there be the case that nat_map status has a different IP address to
> what I would see in sofia status profile internal?  And does sofia
> reregister if there is detected a public IP address change?
>
> Thanks
>
> freeswitch at internal> nat_map reinit
> [DEBUG] libs/esl/src/esl.c:1141 esl_send() SEND
> api nat_map reinit
> console_execute: true
>
>
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type]
> = [api/response]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER
> [Content-Length] = [130]
> Nat Type: NAT-PMP, ExtIP: 210.xx.xxx.xx
> port,proto,proto_num,sticky
> 5060,udp,0,0
> 5060,tcp,1,0
> 5080,udp,0,0
> 5080,tcp,1,0
>
> 4 total.
>
> freeswitch at internal> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV
> HEADER [Content-Type] = [log/data]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER
> [Content-Length] = [68]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Level] =
> [6]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Text-Channel]
> = [0]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-File] =
> [switch_nat.c]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_nat_init]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Line] =
> [410]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [User-Data] =
> []
> [DEBUG] libs/esl/src/esl.c:1115 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 68
> Log-Level: 6
> Text-Channel: 0
> Log-File: switch_nat.c
> Log-Func: switch_nat_init
> Log-Line: 410
> User-Data: _undef_
>
> 2010-10-27 16:10:07.003208 [INFO] switch_nat.c:410 Scanning for NAT
>
> 2010-10-27 16:10:07.003208 [INFO] switch_nat.c:410 Scanning for NAT
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER
> [Content-Length] = [73]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Text-Channel]
> = [0]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-File] =
> [switch_nat.c]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Func] =
> [get_pmp_pubaddr]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Line] =
> [167]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [User-Data] =
> []
> [DEBUG] libs/esl/src/esl.c:1115 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 73
> Log-Level: 7
> Text-Channel: 0
> Log-File: switch_nat.c
> Log-Func: get_pmp_pubaddr
> Log-Line: 167
> User-Data: _undef_
>
> 2010-10-27 16:10:07.003208 [DEBUG] switch_nat.c:167 Checking for PMP 1/5
>
> 2010-10-27 16:10:07.003208 [DEBUG] switch_nat.c:167 Checking for PMP 1/5
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER
> [Content-Length] = [98]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Level] =
> [6]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Text-Channel]
> = [0]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-File] =
> [switch_nat.c]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_nat_init]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Line] =
> [423]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [User-Data] =
> []
> [DEBUG] libs/esl/src/esl.c:1115 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 98
> Log-Level: 6
> Text-Channel: 0
> Log-File: switch_nat.c
> Log-Func: switch_nat_init
> Log-Line: 423
> User-Data: _undef_
>
> 2010-10-27 16:10:07.013797 [INFO] switch_nat.c:423 NAT detected type: pmp,
> ExtIP: '210.xx.xxx.xx'
>
> 2010-10-27 16:10:07.013797 [INFO] switch_nat.c:423 NAT detected type: pmp,
> ExtIP: '210.xx.xxx.xx'
>
>
> On Wed, Oct 27, 2010 at 4:01 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> Then I have no idea why you aren't seeing upnp messages on the console.
>>  What do you see when you do a "nat_map reinit" ?
>>
>>
>> On Tue, Oct 26, 2010 at 8:09 PM, Mark Campbell-Smith <
>> mcampbellsmith at gmail.com> wrote:
>>
>>> I see debug from sofia.c, switch_core_state_machine, mod_sofia, diaplan
>>> parsing, sofia_glue, switch_rtp, switch_ivr_play_say etc.
>>>
>>>
>>>
>>> On Wed, Oct 27, 2010 at 11:41 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>>> Do you see ANY debug messages on your console from any other FS module?
>>>>  fsctl loglevel 7 sets the base log level, but not necessarily the log level
>>>> on your screen.
>>>>
>>>> /log 7
>>>>
>>>> sets the log level in fs_cli.
>>>>
>>>> Your screen should get VERY busy especially when processing a call.
>>>>
>>>> On Tue, Oct 26, 2010 at 4:21 PM, Mark Campbell-Smith <
>>>> mcampbellsmith at gmail.com> wrote:
>>>>
>>>>> I have it set to debug in switch.conf.xml and enable fsctl loglevel 7
>>>>> in fs_cli....
>>>>>
>>>>> :~$ cat /usr/local/freeswitch/conf/autoload_configs/switch.conf.xml |
>>>>> egrep "param.*loglevel"
>>>>>     <param name="loglevel" value="debug"/>
>>>>>
>>>>> With regards to sofia: how does it determine if the external IP has
>>>>> changed?  And when it does, does sofia initiate a reregister of all external
>>>>> sip providers?
>>>>>
>>>>> Thanks!
>>>>>
>>>>> On Wed, Oct 27, 2010 at 3:14 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>
>>>>>> What you said: "And nat_map status shows me which ports have been
>>>>>> forwarded and the external ip address (which is sometimes wrong).  I also
>>>>>> see this in the router gui." is also happening to me on a Linksys/Cisco
>>>>>> RV042 router. Can't keep FS working for more than 2-5 hours without doing a
>>>>>> restart or nat_map reinit. I think the routers upnp has a problem with dual
>>>>>> wans, I have one static and one dynamic DSL.
>>>>>>
>>>>>> On Oct 26, 2010, at 6:43 AM, Rupa Schomaker wrote:
>>>>>>
>>>>>> What do you have set for loglevel in switch.conf.xml?  My guess is
>>>>>> that you have it set to something other than debug.  If that is set to (say)
>>>>>> info, then info is as low as you can go.  Setting the console log level to
>>>>>> debug will not give you debug output.
>>>>>>
>>>>>> sofia is unaware of upnp so a keepalive packet should have no impact
>>>>>> on it's operation.  I'm not familiar enough with sofia to tell what triggers
>>>>>> that event.
>>>>>>
>>>>>> On Tue, Oct 26, 2010 at 6:24 AM, Mark Campbell-Smith <
>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>
>>>>>>> OK..  tcpdump gave me this and I still do not see anything on FS
>>>>>>> except for this line:
>>>>>>>
>>>>>>> 2010-10-26 22:02:51.235608 [DEBUG] sofia.c:957 nua_i_outbound:
>>>>>>> unknown event 8: 102 NAT binding changed
>>>>>>>
>>>>>>> Hypertext Transfer Protocol
>>>>>>>     NOTIFY * HTTP/1.1\r\n
>>>>>>>     HOST:239.255.255.250:1900\r\n
>>>>>>>     Cache-Control:max-age=120\r\n
>>>>>>>     Location:http://192.168.1.1:1278/rootDesc.xml\r\n<http://192.168.1.1:1278/rootDesc.xml%5Cr%5Cn>
>>>>>>>     Server: Tomato UPnP/1.0 MiniUPnPd/1.4\r\n
>>>>>>>     NT:urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>>>>>>>
>>>>>>>  USN:uuid:882dbe81-c28a-4992-baf7-410c05f1caf4::urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>>>>>>>     NTS:ssdp:alive\r\n
>>>>>>>     \r\n
>>>>>>>
>>>>>>> This looks the same as you posted before, so why do I get the unknown
>>>>>>> event 8 (is this related)?
>>>>>>>
>>>>>>> Thanks!
>>>>>>>
>>>>>>> On Tue, Oct 26, 2010 at 3:32 PM, Mark Campbell-Smith <
>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>
>>>>>>>> :)  that was the first thing I did, but I get nothing; no keep alive
>>>>>>>> packets.
>>>>>>>>
>>>>>>>> in the logfile at startup I see:
>>>>>>>>
>>>>>>>> 2010-10-26 10:19:45.484258 [INFO] switch_nat.c:410 Scanning for NAT
>>>>>>>> 2010-10-26 10:19:45.485991 [DEBUG] switch_nat.c:167 Checking for PMP
>>>>>>>> 1/5
>>>>>>>> 2010-10-26 10:19:45.488575 [INFO] switch_nat.c:423 NAT detected
>>>>>>>> type: pmp, ExtIP: '203.xxx.xxx.xxx'
>>>>>>>> 2010-10-26 10:19:45.490683 [DEBUG] switch_nat.c:256 NAT thread
>>>>>>>> configured
>>>>>>>> 2010-10-26 10:19:45.492180 [DEBUG] switch_nat.c:267 NAT thread
>>>>>>>> started
>>>>>>>>
>>>>>>>> And nat_map status shows me which ports have been forwarded and the
>>>>>>>> external ip address (which is sometimes wrong).  I also see this in the
>>>>>>>> router gui.
>>>>>>>>
>>>>>>>> But I never see the keep alive packets.
>>>>>>>>
>>>>>>>> On Tue, Oct 26, 2010 at 3:13 PM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>
>>>>>>>>> Just turn up console logging to level 7.
>>>>>>>>>
>>>>>>>>> /log 7 if using fs_cli
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Mon, Oct 25, 2010 at 6:53 PM, Mark Campbell-Smith <
>>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> Hi!
>>>>>>>>>>
>>>>>>>>>> How do I enable debugging to get the UPnP keep alive packets?
>>>>>>>>>>
>>>>>>>>>> Thanks!
>>>>>>>>>>
>>>>>>>>>> On Tue, Oct 26, 2010 at 1:15 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>>>
>>>>>>>>>>> I need the debug logging from the upnp stuff.  It should look
>>>>>>>>>>> something like:
>>>>>>>>>>>
>>>>>>>>>>> +OK log level 7 [7]
>>>>>>>>>>> freeswitch at internal> 2010-10-25 09:13:59.926861 [DEBUG]
>>>>>>>>>>> switch_nat.c:299 got UPnP keep alive packet:
>>>>>>>>>>> NOTIFY * HTTP/1.1
>>>>>>>>>>> HOST:239.255.255.250:1900
>>>>>>>>>>> Cache-Control:max-age=60
>>>>>>>>>>> Location:http://192.168.1.1:5000/rootDesc.xml
>>>>>>>>>>> Server: Tomato UPnP/1.0 MiniUPnPd/1.4
>>>>>>>>>>> NT:urn:schemas-upnp-org:service:WANIPConnection:1
>>>>>>>>>>>
>>>>>>>>>>> USN:uuid:00000000-0000-0000-0000-000000000000::urn:schemas-upnp-org:service:WANIPConnection:1
>>>>>>>>>>> NTS:ssdp:alive
>>>>>>>>>>>
>>>>>>>>>>> I also run tomato and haven't had trouble with the upnp support
>>>>>>>>>>> so at least we have that part working.
>>>>>>>>>>>
>>>>>>>>>>> You should see the above keepalive appear periodically every 30s
>>>>>>>>>>> or so.  You should see another set of messages when you terminate the DSL
>>>>>>>>>>> connection and when that dsl connection comes back online.
>>>>>>>>>>>
>>>>>>>>>>> On Sat, Oct 23, 2010 at 5:36 PM, Mark Campbell-Smith <
>>>>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Hi!
>>>>>>>>>>>>
>>>>>>>>>>>> The router is a ASUS router but it is running the tomato
>>>>>>>>>>>> firmware which runs miniupnd.
>>>>>>>>>>>>
>>>>>>>>>>>> In the debug I see the message: 2010-10-24 08:02:59.178918
>>>>>>>>>>>> [DEBUG] sofia.c:956 nua_i_outbound: unknown event 8: 101 NAT detected
>>>>>>>>>>>>
>>>>>>>>>>>> This is what I see below. I hope I enabled all debug messages.
>>>>>>>>>>>>  I thought I saw some xml type messages earlier, but not sure how I enabled
>>>>>>>>>>>> them.
>>>>>>>>>>>>
>>>>>>>>>>>> nta_outgoing: RTT is 67.958 ms
>>>>>>>>>>>> outbound(0xb6e07c00): NAT binding changed: [210.xx.xxx.xx]:5080
>>>>>>>>>>>> != [58.xxx.xx.xx]:5080
>>>>>>>>>>>> nua(0xb6e07c00): event i_outbound 102 NAT binding changed
>>>>>>>>>>>> nua: nua_application_event: entering
>>>>>>>>>>>> 2010-10-24 09:05:08.849525 [DEBUG] sofia.c:956 nua_i_outbound:
>>>>>>>>>>>> unknown event 8: 102 NAT binding changed
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> On Sun, Oct 24, 2010 at 1:08 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> btw: what router are you using?
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Sat, Oct 23, 2010 at 9:06 AM, Rupa Schomaker <rupa at rupa.com
>>>>>>>>>>>>> > wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>> In a upnp config, FS is depending on the router to notify it
>>>>>>>>>>>>>> of the ip address change.  It is possible the upnp message(s) aren't
>>>>>>>>>>>>>> formatted as expected.  If you turn on debug logging you'll get the upnp
>>>>>>>>>>>>>> messages on the console.  If you can pastebin the capture of those while
>>>>>>>>>>>>>> dropping and restarting the dsl connection I can maybe see if there is
>>>>>>>>>>>>>> something obvious going on.  The debug should also have messages related to
>>>>>>>>>>>>>> the processing of those upnp messages.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Sat, Oct 23, 2010 at 7:04 AM, Mark Campbell-Smith <
>>>>>>>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> HI!
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> I know this question must have been answered 100's of
>>>>>>>>>>>>>>> times....
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> My adsl is a bit dodgy at the moment and tends to go up and
>>>>>>>>>>>>>>> down; which means I get assigned a new ip address from my ISP.  FS is nat'd
>>>>>>>>>>>>>>> behind a upnp capable router.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> The problem is that FS does not seem to be detecting the
>>>>>>>>>>>>>>> change.   For both the internal and external profile, I have auto-nat setup,
>>>>>>>>>>>>>>> and I have tried stun and host settings in vars.conf.  When I issue
>>>>>>>>>>>>>>> a nat_map status I see my old IP address, and if I then issue a nat_map
>>>>>>>>>>>>>>> reinit, I see the new public IP address.  How can I get this to be
>>>>>>>>>>>>>>> automatic?
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Also the public IP address shown in the nat_map status is not
>>>>>>>>>>>>>>> the same as shown in the sofia profile internal/external printouts.  Why is
>>>>>>>>>>>>>>> this?
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> I'm sure this is configuration, just not sure what to change.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Thanks!
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> --
>>>>>>>>>>>>>> -Rupa
>>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> --
>>>>>>>>>>>>> -Rupa
>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> --
>>>>>>>>>>> -Rupa
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>> -Rupa
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> -Rupa
>>>>>>  _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From Nabble at slickdeals.endjunk.com  Wed Oct 27 07:37:07 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 27 Oct 2010 07:37:07 -0700 (PDT)
Subject: [Freeswitch-users] FreeSWITCH - how to enable channels like
 mod_skypopen for Yahoo, Google Talk?
In-Reply-To: <AANLkTimMfFohwpBhkyK+eVyF0qu86jUyVxUgOPcx-MgL@mail.gmail.com>
References: <AANLkTikFHXgu-nZ5NBDQ4xON4c2D1WRHX33wOSRjPdSB@mail.gmail.com>
	<AANLkTimMfFohwpBhkyK+eVyF0qu86jUyVxUgOPcx-MgL@mail.gmail.com>
Message-ID: <1288190227145-5678811.post@n2.nabble.com>



Steven Ayre wrote:
> Yahoo appears to accept SIP.  But it also looks like they use their
> own extensions on top of the SIP protocol for authentication (Y-Cookie
> header) so I don't think it'll work currently.
> http://wiki.freeswitch.org/wiki/Provider_Configuration:_Yahoo_Messenger
> http://blog.motiwala.com/2007/08/18/yahoo-messenger-voip-service-with-sip-phone
> 
> -Steve
Perhaps, it is not a bad idea for FS developers to start writing a mod_yahoo
to support a Yahoo SIP call.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-how-to-enable-channels-like-mod-skypopen-for-Yahoo-Google-Talk-tp5677527p5678811.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Wed Oct 27 08:13:55 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 27 Oct 2010 16:13:55 +0100
Subject: [Freeswitch-users] FreeSWITCH - how to enable channels like
 mod_skypopen for Yahoo, Google Talk?
In-Reply-To: <1288190227145-5678811.post@n2.nabble.com>
References: <AANLkTikFHXgu-nZ5NBDQ4xON4c2D1WRHX33wOSRjPdSB@mail.gmail.com>
	<AANLkTimMfFohwpBhkyK+eVyF0qu86jUyVxUgOPcx-MgL@mail.gmail.com>
	<1288190227145-5678811.post@n2.nabble.com>
Message-ID: <AANLkTimsrtfmF2-hbu1MnkDcdxvm8x=FmrVkoRuY-YMn@mail.gmail.com>

Since it's SIP it'd be better using the mod_sofia stack to avoid
duplicating the code that already exists. The only issue is connecting
to Yahoo Messenger and authenticating to get a valid cookie to use.
That would need a lot of customisations to be made to mod_sofia.

The only library I can see out there to connect to Yahoo to generate
the cookies is libyahoo2. That's not an option though as it's GPL,
which isn't compatible with FreeSWITCH's MPL licence.

-Steve


On 27 October 2010 15:37, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
>
> Steven Ayre wrote:
>> Yahoo appears to accept SIP. ?But it also looks like they use their
>> own extensions on top of the SIP protocol for authentication (Y-Cookie
>> header) so I don't think it'll work currently.
>> http://wiki.freeswitch.org/wiki/Provider_Configuration:_Yahoo_Messenger
>> http://blog.motiwala.com/2007/08/18/yahoo-messenger-voip-service-with-sip-phone
>>
>> -Steve
> Perhaps, it is not a bad idea for FS developers to start writing a mod_yahoo
> to support a Yahoo SIP call.
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-how-to-enable-channels-like-mod-skypopen-for-Yahoo-Google-Talk-tp5677527p5678811.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From rupa at rupa.com  Wed Oct 27 08:36:08 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 27 Oct 2010 10:36:08 -0500
Subject: [Freeswitch-users] FS not detecting public IP address change
In-Reply-To: <AANLkTi=KL7w=ZeudXyEPu3S78CadmM88HfFFrzkEkcO3@mail.gmail.com>
References: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>
	<AANLkTinEoTcLmUVcpras8-m8pdkfpE=_r0spz1S50iMF@mail.gmail.com>
	<AANLkTi=MoV27FW0Ukd6K+mqmgnkyNgb9hjFx-vBYV1ak@mail.gmail.com>
	<AANLkTimCRfKSWG+5OdQY0=HTt2kOqxwk+0vKvaBNdmAd@mail.gmail.com>
	<AANLkTikfpRi9Shxiqd2w5nz3ZHsBnrHTt+=rHHnvoDJn@mail.gmail.com>
	<AANLkTikH3JJCCpL5s_eF2qjjttnEetE0i=nPn87zhBUL@mail.gmail.com>
	<AANLkTimzkb7qY8nW3M2afTPn18=OxY6jQye8D9TKr9Ts@mail.gmail.com>
	<AANLkTimR2x5AQPNiDDt4fZ8F=z_Wep3TBVR9Lob_72OA@mail.gmail.com>
	<AANLkTi=JgCME8u=27kUyLGU3cMP2nZpFpu7Zd5-4tpj0@mail.gmail.com>
	<AANLkTikL=kvbfLQx3E-vCaErRM-ARnU3_YSAvKo67Uvy@mail.gmail.com>
	<AF5D2E2E-F8A7-44AB-8EA0-D240A28A51F6@mgtech.com>
	<AANLkTikkFaJ-EWipGij2UAzMpmH8JGyhj+bUNKSDS8T5@mail.gmail.com>
	<AANLkTin=hRwNSBLxTRdJ6SqcPedbHiDgAJPCFESM9nCT@mail.gmail.com>
	<AANLkTi=+ggi1L7oAhKGFFR8f=QaXhoj_fMGPJ1pUFk+v@mail.gmail.com>
	<AANLkTin9X9vXL0Sg8BNPO9opuEzSq=iLChwVypuSec2x@mail.gmail.com>
	<AANLkTi=KL7w=ZeudXyEPu3S78CadmM88HfFFrzkEkcO3@mail.gmail.com>
Message-ID: <AANLkTikbBi46_0mhWfp0BAbb_g-dez_sU6W1QCCW50XC@mail.gmail.com>

Ok, first things first.  Your router has both NAT-PMP and Upnp enabled.
 Turn off NAT-PMP and only have upnp.  This should solve most of your
issues.  It is possible there are some issues with NAT-PMP, as far as I know
it has only really been tested against apple hardware.

If the public IP changes, sofia profile(s) using that IP are told to
reinitialize -- unless you have that option turned off.

On Wed, Oct 27, 2010 at 12:13 AM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> That gives me quite a bit....
>
> Can there be the case that nat_map status has a different IP address to
> what I would see in sofia status profile internal?  And does sofia
> reregister if there is detected a public IP address change?
>
> Thanks
>
> freeswitch at internal> nat_map reinit
> [DEBUG] libs/esl/src/esl.c:1141 esl_send() SEND
> api nat_map reinit
> console_execute: true
>
>
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type]
> = [api/response]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER
> [Content-Length] = [130]
> Nat Type: NAT-PMP, ExtIP: 210.xx.xxx.xx
> port,proto,proto_num,sticky
> 5060,udp,0,0
> 5060,tcp,1,0
> 5080,udp,0,0
> 5080,tcp,1,0
>
> 4 total.
>
> freeswitch at internal> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV
> HEADER [Content-Type] = [log/data]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER
> [Content-Length] = [68]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Level] =
> [6]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Text-Channel]
> = [0]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-File] =
> [switch_nat.c]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_nat_init]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Line] =
> [410]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [User-Data] =
> []
> [DEBUG] libs/esl/src/esl.c:1115 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 68
> Log-Level: 6
> Text-Channel: 0
> Log-File: switch_nat.c
> Log-Func: switch_nat_init
> Log-Line: 410
> User-Data: _undef_
>
> 2010-10-27 16:10:07.003208 [INFO] switch_nat.c:410 Scanning for NAT
>
> 2010-10-27 16:10:07.003208 [INFO] switch_nat.c:410 Scanning for NAT
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER
> [Content-Length] = [73]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Text-Channel]
> = [0]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-File] =
> [switch_nat.c]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Func] =
> [get_pmp_pubaddr]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Line] =
> [167]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [User-Data] =
> []
> [DEBUG] libs/esl/src/esl.c:1115 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 73
> Log-Level: 7
> Text-Channel: 0
> Log-File: switch_nat.c
> Log-Func: get_pmp_pubaddr
> Log-Line: 167
> User-Data: _undef_
>
> 2010-10-27 16:10:07.003208 [DEBUG] switch_nat.c:167 Checking for PMP 1/5
>
> 2010-10-27 16:10:07.003208 [DEBUG] switch_nat.c:167 Checking for PMP 1/5
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER
> [Content-Length] = [98]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Level] =
> [6]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Text-Channel]
> = [0]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-File] =
> [switch_nat.c]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_nat_init]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Line] =
> [423]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [User-Data] =
> []
> [DEBUG] libs/esl/src/esl.c:1115 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 98
> Log-Level: 6
> Text-Channel: 0
> Log-File: switch_nat.c
> Log-Func: switch_nat_init
> Log-Line: 423
> User-Data: _undef_
>
> 2010-10-27 16:10:07.013797 [INFO] switch_nat.c:423 NAT detected type: pmp,
> ExtIP: '210.xx.xxx.xx'
>
> 2010-10-27 16:10:07.013797 [INFO] switch_nat.c:423 NAT detected type: pmp,
> ExtIP: '210.xx.xxx.xx'
>
>
> On Wed, Oct 27, 2010 at 4:01 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> Then I have no idea why you aren't seeing upnp messages on the console.
>>  What do you see when you do a "nat_map reinit" ?
>>
>>
>> On Tue, Oct 26, 2010 at 8:09 PM, Mark Campbell-Smith <
>> mcampbellsmith at gmail.com> wrote:
>>
>>> I see debug from sofia.c, switch_core_state_machine, mod_sofia, diaplan
>>> parsing, sofia_glue, switch_rtp, switch_ivr_play_say etc.
>>>
>>>
>>>
>>> On Wed, Oct 27, 2010 at 11:41 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>>> Do you see ANY debug messages on your console from any other FS module?
>>>>  fsctl loglevel 7 sets the base log level, but not necessarily the log level
>>>> on your screen.
>>>>
>>>> /log 7
>>>>
>>>> sets the log level in fs_cli.
>>>>
>>>> Your screen should get VERY busy especially when processing a call.
>>>>
>>>> On Tue, Oct 26, 2010 at 4:21 PM, Mark Campbell-Smith <
>>>> mcampbellsmith at gmail.com> wrote:
>>>>
>>>>> I have it set to debug in switch.conf.xml and enable fsctl loglevel 7
>>>>> in fs_cli....
>>>>>
>>>>> :~$ cat /usr/local/freeswitch/conf/autoload_configs/switch.conf.xml |
>>>>> egrep "param.*loglevel"
>>>>>     <param name="loglevel" value="debug"/>
>>>>>
>>>>> With regards to sofia: how does it determine if the external IP has
>>>>> changed?  And when it does, does sofia initiate a reregister of all external
>>>>> sip providers?
>>>>>
>>>>> Thanks!
>>>>>
>>>>> On Wed, Oct 27, 2010 at 3:14 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>
>>>>>> What you said: "And nat_map status shows me which ports have been
>>>>>> forwarded and the external ip address (which is sometimes wrong).  I also
>>>>>> see this in the router gui." is also happening to me on a Linksys/Cisco
>>>>>> RV042 router. Can't keep FS working for more than 2-5 hours without doing a
>>>>>> restart or nat_map reinit. I think the routers upnp has a problem with dual
>>>>>> wans, I have one static and one dynamic DSL.
>>>>>>
>>>>>> On Oct 26, 2010, at 6:43 AM, Rupa Schomaker wrote:
>>>>>>
>>>>>> What do you have set for loglevel in switch.conf.xml?  My guess is
>>>>>> that you have it set to something other than debug.  If that is set to (say)
>>>>>> info, then info is as low as you can go.  Setting the console log level to
>>>>>> debug will not give you debug output.
>>>>>>
>>>>>> sofia is unaware of upnp so a keepalive packet should have no impact
>>>>>> on it's operation.  I'm not familiar enough with sofia to tell what triggers
>>>>>> that event.
>>>>>>
>>>>>> On Tue, Oct 26, 2010 at 6:24 AM, Mark Campbell-Smith <
>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>
>>>>>>> OK..  tcpdump gave me this and I still do not see anything on FS
>>>>>>> except for this line:
>>>>>>>
>>>>>>> 2010-10-26 22:02:51.235608 [DEBUG] sofia.c:957 nua_i_outbound:
>>>>>>> unknown event 8: 102 NAT binding changed
>>>>>>>
>>>>>>> Hypertext Transfer Protocol
>>>>>>>     NOTIFY * HTTP/1.1\r\n
>>>>>>>     HOST:239.255.255.250:1900\r\n
>>>>>>>     Cache-Control:max-age=120\r\n
>>>>>>>     Location:http://192.168.1.1:1278/rootDesc.xml\r\n<http://192.168.1.1:1278/rootDesc.xml%5Cr%5Cn>
>>>>>>>     Server: Tomato UPnP/1.0 MiniUPnPd/1.4\r\n
>>>>>>>     NT:urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>>>>>>>
>>>>>>>  USN:uuid:882dbe81-c28a-4992-baf7-410c05f1caf4::urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>>>>>>>     NTS:ssdp:alive\r\n
>>>>>>>     \r\n
>>>>>>>
>>>>>>> This looks the same as you posted before, so why do I get the unknown
>>>>>>> event 8 (is this related)?
>>>>>>>
>>>>>>> Thanks!
>>>>>>>
>>>>>>> On Tue, Oct 26, 2010 at 3:32 PM, Mark Campbell-Smith <
>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>
>>>>>>>> :)  that was the first thing I did, but I get nothing; no keep alive
>>>>>>>> packets.
>>>>>>>>
>>>>>>>> in the logfile at startup I see:
>>>>>>>>
>>>>>>>> 2010-10-26 10:19:45.484258 [INFO] switch_nat.c:410 Scanning for NAT
>>>>>>>> 2010-10-26 10:19:45.485991 [DEBUG] switch_nat.c:167 Checking for PMP
>>>>>>>> 1/5
>>>>>>>> 2010-10-26 10:19:45.488575 [INFO] switch_nat.c:423 NAT detected
>>>>>>>> type: pmp, ExtIP: '203.xxx.xxx.xxx'
>>>>>>>> 2010-10-26 10:19:45.490683 [DEBUG] switch_nat.c:256 NAT thread
>>>>>>>> configured
>>>>>>>> 2010-10-26 10:19:45.492180 [DEBUG] switch_nat.c:267 NAT thread
>>>>>>>> started
>>>>>>>>
>>>>>>>> And nat_map status shows me which ports have been forwarded and the
>>>>>>>> external ip address (which is sometimes wrong).  I also see this in the
>>>>>>>> router gui.
>>>>>>>>
>>>>>>>> But I never see the keep alive packets.
>>>>>>>>
>>>>>>>> On Tue, Oct 26, 2010 at 3:13 PM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>
>>>>>>>>> Just turn up console logging to level 7.
>>>>>>>>>
>>>>>>>>> /log 7 if using fs_cli
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Mon, Oct 25, 2010 at 6:53 PM, Mark Campbell-Smith <
>>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> Hi!
>>>>>>>>>>
>>>>>>>>>> How do I enable debugging to get the UPnP keep alive packets?
>>>>>>>>>>
>>>>>>>>>> Thanks!
>>>>>>>>>>
>>>>>>>>>> On Tue, Oct 26, 2010 at 1:15 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>>>
>>>>>>>>>>> I need the debug logging from the upnp stuff.  It should look
>>>>>>>>>>> something like:
>>>>>>>>>>>
>>>>>>>>>>> +OK log level 7 [7]
>>>>>>>>>>> freeswitch at internal> 2010-10-25 09:13:59.926861 [DEBUG]
>>>>>>>>>>> switch_nat.c:299 got UPnP keep alive packet:
>>>>>>>>>>> NOTIFY * HTTP/1.1
>>>>>>>>>>> HOST:239.255.255.250:1900
>>>>>>>>>>> Cache-Control:max-age=60
>>>>>>>>>>> Location:http://192.168.1.1:5000/rootDesc.xml
>>>>>>>>>>> Server: Tomato UPnP/1.0 MiniUPnPd/1.4
>>>>>>>>>>> NT:urn:schemas-upnp-org:service:WANIPConnection:1
>>>>>>>>>>>
>>>>>>>>>>> USN:uuid:00000000-0000-0000-0000-000000000000::urn:schemas-upnp-org:service:WANIPConnection:1
>>>>>>>>>>> NTS:ssdp:alive
>>>>>>>>>>>
>>>>>>>>>>> I also run tomato and haven't had trouble with the upnp support
>>>>>>>>>>> so at least we have that part working.
>>>>>>>>>>>
>>>>>>>>>>> You should see the above keepalive appear periodically every 30s
>>>>>>>>>>> or so.  You should see another set of messages when you terminate the DSL
>>>>>>>>>>> connection and when that dsl connection comes back online.
>>>>>>>>>>>
>>>>>>>>>>> On Sat, Oct 23, 2010 at 5:36 PM, Mark Campbell-Smith <
>>>>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Hi!
>>>>>>>>>>>>
>>>>>>>>>>>> The router is a ASUS router but it is running the tomato
>>>>>>>>>>>> firmware which runs miniupnd.
>>>>>>>>>>>>
>>>>>>>>>>>> In the debug I see the message: 2010-10-24 08:02:59.178918
>>>>>>>>>>>> [DEBUG] sofia.c:956 nua_i_outbound: unknown event 8: 101 NAT detected
>>>>>>>>>>>>
>>>>>>>>>>>> This is what I see below. I hope I enabled all debug messages.
>>>>>>>>>>>>  I thought I saw some xml type messages earlier, but not sure how I enabled
>>>>>>>>>>>> them.
>>>>>>>>>>>>
>>>>>>>>>>>> nta_outgoing: RTT is 67.958 ms
>>>>>>>>>>>> outbound(0xb6e07c00): NAT binding changed: [210.xx.xxx.xx]:5080
>>>>>>>>>>>> != [58.xxx.xx.xx]:5080
>>>>>>>>>>>> nua(0xb6e07c00): event i_outbound 102 NAT binding changed
>>>>>>>>>>>> nua: nua_application_event: entering
>>>>>>>>>>>> 2010-10-24 09:05:08.849525 [DEBUG] sofia.c:956 nua_i_outbound:
>>>>>>>>>>>> unknown event 8: 102 NAT binding changed
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> On Sun, Oct 24, 2010 at 1:08 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> btw: what router are you using?
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Sat, Oct 23, 2010 at 9:06 AM, Rupa Schomaker <rupa at rupa.com
>>>>>>>>>>>>> > wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>> In a upnp config, FS is depending on the router to notify it
>>>>>>>>>>>>>> of the ip address change.  It is possible the upnp message(s) aren't
>>>>>>>>>>>>>> formatted as expected.  If you turn on debug logging you'll get the upnp
>>>>>>>>>>>>>> messages on the console.  If you can pastebin the capture of those while
>>>>>>>>>>>>>> dropping and restarting the dsl connection I can maybe see if there is
>>>>>>>>>>>>>> something obvious going on.  The debug should also have messages related to
>>>>>>>>>>>>>> the processing of those upnp messages.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Sat, Oct 23, 2010 at 7:04 AM, Mark Campbell-Smith <
>>>>>>>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> HI!
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> I know this question must have been answered 100's of
>>>>>>>>>>>>>>> times....
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> My adsl is a bit dodgy at the moment and tends to go up and
>>>>>>>>>>>>>>> down; which means I get assigned a new ip address from my ISP.  FS is nat'd
>>>>>>>>>>>>>>> behind a upnp capable router.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> The problem is that FS does not seem to be detecting the
>>>>>>>>>>>>>>> change.   For both the internal and external profile, I have auto-nat setup,
>>>>>>>>>>>>>>> and I have tried stun and host settings in vars.conf.  When I issue
>>>>>>>>>>>>>>> a nat_map status I see my old IP address, and if I then issue a nat_map
>>>>>>>>>>>>>>> reinit, I see the new public IP address.  How can I get this to be
>>>>>>>>>>>>>>> automatic?
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Also the public IP address shown in the nat_map status is not
>>>>>>>>>>>>>>> the same as shown in the sofia profile internal/external printouts.  Why is
>>>>>>>>>>>>>>> this?
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> I'm sure this is configuration, just not sure what to change.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Thanks!
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> --
>>>>>>>>>>>>>> -Rupa
>>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> --
>>>>>>>>>>>>> -Rupa
>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> --
>>>>>>>>>>> -Rupa
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>> -Rupa
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> -Rupa
>>>>>>  _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From mario_fs at mgtech.com  Wed Oct 27 08:43:21 2010
From: mario_fs at mgtech.com (Mario G)
Date: Wed, 27 Oct 2010 08:43:21 -0700
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <9BAFDAF8-1E66-444C-A024-A88FF65D8BCA@mgtech.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
	<407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>
	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
	<AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
	<6D9FD5E2-21FA-4630-8CF4-A87ADB30AC87@mgtech.com>
	<AANLkTi=BiDiXLgEoj_MfwmZbGjvXoOuNJgUYdetciNCj@mail.gmail.com>
	<9BAFDAF8-1E66-444C-A024-A88FF65D8BCA@mgtech.com>
Message-ID: <3447E946-0BF8-4F7E-B5E9-E28F53DD0968@mgtech.com>

I should mention that I did not have this problem with an SPA9000 PBX (asterisk based) for over two years so FS may be pickier about upnp and/or nat, or just better at it exposing a problem in the router.

I made different changes to the gateways to test different things. One failed after 17 hours, the other two stayed up.  What did not work:
added <variable name="sip-force-expires" value="30"/> to the directory entries as suggested.
set the gateway expire times to 30 seconds.

What worked (could be coincidental) for the two gateways that stayed up: I Added <param name="aggressive-nat-detection" value="true"/>

I originally setup FS to use the static ip by setting external sip/rtp to just the static ip (no autonat:) and ran with -nonat but I could not get incoming calls. The only way it worked was to use autonat:1.2.3.4. The router has 1 static public address and 1 dynamic external IP, this is the root of the problem, upnp only tells FS about the dynamic ip  Will keep this thread up-to-date for anyone who may be in the same boat someday. Thanks again for looking at the trace.
Mario

>> You should be setting the req freq to a low number on the outbound gateways
>> The examples you showed had a series of inbound reg
>> 
>> also set expire-seconds to 30 in your gateway xml
>> 
>> 
>> The problem is if you are not constantly sending traffic to the box
>> the nat mapping will go away.
>> 
>> If you are in production you should be using a static ip with a static
>> mapping, any trouble you are having is your own fault for playing with
>> fire.  The best we can do is tell you how to keep it contained.
>> 
>> 
>> 
>> 
>> On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com> wrote:
>>> I made the change. I had no idea the settings for the inside phones effected nat for the outside sip accounts. I was looking into aggressive-nat- detection since the internal profile status always shows the right external static IP but the nat_ap status always shows the dynamic ip. Crossing fingers/etc since this problem is 85% of time (weeks!) into FS changeover. Thanks!
>>> Mario
>>> 
>>> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
>>> 
>>>> add
>>>> 
>>>> <variable name="sip-force-expires" value="30"/>
>>>> 
>>>> to the <variables> section of your <user>
>>>> 
>>>> you have it at 600 and the nat mapping is timing out while the 600
>>>> seconds is ticking away
>>>> 
>>>> 
>>>> 
>>>> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>> From the TSP:
>>>>> "I have enabled the SIP trace on your account. We are not currently seeing
>>>>> any registration attempts to your account within the last 15 minutes. Please
>>>>> restart FreeSwitch so that registration attempts begin again. Thank you. ".
>>>>> So FS is not getting past router.
>>>>> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>>>>> 
>>>>> I ran the global trace during the problem and it is
>>>>> at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
>>>>> "accttwo", "acct3". The trace includes phones since it was global. I am
>>>>> using:
>>>>>    <param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>>>>>    <param name="ext-sip-ip" value="autonat:my-static.ip"/>
>>>>> I tried dumping nat and removing the autonat: above and using -nonat but
>>>>> that did not work, registration proceeded but no calls inbound.
>>>>> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>>>>> 
>>>>> Whoops, I am using an IP address for at least one gateway so that is not the
>>>>> problem:
>>>>> They look outbound to me and I am using dns for 2 and an IP for one so that
>>>>> is not the issue. I was able to get FS to clear this up by doing "nat_map
>>>>> reinit" which is why I think this is a nat problem. I will do the trace you
>>>>> mentioned. I will plug an ip address into one of the gateways to see what
>>>>> happens, they all fail at once. Thanks for responding!
>>>>> Mario
>>>>> 
>>>>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>>>>> 
>>>>> I really need help on this as I have weeks into this problem. I thought I
>>>>> had it nailed but I guess not. After 5.5 hours I get:
>>>>> 
>>>>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>>>>> Registration, setting retry to 15 seconds.
>>>>> 
>>>>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>>> 
>>>>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>>> 
>>>>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
>>>>> Registration, setting retry to 30 seconds.
>>>>> 
>>>>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>>>>> Registration, setting retry to 15 seconds.
>>>>> 
>>>>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>>>>> Registration, setting retry to 15 seconds.
>>>>> 
>>>>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>>> 
>>>>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>>> 
>>>>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>>>>> 
>>>>> and no way to make/get calls until I restart FS. I did this:
>>>>> 
>>>>> 1. log 7
>>>>> 
>>>>> 2. sofia profile xxxx siptrace on   for each profile/gateway
>>>>> 
>>>>> 3. restarted router
>>>>> 
>>>>> All three did not solve the problem. The trace and log produced no
>>>>> additional lines which is why I am wondering if FS has a problem since the
>>>>> trace shows no SIP activity.
>>>>> 
>>>>> 3 gateways with 2 ITSPs
>>>>> 
>>>>> 2 DSL/WAN lines, 1 static and 1 dynamic
>>>>> 
>>>>> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
>>>>> external static ip.
>>>>> 
>>>>> sofia status profile ... has the right ext ip
>>>>> 
>>>>> nat_map status shows the dynamic (wrong) IP
>>>>> 
>>>>> I tried starting with -nonat but that was worse
>>>>> 
>>>>> the only way to fix is restart FS.
>>>>> 
>>>>> I read the wiki on external nat, auto_nat and everything else many times.
>>>>> 
>>>>> Thanks Mario
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> 
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> 
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
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>>>>> 
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _______________________________________________
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>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>>>>> 
>>>>> _______________________________________________
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>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> 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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>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> _______________________________________________
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>>> 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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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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> 
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From mario_fs at mgtech.com  Wed Oct 27 08:47:58 2010
From: mario_fs at mgtech.com (Mario G)
Date: Wed, 27 Oct 2010 08:47:58 -0700
Subject: [Freeswitch-users] FS not detecting public IP address change
In-Reply-To: <AANLkTi=KL7w=ZeudXyEPu3S78CadmM88HfFFrzkEkcO3@mail.gmail.com>
References: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>
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	<AANLkTikfpRi9Shxiqd2w5nz3ZHsBnrHTt+=rHHnvoDJn@mail.gmail.com>
	<AANLkTikH3JJCCpL5s_eF2qjjttnEetE0i=nPn87zhBUL@mail.gmail.com>
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	<AF5D2E2E-F8A7-44AB-8EA0-D240A28A51F6@mgtech.com>
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Message-ID: <BC3127AD-18EB-4A1E-9A80-9A2BF4C55FE5@mgtech.com>

FYI, same thing I seeing on my R042 problem. sofia status shows correct (static) external IP in internal and gateway profiles, but nat_map has the dynamic IP.

On Oct 26, 2010, at 10:13 PM, Mark Campbell-Smith wrote:

> That gives me quite a bit....
> 
> Can there be the case that nat_map status has a different IP address to what I would see in sofia status profile internal?  And does sofia reregister if there is detected a public IP address change?
> 
> Thanks
> 
> freeswitch at internal> nat_map reinit
> [DEBUG] libs/esl/src/esl.c:1141 esl_send() SEND
> api nat_map reinit
> console_execute: true
> 
> 
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type] = [api/response]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Length] = [130]
> Nat Type: NAT-PMP, ExtIP: 210.xx.xxx.xx
> port,proto,proto_num,sticky
> 5060,udp,0,0
> 5060,tcp,1,0
> 5080,udp,0,0
> 5080,tcp,1,0
> 
> 4 total.
> 
> freeswitch at internal> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type] = [log/data]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Length] = [68]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Level] = [6]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Text-Channel] = [0]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-File] = [switch_nat.c]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Func] = [switch_nat_init]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Line] = [410]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [User-Data] = []
> [DEBUG] libs/esl/src/esl.c:1115 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 68
> Log-Level: 6
> Text-Channel: 0
> Log-File: switch_nat.c
> Log-Func: switch_nat_init
> Log-Line: 410
> User-Data: _undef_
> 
> 2010-10-27 16:10:07.003208 [INFO] switch_nat.c:410 Scanning for NAT
> 
> 2010-10-27 16:10:07.003208 [INFO] switch_nat.c:410 Scanning for NAT
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type] = [log/data]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Length] = [73]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Level] = [7]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Text-Channel] = [0]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-File] = [switch_nat.c]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Func] = [get_pmp_pubaddr]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Line] = [167]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [User-Data] = []
> [DEBUG] libs/esl/src/esl.c:1115 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 73
> Log-Level: 7
> Text-Channel: 0
> Log-File: switch_nat.c
> Log-Func: get_pmp_pubaddr
> Log-Line: 167
> User-Data: _undef_
> 
> 2010-10-27 16:10:07.003208 [DEBUG] switch_nat.c:167 Checking for PMP 1/5
> 
> 2010-10-27 16:10:07.003208 [DEBUG] switch_nat.c:167 Checking for PMP 1/5
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type] = [log/data]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Length] = [98]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Level] = [6]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Text-Channel] = [0]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-File] = [switch_nat.c]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Func] = [switch_nat_init]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Line] = [423]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [User-Data] = []
> [DEBUG] libs/esl/src/esl.c:1115 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 98
> Log-Level: 6
> Text-Channel: 0
> Log-File: switch_nat.c
> Log-Func: switch_nat_init
> Log-Line: 423
> User-Data: _undef_
> 
> 2010-10-27 16:10:07.013797 [INFO] switch_nat.c:423 NAT detected type: pmp, ExtIP: '210.xx.xxx.xx'
> 
> 2010-10-27 16:10:07.013797 [INFO] switch_nat.c:423 NAT detected type: pmp, ExtIP: '210.xx.xxx.xx'
> 
> 
> On Wed, Oct 27, 2010 at 4:01 PM, Rupa Schomaker <rupa at rupa.com> wrote:
> Then I have no idea why you aren't seeing upnp messages on the console.  What do you see when you do a "nat_map reinit" ?
> 
> 
> On Tue, Oct 26, 2010 at 8:09 PM, Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
> I see debug from sofia.c, switch_core_state_machine, mod_sofia, diaplan parsing, sofia_glue, switch_rtp, switch_ivr_play_say etc.
> 
> 
> 
> On Wed, Oct 27, 2010 at 11:41 AM, Rupa Schomaker <rupa at rupa.com> wrote:
> Do you see ANY debug messages on your console from any other FS module?  fsctl loglevel 7 sets the base log level, but not necessarily the log level on your screen.
> 
> /log 7
> 
> sets the log level in fs_cli.
> 
> Your screen should get VERY busy especially when processing a call.
> 
> On Tue, Oct 26, 2010 at 4:21 PM, Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
> I have it set to debug in switch.conf.xml and enable fsctl loglevel 7 in fs_cli....
> 
> :~$ cat /usr/local/freeswitch/conf/autoload_configs/switch.conf.xml | egrep "param.*loglevel"
>     <param name="loglevel" value="debug"/>
> 
> With regards to sofia: how does it determine if the external IP has changed?  And when it does, does sofia initiate a reregister of all external sip providers?
> 
> Thanks!
> 
> On Wed, Oct 27, 2010 at 3:14 AM, Mario G <mario_fs at mgtech.com> wrote:
> What you said: "And nat_map status shows me which ports have been forwarded and the external ip address (which is sometimes wrong).  I also see this in the router gui." is also happening to me on a Linksys/Cisco RV042 router. Can't keep FS working for more than 2-5 hours without doing a restart or nat_map reinit. I think the routers upnp has a problem with dual wans, I have one static and one dynamic DSL.
> 
> On Oct 26, 2010, at 6:43 AM, Rupa Schomaker wrote:
> 
>> What do you have set for loglevel in switch.conf.xml?  My guess is that you have it set to something other than debug.  If that is set to (say) info, then info is as low as you can go.  Setting the console log level to debug will not give you debug output.
>> 
>> sofia is unaware of upnp so a keepalive packet should have no impact on it's operation.  I'm not familiar enough with sofia to tell what triggers that event.
>> 
>> On Tue, Oct 26, 2010 at 6:24 AM, Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
>> OK..  tcpdump gave me this and I still do not see anything on FS except for this line:
>> 
>> 2010-10-26 22:02:51.235608 [DEBUG] sofia.c:957 nua_i_outbound: unknown event 8: 102 NAT binding changed
>> 
>> Hypertext Transfer Protocol
>>     NOTIFY * HTTP/1.1\r\n
>>     HOST:239.255.255.250:1900\r\n
>>     Cache-Control:max-age=120\r\n
>>     Location:http://192.168.1.1:1278/rootDesc.xml\r\n
>>     Server: Tomato UPnP/1.0 MiniUPnPd/1.4\r\n
>>     NT:urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>>     USN:uuid:882dbe81-c28a-4992-baf7-410c05f1caf4::urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>>     NTS:ssdp:alive\r\n
>>     \r\n
>> 
>> This looks the same as you posted before, so why do I get the unknown event 8 (is this related)?
>> 
>> Thanks!
>> 
>> On Tue, Oct 26, 2010 at 3:32 PM, Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
>> :)  that was the first thing I did, but I get nothing; no keep alive packets.
>> 
>> in the logfile at startup I see:
>> 
>> 2010-10-26 10:19:45.484258 [INFO] switch_nat.c:410 Scanning for NAT
>> 2010-10-26 10:19:45.485991 [DEBUG] switch_nat.c:167 Checking for PMP 1/5
>> 2010-10-26 10:19:45.488575 [INFO] switch_nat.c:423 NAT detected type: pmp, ExtIP: '203.xxx.xxx.xxx'
>> 2010-10-26 10:19:45.490683 [DEBUG] switch_nat.c:256 NAT thread configured
>> 2010-10-26 10:19:45.492180 [DEBUG] switch_nat.c:267 NAT thread started
>> 
>> And nat_map status shows me which ports have been forwarded and the external ip address (which is sometimes wrong).  I also see this in the router gui.  
>> 
>> But I never see the keep alive packets.
>> 
>> On Tue, Oct 26, 2010 at 3:13 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>> Just turn up console logging to level 7.
>> 
>> /log 7 if using fs_cli
>> 
>> 
>> On Mon, Oct 25, 2010 at 6:53 PM, Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
>> Hi!
>> 
>> How do I enable debugging to get the UPnP keep alive packets?
>> 
>> Thanks!
>> 
>> On Tue, Oct 26, 2010 at 1:15 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>> I need the debug logging from the upnp stuff.  It should look something like:
>> 
>> +OK log level 7 [7]
>> freeswitch at internal> 2010-10-25 09:13:59.926861 [DEBUG] switch_nat.c:299 got UPnP keep alive packet:
>> NOTIFY * HTTP/1.1
>> HOST:239.255.255.250:1900
>> Cache-Control:max-age=60
>> Location:http://192.168.1.1:5000/rootDesc.xml
>> Server: Tomato UPnP/1.0 MiniUPnPd/1.4
>> NT:urn:schemas-upnp-org:service:WANIPConnection:1
>> USN:uuid:00000000-0000-0000-0000-000000000000::urn:schemas-upnp-org:service:WANIPConnection:1
>> NTS:ssdp:alive
>> 
>> I also run tomato and haven't had trouble with the upnp support so at least we have that part working.
>> 
>> You should see the above keepalive appear periodically every 30s or so.  You should see another set of messages when you terminate the DSL connection and when that dsl connection comes back online.
>> 
>> On Sat, Oct 23, 2010 at 5:36 PM, Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
>> Hi!
>> 
>> The router is a ASUS router but it is running the tomato firmware which runs miniupnd.
>> 
>> In the debug I see the message: 2010-10-24 08:02:59.178918 [DEBUG] sofia.c:956 nua_i_outbound: unknown event 8: 101 NAT detected
>> 
>> This is what I see below. I hope I enabled all debug messages.  I thought I saw some xml type messages earlier, but not sure how I enabled them.
>> 
>> nta_outgoing: RTT is 67.958 ms
>> outbound(0xb6e07c00): NAT binding changed: [210.xx.xxx.xx]:5080 != [58.xxx.xx.xx]:5080
>> nua(0xb6e07c00): event i_outbound 102 NAT binding changed
>> nua: nua_application_event: entering
>> 2010-10-24 09:05:08.849525 [DEBUG] sofia.c:956 nua_i_outbound: unknown event 8: 102 NAT binding changed
>> 
>> 
>> On Sun, Oct 24, 2010 at 1:08 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>> btw: what router are you using?
>> 
>> 
>> On Sat, Oct 23, 2010 at 9:06 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>> In a upnp config, FS is depending on the router to notify it of the ip address change.  It is possible the upnp message(s) aren't formatted as expected.  If you turn on debug logging you'll get the upnp messages on the console.  If you can pastebin the capture of those while dropping and restarting the dsl connection I can maybe see if there is something obvious going on.  The debug should also have messages related to the processing of those upnp messages.
>> 
>> On Sat, Oct 23, 2010 at 7:04 AM, Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
>> HI!
>> 
>> I know this question must have been answered 100's of times....
>> 
>> My adsl is a bit dodgy at the moment and tends to go up and down; which means I get assigned a new ip address from my ISP.  FS is nat'd behind a upnp capable router.
>> 
>> The problem is that FS does not seem to be detecting the change.   For both the internal and external profile, I have auto-nat setup, and I have tried stun and host settings in vars.conf.  When I issue a nat_map status I see my old IP address, and if I then issue a nat_map reinit, I see the new public IP address.  How can I get this to be automatic?
>> 
>> Also the public IP address shown in the nat_map status is not the same as shown in the sofia profile internal/external printouts.  Why is this?
>> 
>> I'm sure this is configuration, just not sure what to change.
>> 
>> Thanks!
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
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>> 
>> 
>> 
>> -- 
>> -Rupa
>> 
>> 
>> 
>> -- 
>> -Rupa
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
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>> 
>> 
>> _______________________________________________
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>> 
>> 
>> 
>> 
>> -- 
>> -Rupa
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
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>> 
>> 
>> _______________________________________________
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>> 
>> 
>> 
>> -- 
>> -Rupa
>> 
>> _______________________________________________
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>> 
>> 
>> 
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>> 
>> 
>> 
>> -- 
>> -Rupa
>> _______________________________________________
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> 
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> 
> 
> 
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> -Rupa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> 
> 
> 
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> 
> 
> 
> -- 
> -Rupa
> 
> _______________________________________________
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From rupa at rupa.com  Wed Oct 27 09:03:38 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 27 Oct 2010 11:03:38 -0500
Subject: [Freeswitch-users] FS not detecting public IP address change
In-Reply-To: <BC3127AD-18EB-4A1E-9A80-9A2BF4C55FE5@mgtech.com>
References: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>
	<AANLkTinEoTcLmUVcpras8-m8pdkfpE=_r0spz1S50iMF@mail.gmail.com>
	<AANLkTi=MoV27FW0Ukd6K+mqmgnkyNgb9hjFx-vBYV1ak@mail.gmail.com>
	<AANLkTimCRfKSWG+5OdQY0=HTt2kOqxwk+0vKvaBNdmAd@mail.gmail.com>
	<AANLkTikfpRi9Shxiqd2w5nz3ZHsBnrHTt+=rHHnvoDJn@mail.gmail.com>
	<AANLkTikH3JJCCpL5s_eF2qjjttnEetE0i=nPn87zhBUL@mail.gmail.com>
	<AANLkTimzkb7qY8nW3M2afTPn18=OxY6jQye8D9TKr9Ts@mail.gmail.com>
	<AANLkTimR2x5AQPNiDDt4fZ8F=z_Wep3TBVR9Lob_72OA@mail.gmail.com>
	<AANLkTi=JgCME8u=27kUyLGU3cMP2nZpFpu7Zd5-4tpj0@mail.gmail.com>
	<AANLkTikL=kvbfLQx3E-vCaErRM-ARnU3_YSAvKo67Uvy@mail.gmail.com>
	<AF5D2E2E-F8A7-44AB-8EA0-D240A28A51F6@mgtech.com>
	<AANLkTikkFaJ-EWipGij2UAzMpmH8JGyhj+bUNKSDS8T5@mail.gmail.com>
	<AANLkTin=hRwNSBLxTRdJ6SqcPedbHiDgAJPCFESM9nCT@mail.gmail.com>
	<AANLkTi=+ggi1L7oAhKGFFR8f=QaXhoj_fMGPJ1pUFk+v@mail.gmail.com>
	<AANLkTin9X9vXL0Sg8BNPO9opuEzSq=iLChwVypuSec2x@mail.gmail.com>
	<AANLkTi=KL7w=ZeudXyEPu3S78CadmM88HfFFrzkEkcO3@mail.gmail.com>
	<BC3127AD-18EB-4A1E-9A80-9A2BF4C55FE5@mgtech.com>
Message-ID: <AANLkTinvSh0DJJAWxv_A+=WgcVubu1u5EKQMo-kaKVDp@mail.gmail.com>

Mario, your issue is a different (unrelated) issue.

On Wed, Oct 27, 2010 at 10:47 AM, Mario G <mario_fs at mgtech.com> wrote:

> FYI, same thing I seeing on my R042 problem. sofia status shows correct
> (static) external IP in internal and gateway profiles, but nat_map has the
> dynamic IP.
>
> On Oct 26, 2010, at 10:13 PM, Mark Campbell-Smith wrote:
>
> That gives me quite a bit....
>
> Can there be the case that nat_map status has a different IP address to
> what I would see in sofia status profile internal?  And does sofia
> reregister if there is detected a public IP address change?
>
> Thanks
>
> freeswitch at internal> nat_map reinit
> [DEBUG] libs/esl/src/esl.c:1141 esl_send() SEND
> api nat_map reinit
> console_execute: true
>
>
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type]
> = [api/response]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER
> [Content-Length] = [130]
> Nat Type: NAT-PMP, ExtIP: 210.xx.xxx.xx
> port,proto,proto_num,sticky
> 5060,udp,0,0
> 5060,tcp,1,0
> 5080,udp,0,0
> 5080,tcp,1,0
>
> 4 total.
>
> freeswitch at internal> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV
> HEADER [Content-Type] = [log/data]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER
> [Content-Length] = [68]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Level] =
> [6]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Text-Channel]
> = [0]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-File] =
> [switch_nat.c]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_nat_init]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Line] =
> [410]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [User-Data] =
> []
> [DEBUG] libs/esl/src/esl.c:1115 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 68
> Log-Level: 6
> Text-Channel: 0
> Log-File: switch_nat.c
> Log-Func: switch_nat_init
> Log-Line: 410
> User-Data: _undef_
>
> 2010-10-27 16:10:07.003208 [INFO] switch_nat.c:410 Scanning for NAT
>
> 2010-10-27 16:10:07.003208 [INFO] switch_nat.c:410 Scanning for NAT
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER
> [Content-Length] = [73]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Level] =
> [7]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Text-Channel]
> = [0]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-File] =
> [switch_nat.c]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Func] =
> [get_pmp_pubaddr]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Line] =
> [167]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [User-Data] =
> []
> [DEBUG] libs/esl/src/esl.c:1115 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 73
> Log-Level: 7
> Text-Channel: 0
> Log-File: switch_nat.c
> Log-Func: get_pmp_pubaddr
> Log-Line: 167
> User-Data: _undef_
>
> 2010-10-27 16:10:07.003208 [DEBUG] switch_nat.c:167 Checking for PMP 1/5
>
> 2010-10-27 16:10:07.003208 [DEBUG] switch_nat.c:167 Checking for PMP 1/5
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type]
> = [log/data]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER
> [Content-Length] = [98]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Level] =
> [6]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Text-Channel]
> = [0]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-File] =
> [switch_nat.c]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Func] =
> [switch_nat_init]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Line] =
> [423]
> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [User-Data] =
> []
> [DEBUG] libs/esl/src/esl.c:1115 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: log/data
> Content-Length: 98
> Log-Level: 6
> Text-Channel: 0
> Log-File: switch_nat.c
> Log-Func: switch_nat_init
> Log-Line: 423
> User-Data: _undef_
>
> 2010-10-27 16:10:07.013797 [INFO] switch_nat.c:423 NAT detected type: pmp,
> ExtIP: '210.xx.xxx.xx'
>
> 2010-10-27 16:10:07.013797 [INFO] switch_nat.c:423 NAT detected type: pmp,
> ExtIP: '210.xx.xxx.xx'
>
>
> On Wed, Oct 27, 2010 at 4:01 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> Then I have no idea why you aren't seeing upnp messages on the console.
>>  What do you see when you do a "nat_map reinit" ?
>>
>>
>> On Tue, Oct 26, 2010 at 8:09 PM, Mark Campbell-Smith <
>> mcampbellsmith at gmail.com> wrote:
>>
>>> I see debug from sofia.c, switch_core_state_machine, mod_sofia, diaplan
>>> parsing, sofia_glue, switch_rtp, switch_ivr_play_say etc.
>>>
>>>
>>>
>>> On Wed, Oct 27, 2010 at 11:41 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>>> Do you see ANY debug messages on your console from any other FS module?
>>>>  fsctl loglevel 7 sets the base log level, but not necessarily the log level
>>>> on your screen.
>>>>
>>>> /log 7
>>>>
>>>> sets the log level in fs_cli.
>>>>
>>>> Your screen should get VERY busy especially when processing a call.
>>>>
>>>> On Tue, Oct 26, 2010 at 4:21 PM, Mark Campbell-Smith <
>>>> mcampbellsmith at gmail.com> wrote:
>>>>
>>>>> I have it set to debug in switch.conf.xml and enable fsctl loglevel 7
>>>>> in fs_cli....
>>>>>
>>>>> :~$ cat /usr/local/freeswitch/conf/autoload_configs/switch.conf.xml |
>>>>> egrep "param.*loglevel"
>>>>>     <param name="loglevel" value="debug"/>
>>>>>
>>>>> With regards to sofia: how does it determine if the external IP has
>>>>> changed?  And when it does, does sofia initiate a reregister of all external
>>>>> sip providers?
>>>>>
>>>>> Thanks!
>>>>>
>>>>> On Wed, Oct 27, 2010 at 3:14 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>
>>>>>> What you said: "And nat_map status shows me which ports have been
>>>>>> forwarded and the external ip address (which is sometimes wrong).  I also
>>>>>> see this in the router gui." is also happening to me on a Linksys/Cisco
>>>>>> RV042 router. Can't keep FS working for more than 2-5 hours without doing a
>>>>>> restart or nat_map reinit. I think the routers upnp has a problem with dual
>>>>>> wans, I have one static and one dynamic DSL.
>>>>>>
>>>>>> On Oct 26, 2010, at 6:43 AM, Rupa Schomaker wrote:
>>>>>>
>>>>>> What do you have set for loglevel in switch.conf.xml?  My guess is
>>>>>> that you have it set to something other than debug.  If that is set to (say)
>>>>>> info, then info is as low as you can go.  Setting the console log level to
>>>>>> debug will not give you debug output.
>>>>>>
>>>>>> sofia is unaware of upnp so a keepalive packet should have no impact
>>>>>> on it's operation.  I'm not familiar enough with sofia to tell what triggers
>>>>>> that event.
>>>>>>
>>>>>> On Tue, Oct 26, 2010 at 6:24 AM, Mark Campbell-Smith <
>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>
>>>>>>> OK..  tcpdump gave me this and I still do not see anything on FS
>>>>>>> except for this line:
>>>>>>>
>>>>>>> 2010-10-26 22:02:51.235608 [DEBUG] sofia.c:957 nua_i_outbound:
>>>>>>> unknown event 8: 102 NAT binding changed
>>>>>>>
>>>>>>> Hypertext Transfer Protocol
>>>>>>>     NOTIFY * HTTP/1.1\r\n
>>>>>>>     HOST:239.255.255.250:1900\r\n
>>>>>>>     Cache-Control:max-age=120\r\n
>>>>>>>     Location:http://192.168.1.1:1278/rootDesc.xml\r\n<http://192.168.1.1:1278/rootDesc.xml%5Cr%5Cn>
>>>>>>>     Server: Tomato UPnP/1.0 MiniUPnPd/1.4\r\n
>>>>>>>     NT:urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>>>>>>>
>>>>>>>  USN:uuid:882dbe81-c28a-4992-baf7-410c05f1caf4::urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>>>>>>>     NTS:ssdp:alive\r\n
>>>>>>>     \r\n
>>>>>>>
>>>>>>> This looks the same as you posted before, so why do I get the unknown
>>>>>>> event 8 (is this related)?
>>>>>>>
>>>>>>> Thanks!
>>>>>>>
>>>>>>> On Tue, Oct 26, 2010 at 3:32 PM, Mark Campbell-Smith <
>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>
>>>>>>>> :)  that was the first thing I did, but I get nothing; no keep alive
>>>>>>>> packets.
>>>>>>>>
>>>>>>>> in the logfile at startup I see:
>>>>>>>>
>>>>>>>> 2010-10-26 10:19:45.484258 [INFO] switch_nat.c:410 Scanning for NAT
>>>>>>>> 2010-10-26 10:19:45.485991 [DEBUG] switch_nat.c:167 Checking for PMP
>>>>>>>> 1/5
>>>>>>>> 2010-10-26 10:19:45.488575 [INFO] switch_nat.c:423 NAT detected
>>>>>>>> type: pmp, ExtIP: '203.xxx.xxx.xxx'
>>>>>>>> 2010-10-26 10:19:45.490683 [DEBUG] switch_nat.c:256 NAT thread
>>>>>>>> configured
>>>>>>>> 2010-10-26 10:19:45.492180 [DEBUG] switch_nat.c:267 NAT thread
>>>>>>>> started
>>>>>>>>
>>>>>>>> And nat_map status shows me which ports have been forwarded and the
>>>>>>>> external ip address (which is sometimes wrong).  I also see this in the
>>>>>>>> router gui.
>>>>>>>>
>>>>>>>> But I never see the keep alive packets.
>>>>>>>>
>>>>>>>> On Tue, Oct 26, 2010 at 3:13 PM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>
>>>>>>>>> Just turn up console logging to level 7.
>>>>>>>>>
>>>>>>>>> /log 7 if using fs_cli
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Mon, Oct 25, 2010 at 6:53 PM, Mark Campbell-Smith <
>>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> Hi!
>>>>>>>>>>
>>>>>>>>>> How do I enable debugging to get the UPnP keep alive packets?
>>>>>>>>>>
>>>>>>>>>> Thanks!
>>>>>>>>>>
>>>>>>>>>> On Tue, Oct 26, 2010 at 1:15 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>>>
>>>>>>>>>>> I need the debug logging from the upnp stuff.  It should look
>>>>>>>>>>> something like:
>>>>>>>>>>>
>>>>>>>>>>> +OK log level 7 [7]
>>>>>>>>>>> freeswitch at internal> 2010-10-25 09:13:59.926861 [DEBUG]
>>>>>>>>>>> switch_nat.c:299 got UPnP keep alive packet:
>>>>>>>>>>> NOTIFY * HTTP/1.1
>>>>>>>>>>> HOST:239.255.255.250:1900
>>>>>>>>>>> Cache-Control:max-age=60
>>>>>>>>>>> Location:http://192.168.1.1:5000/rootDesc.xml
>>>>>>>>>>> Server: Tomato UPnP/1.0 MiniUPnPd/1.4
>>>>>>>>>>> NT:urn:schemas-upnp-org:service:WANIPConnection:1
>>>>>>>>>>>
>>>>>>>>>>> USN:uuid:00000000-0000-0000-0000-000000000000::urn:schemas-upnp-org:service:WANIPConnection:1
>>>>>>>>>>> NTS:ssdp:alive
>>>>>>>>>>>
>>>>>>>>>>> I also run tomato and haven't had trouble with the upnp support
>>>>>>>>>>> so at least we have that part working.
>>>>>>>>>>>
>>>>>>>>>>> You should see the above keepalive appear periodically every 30s
>>>>>>>>>>> or so.  You should see another set of messages when you terminate the DSL
>>>>>>>>>>> connection and when that dsl connection comes back online.
>>>>>>>>>>>
>>>>>>>>>>> On Sat, Oct 23, 2010 at 5:36 PM, Mark Campbell-Smith <
>>>>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Hi!
>>>>>>>>>>>>
>>>>>>>>>>>> The router is a ASUS router but it is running the tomato
>>>>>>>>>>>> firmware which runs miniupnd.
>>>>>>>>>>>>
>>>>>>>>>>>> In the debug I see the message: 2010-10-24 08:02:59.178918
>>>>>>>>>>>> [DEBUG] sofia.c:956 nua_i_outbound: unknown event 8: 101 NAT detected
>>>>>>>>>>>>
>>>>>>>>>>>> This is what I see below. I hope I enabled all debug messages.
>>>>>>>>>>>>  I thought I saw some xml type messages earlier, but not sure how I enabled
>>>>>>>>>>>> them.
>>>>>>>>>>>>
>>>>>>>>>>>> nta_outgoing: RTT is 67.958 ms
>>>>>>>>>>>> outbound(0xb6e07c00): NAT binding changed: [210.xx.xxx.xx]:5080
>>>>>>>>>>>> != [58.xxx.xx.xx]:5080
>>>>>>>>>>>> nua(0xb6e07c00): event i_outbound 102 NAT binding changed
>>>>>>>>>>>> nua: nua_application_event: entering
>>>>>>>>>>>> 2010-10-24 09:05:08.849525 [DEBUG] sofia.c:956 nua_i_outbound:
>>>>>>>>>>>> unknown event 8: 102 NAT binding changed
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> On Sun, Oct 24, 2010 at 1:08 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> btw: what router are you using?
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Sat, Oct 23, 2010 at 9:06 AM, Rupa Schomaker <rupa at rupa.com
>>>>>>>>>>>>> > wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>> In a upnp config, FS is depending on the router to notify it
>>>>>>>>>>>>>> of the ip address change.  It is possible the upnp message(s) aren't
>>>>>>>>>>>>>> formatted as expected.  If you turn on debug logging you'll get the upnp
>>>>>>>>>>>>>> messages on the console.  If you can pastebin the capture of those while
>>>>>>>>>>>>>> dropping and restarting the dsl connection I can maybe see if there is
>>>>>>>>>>>>>> something obvious going on.  The debug should also have messages related to
>>>>>>>>>>>>>> the processing of those upnp messages.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Sat, Oct 23, 2010 at 7:04 AM, Mark Campbell-Smith <
>>>>>>>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> HI!
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> I know this question must have been answered 100's of
>>>>>>>>>>>>>>> times....
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> My adsl is a bit dodgy at the moment and tends to go up and
>>>>>>>>>>>>>>> down; which means I get assigned a new ip address from my ISP.  FS is nat'd
>>>>>>>>>>>>>>> behind a upnp capable router.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> The problem is that FS does not seem to be detecting the
>>>>>>>>>>>>>>> change.   For both the internal and external profile, I have auto-nat setup,
>>>>>>>>>>>>>>> and I have tried stun and host settings in vars.conf.  When I issue
>>>>>>>>>>>>>>> a nat_map status I see my old IP address, and if I then issue a nat_map
>>>>>>>>>>>>>>> reinit, I see the new public IP address.  How can I get this to be
>>>>>>>>>>>>>>> automatic?
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Also the public IP address shown in the nat_map status is not
>>>>>>>>>>>>>>> the same as shown in the sofia profile internal/external printouts.  Why is
>>>>>>>>>>>>>>> this?
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> I'm sure this is configuration, just not sure what to change.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Thanks!
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> --
>>>>>>>>>>>>>> -Rupa
>>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> --
>>>>>>>>>>>>> -Rupa
>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> --
>>>>>>>>>>> -Rupa
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>> -Rupa
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> -Rupa
>>>>>>  _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From anthony.minessale at gmail.com  Wed Oct 27 09:10:28 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 27 Oct 2010 11:10:28 -0500
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <3447E946-0BF8-4F7E-B5E9-E28F53DD0968@mgtech.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
	<407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>
	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
	<AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
	<6D9FD5E2-21FA-4630-8CF4-A87ADB30AC87@mgtech.com>
	<AANLkTi=BiDiXLgEoj_MfwmZbGjvXoOuNJgUYdetciNCj@mail.gmail.com>
	<9BAFDAF8-1E66-444C-A024-A88FF65D8BCA@mgtech.com>
	<3447E946-0BF8-4F7E-B5E9-E28F53DD0968@mgtech.com>
Message-ID: <AANLkTi=LRbz6ADhLvBKkjGxOywAVLf--GTkzXEpUoDbi@mail.gmail.com>

you are completely guessing at things.
I want you to understand that the only reason you are having problems
with this is because you don't understand how it works enough to know
what you are doing 100%

Its a given that the pnp stuff is only for your dynamic IP.
aggressive-nat-detection and sip-force-expires are all related to
inbound calls when the things who are registering to you may be behind
nat.

You need to learn the difference between which nat tools are
*) designed for your FS to run behind nat
*) designed for FS to run public and accept connections from devices behind nat.

If you have a static IP, you don't need the pnp stuff so -nonat is fine
What you need to do is set

1) set the params ext-sip-ip and ext-rtp-ip to your external static IP
2) map the sip ports and all of the rtp ports from your static IP to FS lan addr
3) set sip-ip and rtp-ip to the lan addr you forwarded through.

If you don't do this: your outbound registration will use NAT to your
provider and if there is no activity for the expire time on your NAT
mapping the reverse port mapping from your provider back to you is
lost.  This is why you set your register expires to a very low number,
(you need to make sure the provider does not turn the expires back up
in the reply because it will beat your choice *see sip trace) if this
is the case then you need the "ping" option set to 30, to continuously
send an options to your provider.

The static mapping is obviously the better, easier and more reliable solution.

So I want you to understand that the only way to keep a nat mapped
port alive is to continuously send traffic, all the other methods that
you are mentioning are to detect that phones registered to your are
behind nat, I gave you that force-expires option before because your
trace was full of inbound reg so I thought that is what you wanted
help with.








On Wed, Oct 27, 2010 at 10:43 AM, Mario G <mario_fs at mgtech.com> wrote:
> I should mention that I did not have this problem with an SPA9000 PBX
> (asterisk based) for over two years so FS may be pickier about upnp and/or
> nat, or just better at it exposing a problem in the router.
> I made different changes to the gateways to test different things. One
> failed after 17 hours, the other two stayed up. ?What?did not work:
> added?<variable name="sip-force-expires" value="30"/> to the directory
> entries as suggested.
> set the gateway expire times to 30 seconds.
> What worked (could be coincidental) for the two gateways that stayed up:
> I?Added?<param name="aggressive-nat-detection" value="true"/>
> I originally setup FS to use the static ip by setting external sip/rtp to
> just the static ip (no autonat:) and ran with -nonat but I could not get
> incoming calls. The only way it worked was to use autonat:1.2.3.4.?The
> router has 1 static public address and 1 dynamic external IP, this is the
> root of the problem,?upnp only tells FS about the dynamic ip??Will keep this
> thread up-to-date for anyone who may be in the same boat someday. Thanks
> again for looking at the trace.
> Mario
>
> You should be setting the req freq to a low number on the outbound gateways
>
> The examples you showed had a series of inbound reg
>
> also set expire-seconds to 30 in your gateway xml
>
>
> The problem is if you are not constantly sending traffic to the box
>
> the nat mapping will go away.
>
> If you are in production you should be using a static ip with a static
>
> mapping, any trouble you are having is your own fault for playing with
>
> fire. ?The best we can do is tell you how to keep it contained.
>
>
>
>
> On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com> wrote:
>
> I made the change. I had no idea the settings for the inside phones effected
> nat for the outside sip accounts. I was looking into aggressive-nat-
> detection since the internal profile status always shows the right external
> static IP but the nat_ap status always shows the dynamic ip. Crossing
> fingers/etc since this problem is 85% of time (weeks!) into FS changeover.
> Thanks!
>
> Mario
>
> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
>
> add
>
> <variable name="sip-force-expires" value="30"/>
>
> to the <variables> section of your <user>
>
> you have it at 600 and the nat mapping is timing out while the 600
>
> seconds is ticking away
>
>
>
> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
>
> From the TSP:
>
> "I have enabled the SIP trace on your account. We are not currently seeing
>
> any registration attempts to your account within the last 15 minutes. Please
>
> restart FreeSwitch so that registration attempts begin again. Thank you. ".
>
> So FS is not getting past router.
>
> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>
> I ran the global trace during the problem and it is
>
> at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
>
> "accttwo", "acct3". The trace includes phones since it was global. I am
>
> using:
>
> ???<param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>
> ???<param name="ext-sip-ip" value="autonat:my-static.ip"/>
>
> I tried dumping nat and removing the autonat: above and using -nonat but
>
> that did not work, registration proceeded but no calls inbound.
>
> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>
> Whoops, I am using an IP address for at least one gateway so that is not the
>
> problem:
>
> They look outbound to me and I am using dns for 2 and an IP for one so that
>
> is not the issue. I was able to get FS to clear this up by doing "nat_map
>
> reinit" which is why I think this is a nat problem. I will do the trace you
>
> mentioned. I will plug an ip address into one of the gateways to see what
>
> happens, they all fail at once. Thanks for responding!
>
> Mario
>
> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>
> I really need help on this as I have weeks into this problem. I thought I
>
> had it nailed but I guess not. After 5.5 hours I get:
>
> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>
> Registration, setting retry to 15 seconds.
>
> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>
> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>
> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
>
> Registration, setting retry to 30 seconds.
>
> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>
> Registration, setting retry to 15 seconds.
>
> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>
> Registration, setting retry to 15 seconds.
>
> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>
> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>
> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>
> and no way to make/get calls until I restart FS. I did this:
>
> 1. log 7
>
> 2. sofia profile xxxx siptrace on ??for each profile/gateway
>
> 3. restarted router
>
> All three did not solve the problem. The trace and log produced no
>
> additional lines which is why I am wondering if FS has a problem since the
>
> trace shows no SIP activity.
>
> 3 gateways with 2 ITSPs
>
> 2 DSL/WAN lines, 1 static and 1 dynamic
>
> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
>
> external static ip.
>
> sofia status profile ... has the right ext ip
>
> nat_map status shows the dynamic (wrong) IP
>
> I tried starting with -nonat but that was worse
>
> the only way to fix is restart FS.
>
> I read the wiki on external nat, auto_nat and everything else many times.
>
> Thanks Mario
>
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> 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
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> 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
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From shamun.toha at gmail.com  Wed Oct 27 09:21:43 2010
From: shamun.toha at gmail.com (Shamun toha md)
Date: Wed, 27 Oct 2010 18:21:43 +0200
Subject: [Freeswitch-users] FreeSWITCH - how to enable channels like
 mod_skypopen for Yahoo, Google Talk?
In-Reply-To: <AANLkTimsrtfmF2-hbu1MnkDcdxvm8x=FmrVkoRuY-YMn@mail.gmail.com>
References: <AANLkTikFHXgu-nZ5NBDQ4xON4c2D1WRHX33wOSRjPdSB@mail.gmail.com>
	<AANLkTimMfFohwpBhkyK+eVyF0qu86jUyVxUgOPcx-MgL@mail.gmail.com>
	<1288190227145-5678811.post@n2.nabble.com>
	<AANLkTimsrtfmF2-hbu1MnkDcdxvm8x=FmrVkoRuY-YMn@mail.gmail.com>
Message-ID: <AANLkTimT+j7itsgpYTbbHiLiyqjEhnYtttCfQgdX6FAC@mail.gmail.com>

Thanks, Ok but can we not have a "INTRANET" not connecting to Yahoo or
Google or Skype. Packets switching... in intranet-work. Where anything it
dumps in our FreeSwitch debug logs? To allow see more deep inside.

Example:
============
[ Ether net hub ]
============
|
+ FreeSwitch Box 1 (hostname: www.yahoo.com or www.skype.com ) <-->
FreeSwitch Box 2 (hostname: experts.freeswitch.org)
++
|___ Box 1 (listen to the packets as decoder from Box 2)  <---> Box 2 (sends
regular media or anything to Box 1)
|
|___ To find the all the mystery of wideband and audio perfection algorithm?


Thanks & Regards



On Wed, Oct 27, 2010 at 5:13 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Since it's SIP it'd be better using the mod_sofia stack to avoid
> duplicating the code that already exists. The only issue is connecting
> to Yahoo Messenger and authenticating to get a valid cookie to use.
> That would need a lot of customisations to be made to mod_sofia.
>
> The only library I can see out there to connect to Yahoo to generate
> the cookies is libyahoo2. That's not an option though as it's GPL,
> which isn't compatible with FreeSWITCH's MPL licence.
>
> -Steve
>
>
> On 27 October 2010 15:37, mazilo <Nabble at slickdeals.endjunk.com> wrote:
> >
> >
> > Steven Ayre wrote:
> >> Yahoo appears to accept SIP.  But it also looks like they use their
> >> own extensions on top of the SIP protocol for authentication (Y-Cookie
> >> header) so I don't think it'll work currently.
> >> http://wiki.freeswitch.org/wiki/Provider_Configuration:_Yahoo_Messenger
> >>
> http://blog.motiwala.com/2007/08/18/yahoo-messenger-voip-service-with-sip-phone
> >>
> >> -Steve
> > Perhaps, it is not a bad idea for FS developers to start writing a
> mod_yahoo
> > to support a Yahoo SIP call.
> >
> > -----
> > don't and stop are the ONLY two 4-letter words considered offensive to
> men,
> > but not when used together.
> > --
> > View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-how-to-enable-channels-like-mod-skypopen-for-Yahoo-Google-Talk-tp5677527p5678811.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Oct 27 09:24:10 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 27 Oct 2010 09:24:10 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <AANLkTinDmSjVbUr3YL+N3s4t_wGuVeeH_eQrBuSaLp=H@mail.gmail.com>

Hey folks,

Our FreeSWITCH conference call is coming up in less than an hour. Here's the
agenda:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_10_27

We don't have a specific presentation today but there are a few things I
need to discuss with you and we also have a few user tips and tricks that
I'd like to have everyone try and give their feedback.

Talk to you shortly.

-MC
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From mario_fs at mgtech.com  Wed Oct 27 09:50:03 2010
From: mario_fs at mgtech.com (Mario G)
Date: Wed, 27 Oct 2010 09:50:03 -0700
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <AANLkTi=LRbz6ADhLvBKkjGxOywAVLf--GTkzXEpUoDbi@mail.gmail.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
	<407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>
	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
	<AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
	<6D9FD5E2-21FA-4630-8CF4-A87ADB30AC87@mgtech.com>
	<AANLkTi=BiDiXLgEoj_MfwmZbGjvXoOuNJgUYdetciNCj@mail.gmail.com>
	<9BAFDAF8-1E66-444C-A024-A88FF65D8BCA@mgtech.com>
	<3447E946-0BF8-4F7E-B5E9-E28F53DD0968@mgtech.com>
	<AANLkTi=LRbz6ADhLvBKkjGxOywAVLf--GTkzXEpUoDbi@mail.gmail.com>
Message-ID: <BE754258-BE27-454F-B740-527A35DBDFFE@mgtech.com>

Thanks so much! I am sure many others will find this info invaluable. I will try the static route again but have one question: When I started with FS I found a "sip scanner"  in FS and someone on this group said not to use port mapping since it was a security risk. Is that true?

On Oct 27, 2010, at 9:10 AM, Anthony Minessale wrote:

> you are completely guessing at things.
> I want you to understand that the only reason you are having problems
> with this is because you don't understand how it works enough to know
> what you are doing 100%
> 
> Its a given that the pnp stuff is only for your dynamic IP.
> aggressive-nat-detection and sip-force-expires are all related to
> inbound calls when the things who are registering to you may be behind
> nat.
> 
> You need to learn the difference between which nat tools are
> *) designed for your FS to run behind nat
> *) designed for FS to run public and accept connections from devices behind nat.
> 
> If you have a static IP, you don't need the pnp stuff so -nonat is fine
> What you need to do is set
> 
> 1) set the params ext-sip-ip and ext-rtp-ip to your external static IP
> 2) map the sip ports and all of the rtp ports from your static IP to FS lan addr
> 3) set sip-ip and rtp-ip to the lan addr you forwarded through.
> 
> If you don't do this: your outbound registration will use NAT to your
> provider and if there is no activity for the expire time on your NAT
> mapping the reverse port mapping from your provider back to you is
> lost.  This is why you set your register expires to a very low number,
> (you need to make sure the provider does not turn the expires back up
> in the reply because it will beat your choice *see sip trace) if this
> is the case then you need the "ping" option set to 30, to continuously
> send an options to your provider.
> 
> The static mapping is obviously the better, easier and more reliable solution.
> 
> So I want you to understand that the only way to keep a nat mapped
> port alive is to continuously send traffic, all the other methods that
> you are mentioning are to detect that phones registered to your are
> behind nat, I gave you that force-expires option before because your
> trace was full of inbound reg so I thought that is what you wanted
> help with.
> 
> 
> 
> 
> 
> 
> 
> 
> On Wed, Oct 27, 2010 at 10:43 AM, Mario G <mario_fs at mgtech.com> wrote:
>> I should mention that I did not have this problem with an SPA9000 PBX
>> (asterisk based) for over two years so FS may be pickier about upnp and/or
>> nat, or just better at it exposing a problem in the router.
>> I made different changes to the gateways to test different things. One
>> failed after 17 hours, the other two stayed up.  What did not work:
>> added <variable name="sip-force-expires" value="30"/> to the directory
>> entries as suggested.
>> set the gateway expire times to 30 seconds.
>> What worked (could be coincidental) for the two gateways that stayed up:
>> I Added <param name="aggressive-nat-detection" value="true"/>
>> I originally setup FS to use the static ip by setting external sip/rtp to
>> just the static ip (no autonat:) and ran with -nonat but I could not get
>> incoming calls. The only way it worked was to use autonat:1.2.3.4. The
>> router has 1 static public address and 1 dynamic external IP, this is the
>> root of the problem, upnp only tells FS about the dynamic ip  Will keep this
>> thread up-to-date for anyone who may be in the same boat someday. Thanks
>> again for looking at the trace.
>> Mario
>> 
>> You should be setting the req freq to a low number on the outbound gateways
>> 
>> The examples you showed had a series of inbound reg
>> 
>> also set expire-seconds to 30 in your gateway xml
>> 
>> 
>> The problem is if you are not constantly sending traffic to the box
>> 
>> the nat mapping will go away.
>> 
>> If you are in production you should be using a static ip with a static
>> 
>> mapping, any trouble you are having is your own fault for playing with
>> 
>> fire.  The best we can do is tell you how to keep it contained.
>> 
>> 
>> 
>> 
>> On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com> wrote:
>> 
>> I made the change. I had no idea the settings for the inside phones effected
>> nat for the outside sip accounts. I was looking into aggressive-nat-
>> detection since the internal profile status always shows the right external
>> static IP but the nat_ap status always shows the dynamic ip. Crossing
>> fingers/etc since this problem is 85% of time (weeks!) into FS changeover.
>> Thanks!
>> 
>> Mario
>> 
>> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
>> 
>> add
>> 
>> <variable name="sip-force-expires" value="30"/>
>> 
>> to the <variables> section of your <user>
>> 
>> you have it at 600 and the nat mapping is timing out while the 600
>> 
>> seconds is ticking away
>> 
>> 
>> 
>> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
>> 
>> From the TSP:
>> 
>> "I have enabled the SIP trace on your account. We are not currently seeing
>> 
>> any registration attempts to your account within the last 15 minutes. Please
>> 
>> restart FreeSwitch so that registration attempts begin again. Thank you. ".
>> 
>> So FS is not getting past router.
>> 
>> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>> 
>> I ran the global trace during the problem and it is
>> 
>> at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
>> 
>> "accttwo", "acct3". The trace includes phones since it was global. I am
>> 
>> using:
>> 
>>    <param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>> 
>>    <param name="ext-sip-ip" value="autonat:my-static.ip"/>
>> 
>> I tried dumping nat and removing the autonat: above and using -nonat but
>> 
>> that did not work, registration proceeded but no calls inbound.
>> 
>> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>> 
>> Whoops, I am using an IP address for at least one gateway so that is not the
>> 
>> problem:
>> 
>> They look outbound to me and I am using dns for 2 and an IP for one so that
>> 
>> is not the issue. I was able to get FS to clear this up by doing "nat_map
>> 
>> reinit" which is why I think this is a nat problem. I will do the trace you
>> 
>> mentioned. I will plug an ip address into one of the gateways to see what
>> 
>> happens, they all fail at once. Thanks for responding!
>> 
>> Mario
>> 
>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>> 
>> I really need help on this as I have weeks into this problem. I thought I
>> 
>> had it nailed but I guess not. After 5.5 hours I get:
>> 
>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>> 
>> Registration, setting retry to 15 seconds.
>> 
>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>> 
>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>> 
>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
>> 
>> Registration, setting retry to 30 seconds.
>> 
>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>> 
>> Registration, setting retry to 15 seconds.
>> 
>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>> 
>> Registration, setting retry to 15 seconds.
>> 
>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>> 
>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>> 
>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>> 
>> and no way to make/get calls until I restart FS. I did this:
>> 
>> 1. log 7
>> 
>> 2. sofia profile xxxx siptrace on   for each profile/gateway
>> 
>> 3. restarted router
>> 
>> All three did not solve the problem. The trace and log produced no
>> 
>> additional lines which is why I am wondering if FS has a problem since the
>> 
>> trace shows no SIP activity.
>> 
>> 3 gateways with 2 ITSPs
>> 
>> 2 DSL/WAN lines, 1 static and 1 dynamic
>> 
>> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
>> 
>> external static ip.
>> 
>> sofia status profile ... has the right ext ip
>> 
>> nat_map status shows the dynamic (wrong) IP
>> 
>> I tried starting with -nonat but that was worse
>> 
>> the only way to fix is restart FS.
>> 
>> I read the wiki on external nat, auto_nat and everything else many times.
>> 
>> Thanks Mario
>> 
>> 
>> _______________________________________________
>> 
>> FreeSWITCH-users mailing list
>> 
>> FreeSWITCH-users at lists.freeswitch.org
>> 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> 
>> FreeSWITCH-users mailing list
>> 
>> FreeSWITCH-users at lists.freeswitch.org
>> 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 
>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> 
>> FreeSWITCH-users mailing list
>> 
>> FreeSWITCH-users at lists.freeswitch.org
>> 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> 
>> FreeSWITCH-users mailing list
>> 
>> FreeSWITCH-users at lists.freeswitch.org
>> 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 
>> 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
>> 
>> _______________________________________________
>> 
>> FreeSWITCH-users mailing list
>> 
>> FreeSWITCH-users at lists.freeswitch.org
>> 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> 
>> FreeSWITCH-users mailing list
>> 
>> FreeSWITCH-users at lists.freeswitch.org
>> 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 
>> 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
>> 
>> _______________________________________________
>> 
>> FreeSWITCH-users mailing list
>> 
>> FreeSWITCH-users at lists.freeswitch.org
>> 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Wed Oct 27 10:04:01 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 27 Oct 2010 12:04:01 -0500
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <BE754258-BE27-454F-B740-527A35DBDFFE@mgtech.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
	<407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>
	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
	<AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
	<6D9FD5E2-21FA-4630-8CF4-A87ADB30AC87@mgtech.com>
	<AANLkTi=BiDiXLgEoj_MfwmZbGjvXoOuNJgUYdetciNCj@mail.gmail.com>
	<9BAFDAF8-1E66-444C-A024-A88FF65D8BCA@mgtech.com>
	<3447E946-0BF8-4F7E-B5E9-E28F53DD0968@mgtech.com>
	<AANLkTi=LRbz6ADhLvBKkjGxOywAVLf--GTkzXEpUoDbi@mail.gmail.com>
	<BE754258-BE27-454F-B740-527A35DBDFFE@mgtech.com>
Message-ID: <AANLkTi=3Kd-bCHJSED7VEh-cvtpdXMs_EkWZL=s8J4Ux@mail.gmail.com>

if you map it or not, a scanner would penetrate it.
There are lot of sip scanners out there now, you just need to beware of them.


On Wed, Oct 27, 2010 at 11:50 AM, Mario G <mario_fs at mgtech.com> wrote:
> Thanks so much! I am sure many others will find this info invaluable. I will try the static route again but have one question: When I started with FS I found a "sip scanner" ?in FS and someone on this group said not to use port mapping since it was a security risk. Is that true?
>
> On Oct 27, 2010, at 9:10 AM, Anthony Minessale wrote:
>
>> you are completely guessing at things.
>> I want you to understand that the only reason you are having problems
>> with this is because you don't understand how it works enough to know
>> what you are doing 100%
>>
>> Its a given that the pnp stuff is only for your dynamic IP.
>> aggressive-nat-detection and sip-force-expires are all related to
>> inbound calls when the things who are registering to you may be behind
>> nat.
>>
>> You need to learn the difference between which nat tools are
>> *) designed for your FS to run behind nat
>> *) designed for FS to run public and accept connections from devices behind nat.
>>
>> If you have a static IP, you don't need the pnp stuff so -nonat is fine
>> What you need to do is set
>>
>> 1) set the params ext-sip-ip and ext-rtp-ip to your external static IP
>> 2) map the sip ports and all of the rtp ports from your static IP to FS lan addr
>> 3) set sip-ip and rtp-ip to the lan addr you forwarded through.
>>
>> If you don't do this: your outbound registration will use NAT to your
>> provider and if there is no activity for the expire time on your NAT
>> mapping the reverse port mapping from your provider back to you is
>> lost. ?This is why you set your register expires to a very low number,
>> (you need to make sure the provider does not turn the expires back up
>> in the reply because it will beat your choice *see sip trace) if this
>> is the case then you need the "ping" option set to 30, to continuously
>> send an options to your provider.
>>
>> The static mapping is obviously the better, easier and more reliable solution.
>>
>> So I want you to understand that the only way to keep a nat mapped
>> port alive is to continuously send traffic, all the other methods that
>> you are mentioning are to detect that phones registered to your are
>> behind nat, I gave you that force-expires option before because your
>> trace was full of inbound reg so I thought that is what you wanted
>> help with.
>>
>>
>>
>>
>>
>>
>>
>>
>> On Wed, Oct 27, 2010 at 10:43 AM, Mario G <mario_fs at mgtech.com> wrote:
>>> I should mention that I did not have this problem with an SPA9000 PBX
>>> (asterisk based) for over two years so FS may be pickier about upnp and/or
>>> nat, or just better at it exposing a problem in the router.
>>> I made different changes to the gateways to test different things. One
>>> failed after 17 hours, the other two stayed up. ?What did not work:
>>> added <variable name="sip-force-expires" value="30"/> to the directory
>>> entries as suggested.
>>> set the gateway expire times to 30 seconds.
>>> What worked (could be coincidental) for the two gateways that stayed up:
>>> I Added <param name="aggressive-nat-detection" value="true"/>
>>> I originally setup FS to use the static ip by setting external sip/rtp to
>>> just the static ip (no autonat:) and ran with -nonat but I could not get
>>> incoming calls. The only way it worked was to use autonat:1.2.3.4. The
>>> router has 1 static public address and 1 dynamic external IP, this is the
>>> root of the problem, upnp only tells FS about the dynamic ip ?Will keep this
>>> thread up-to-date for anyone who may be in the same boat someday. Thanks
>>> again for looking at the trace.
>>> Mario
>>>
>>> You should be setting the req freq to a low number on the outbound gateways
>>>
>>> The examples you showed had a series of inbound reg
>>>
>>> also set expire-seconds to 30 in your gateway xml
>>>
>>>
>>> The problem is if you are not constantly sending traffic to the box
>>>
>>> the nat mapping will go away.
>>>
>>> If you are in production you should be using a static ip with a static
>>>
>>> mapping, any trouble you are having is your own fault for playing with
>>>
>>> fire. ?The best we can do is tell you how to keep it contained.
>>>
>>>
>>>
>>>
>>> On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>
>>> I made the change. I had no idea the settings for the inside phones effected
>>> nat for the outside sip accounts. I was looking into aggressive-nat-
>>> detection since the internal profile status always shows the right external
>>> static IP but the nat_ap status always shows the dynamic ip. Crossing
>>> fingers/etc since this problem is 85% of time (weeks!) into FS changeover.
>>> Thanks!
>>>
>>> Mario
>>>
>>> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
>>>
>>> add
>>>
>>> <variable name="sip-force-expires" value="30"/>
>>>
>>> to the <variables> section of your <user>
>>>
>>> you have it at 600 and the nat mapping is timing out while the 600
>>>
>>> seconds is ticking away
>>>
>>>
>>>
>>> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>
>>> From the TSP:
>>>
>>> "I have enabled the SIP trace on your account. We are not currently seeing
>>>
>>> any registration attempts to your account within the last 15 minutes. Please
>>>
>>> restart FreeSwitch so that registration attempts begin again. Thank you. ".
>>>
>>> So FS is not getting past router.
>>>
>>> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>>>
>>> I ran the global trace during the problem and it is
>>>
>>> at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
>>>
>>> "accttwo", "acct3". The trace includes phones since it was global. I am
>>>
>>> using:
>>>
>>> ? ?<param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>>>
>>> ? ?<param name="ext-sip-ip" value="autonat:my-static.ip"/>
>>>
>>> I tried dumping nat and removing the autonat: above and using -nonat but
>>>
>>> that did not work, registration proceeded but no calls inbound.
>>>
>>> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>>>
>>> Whoops, I am using an IP address for at least one gateway so that is not the
>>>
>>> problem:
>>>
>>> They look outbound to me and I am using dns for 2 and an IP for one so that
>>>
>>> is not the issue. I was able to get FS to clear this up by doing "nat_map
>>>
>>> reinit" which is why I think this is a nat problem. I will do the trace you
>>>
>>> mentioned. I will plug an ip address into one of the gateways to see what
>>>
>>> happens, they all fail at once. Thanks for responding!
>>>
>>> Mario
>>>
>>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>>>
>>> I really need help on this as I have weeks into this problem. I thought I
>>>
>>> had it nailed but I guess not. After 5.5 hours I get:
>>>
>>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>>>
>>> Registration, setting retry to 15 seconds.
>>>
>>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>
>>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>
>>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
>>>
>>> Registration, setting retry to 30 seconds.
>>>
>>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>>>
>>> Registration, setting retry to 15 seconds.
>>>
>>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>>>
>>> Registration, setting retry to 15 seconds.
>>>
>>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>
>>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>
>>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>>>
>>> and no way to make/get calls until I restart FS. I did this:
>>>
>>> 1. log 7
>>>
>>> 2. sofia profile xxxx siptrace on ? for each profile/gateway
>>>
>>> 3. restarted router
>>>
>>> All three did not solve the problem. The trace and log produced no
>>>
>>> additional lines which is why I am wondering if FS has a problem since the
>>>
>>> trace shows no SIP activity.
>>>
>>> 3 gateways with 2 ITSPs
>>>
>>> 2 DSL/WAN lines, 1 static and 1 dynamic
>>>
>>> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
>>>
>>> external static ip.
>>>
>>> sofia status profile ... has the right ext ip
>>>
>>> nat_map status shows the dynamic (wrong) IP
>>>
>>> I tried starting with -nonat but that was worse
>>>
>>> the only way to fix is restart FS.
>>>
>>> I read the wiki on external nat, auto_nat and everything else many times.
>>>
>>> Thanks Mario
>>>
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> 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
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> 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
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From steveayre at gmail.com  Wed Oct 27 11:19:02 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 27 Oct 2010 19:19:02 +0100
Subject: [Freeswitch-users] FreeSWITCH - how to enable channels like
 mod_skypopen for Yahoo, Google Talk?
In-Reply-To: <AANLkTimT+j7itsgpYTbbHiLiyqjEhnYtttCfQgdX6FAC@mail.gmail.com>
References: <AANLkTikFHXgu-nZ5NBDQ4xON4c2D1WRHX33wOSRjPdSB@mail.gmail.com>
	<AANLkTimMfFohwpBhkyK+eVyF0qu86jUyVxUgOPcx-MgL@mail.gmail.com>
	<1288190227145-5678811.post@n2.nabble.com>
	<AANLkTimsrtfmF2-hbu1MnkDcdxvm8x=FmrVkoRuY-YMn@mail.gmail.com>
	<AANLkTimT+j7itsgpYTbbHiLiyqjEhnYtttCfQgdX6FAC@mail.gmail.com>
Message-ID: <AANLkTinJVKeTFRooVoV4jEsQvMtEfdWC0=rAXd_CnTFH@mail.gmail.com>

Yes you can configure FS on the google/yahoo hostnames, but DNS means
clients will get confused about where to connect to. You'd need to
serve the clients different IPs for those domains which is likely to
cause other problems. Google Talk might work that way though.

Yahoo won't because there's no YMSG service for Yahoo Messenger to
connect to, and I doubt it can make VoIP calls without signing into
the YMSG service.

-Steve



On 27 October 2010 17:21, Shamun toha md <shamun.toha at gmail.com> wrote:
> Thanks, Ok but can we not have a "INTRANET" not connecting to Yahoo or
> Google or Skype. Packets switching... in intranet-work. Where anything it
> dumps in our FreeSwitch debug logs? To allow see more deep inside.
>
> Example:
> ============
> [ Ether net hub ]
> ============
> |
> + FreeSwitch Box 1 (hostname: www.yahoo.com or www.skype.com ) <-->
> FreeSwitch Box 2 (hostname: experts.freeswitch.org)
> ++
> |___ Box 1 (listen to the packets as decoder from Box 2)? <---> Box 2 (sends
> regular media or anything to Box 1)
> |
> |___ To find the all the mystery of wideband and audio perfection algorithm?
>
>
> Thanks & Regards
>
>
>
> On Wed, Oct 27, 2010 at 5:13 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> Since it's SIP it'd be better using the mod_sofia stack to avoid
>> duplicating the code that already exists. The only issue is connecting
>> to Yahoo Messenger and authenticating to get a valid cookie to use.
>> That would need a lot of customisations to be made to mod_sofia.
>>
>> The only library I can see out there to connect to Yahoo to generate
>> the cookies is libyahoo2. That's not an option though as it's GPL,
>> which isn't compatible with FreeSWITCH's MPL licence.
>>
>> -Steve
>>
>>
>> On 27 October 2010 15:37, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>> >
>> >
>> > Steven Ayre wrote:
>> >> Yahoo appears to accept SIP. ?But it also looks like they use their
>> >> own extensions on top of the SIP protocol for authentication (Y-Cookie
>> >> header) so I don't think it'll work currently.
>> >> http://wiki.freeswitch.org/wiki/Provider_Configuration:_Yahoo_Messenger
>> >>
>> >> http://blog.motiwala.com/2007/08/18/yahoo-messenger-voip-service-with-sip-phone
>> >>
>> >> -Steve
>> > Perhaps, it is not a bad idea for FS developers to start writing a
>> > mod_yahoo
>> > to support a Yahoo SIP call.
>> >
>> > -----
>> > don't and stop are the ONLY two 4-letter words considered offensive to
>> > men,
>> > but not when used together.
>> > --
>> > View this message in context:
>> > http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-how-to-enable-channels-like-mod-skypopen-for-Yahoo-Google-Talk-tp5677527p5678811.html
>> > Sent from the freeswitch-users mailing list archive at Nabble.com.
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From david.villasmil.work at gmail.com  Wed Oct 27 15:17:25 2010
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 28 Oct 2010 00:17:25 +0200
Subject: [Freeswitch-users] xml_mod_cdr
In-Reply-To: <AANLkTi=NQ9HC98Q3QPDOkfvx45RiFWSmHuq7VWvnaKfZ@mail.gmail.com>
References: <AANLkTi=NQ9HC98Q3QPDOkfvx45RiFWSmHuq7VWvnaKfZ@mail.gmail.com>
Message-ID: <AANLkTinoJUd7OwtZkTh7wj3oWW4OU06-GpjYW-Qfgg7K@mail.gmail.com>

anyone?

On Tue, Oct 26, 2010 at 5:34 PM, David Villasmil
<david.villasmil.work at gmail.com> wrote:
> Hello Guys,
>
> I'm trying to configure mod_xml_cdr and i keep on getting this error on startup:
>
> 2010-10-26 17:06:06.364270 [CRIT] switch_loadable_module.c:882 Error
> Loading module /usr/local/freeswitch/mod/mod_xml_cdr.so
> **/usr/local/freeswitch/mod/mod_xml_cdr.so: undefined symbol:
> switch_channel_get_variable**
>
> my xml config is the following:
>
> <configuration name="xml_cdr.conf" description="XML CDR CURL logger">
> ?<settings>
> ? ?<param name="url" value="http://server_ip/cdrpost/post.php"/>
> ? ?<param name="retries" value="2"/>
> ? ?<param name="log-dir" value="/var/log/freeswitch"/>
> ? ?<param name="log-b-leg" value="true"/>
> ? ?<param name="prefix-a-leg" value="true"/>
> ? ?<param name="encode" value="true"/>
> ? ?<param name="err-log-dir" value="/var/log/freeswitch/errors"/>
> ?</settings>
> </configuration>
>
> My FreeSWITCH version is:
>
> 2010-10-26 17:25:38.227987 [CONSOLE] switch_core.c:1605
> FreeSWITCH Version 1.0.head (git-) Started.
>
>
> Can anyone help me out?
>
>
> Thanks!
>
>
> David
>


From johnsonw at eallway.com  Wed Oct 27 13:56:18 2010
From: johnsonw at eallway.com (Johnson)
Date: Wed, 27 Oct 2010 13:56:18 -0700 (PDT)
Subject: [Freeswitch-users] FS incoming calls Queue ACD implement
Message-ID: <1288212978229-5680281.post@n2.nabble.com>


Hi FS Group,
I am a new vanilla. Just want to realize Queue ACD function in Freeswitch.
Is any one know how to configure or create the script to process incoming
calls put into defined Queues without using mod_FIFO. Any additional
prompting is very appreciated. 

Johnson  
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-incoming-calls-Queue-ACD-implement-tp5680281p5680281.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From bwibowo at gmail.com  Wed Oct 27 15:44:06 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Thu, 28 Oct 2010 05:44:06 +0700
Subject: [Freeswitch-users] Problems making a receiveing calls with
	mod_jingling
In-Reply-To: <AANLkTi=K65YxFQAXudWiPTSAUSH2Xzjd4ks5_coSJ5K_@mail.gmail.com>
References: <201010131244.41086.justlikeef@gmail.com>
	<AANLkTim60-09hvL1=SBrb8R86_1qqPXE2-+42ZSxZ9Ja@mail.gmail.com>
	<201010131807.46035.justlikeef@gmail.com>
	<AANLkTi=K65YxFQAXudWiPTSAUSH2Xzjd4ks5_coSJ5K_@mail.gmail.com>
Message-ID: <AANLkTi=Exfm08ZMfm7Eur8GQBViX6KuPt_ntRYJ48qW_@mail.gmail.com>

hi
i use FreeSWITCH version: 1.0.head (git-cf5c1d2 2010-10-20 16-40-26 -0400)
i have followed the suggestion but still found this error



2010-10-28 06:41:45.720905 [ERR] libdingaling.c:1205 NODE ERROR!

2010-10-28 06:41:45.934903 [ERR] libdingaling.c:1205 NODE ERROR!

2010-10-28 06:41:46.158907 [ERR] libdingaling.c:1205 NODE ERROR!

2010-10-28 06:41:46.379903 [ERR] libdingaling.c:1205 NODE ERROR!

2010-10-28 06:41:46.589904 [ERR] libdingaling.c:1209 DISCONNECTED!

any help is highly appreciate


On Thu, Oct 14, 2010 at 12:28 PM, Jeffrey Leung <curriegrad2004 at gmail.com>wrote:

> This is my config inside the jingle_profiles:
>  <profile type="client">
> *    <param name="name" value="gtalk[Example]"/>*
>    <param name="login" value="[EXAMPLE]@gmail.com/talk"/>
>    <param name="password" value="[REDACTED]"/>
>
>    <param name="dialplan" value="XML"/>
>    <param name="context" value="public"/>
>    <param name="message" value="Online"/>
>
>    <param name="rtp-ip" value="$${bind_server_ip}"/>
>    <!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
>
>    <param name="auto-login" value="true"/>
>    <!-- SASL "plain" or "md5" -->
>
>    <param name="sasl" value="plain"/>
>    <!-- if the server where the jabber is hosted is not the same as the one
> in the jid -->
>
>    <param name="server" value="talk.google.com"/>
>    <!-- Enable TLS or not -->
>
>    <param name="tls" value="true"/>
>    <!-- disable to trade async for more calls -->
>
>    <param name="use-rtp-timer" value="true"/>
>    <!-- default extension (if one cannot be determined) -->
>    <param name="exten" value="[EXTENSION]"/>
>    <!-- VAD choose one -->
>    <!-- <param name="vad" value="in"/> -->
>    <!-- <param name="vad" value="out"/> -->
>
>    <param name="vad" value="both"/>
>    <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
>
>    <param name="candidate-acl" value="wan.auto"/>
>    <param name="local-network-acl" value="localnet.auto"/>
>  </profile>
>
> And dialplan to follow:
>
>  <extension name="gvoice_out">
>    <condition field="destination_number" expression="^4664(\d{10})$">
> *      <action application="bridge" data="dingaling/gtalk[EXAMPLE]/+1$
> 1 at voice.google.com"/>*
>    </condition>
>  </extension>
>
> Pay special attention to the bolded areas of the configuration example I've
> provided. Configuring Google talk to work with FreeSwitch is quite
> straightforward with my configuration example I've provided above. I've ran
> to this same configuration ambiguity when they first completed this feature
> anyways, so I hope my configuration example does help you into resolving
> that problem
>
>
> On Wed, Oct 13, 2010 at 3:07 PM, Rob Hutton <justlikeef at gmail.com> wrote:
> > Caught and changed that, but it does the same thing.  I also changed the
> profile name to gtalk in case there was a case sensitivity issue or
> something, but same result.
> >
> > I am told that the wiki is not quite correct either, but with everyone
> away from their development PCs, it will be the weekend before the
> corrections are available.
> >
> > --
> > Thanks,
> > Rob
> > On Wednesday 13 October 2010 17:51:07 Michael Collins wrote:
> >> > Here is the profile:
> >> >
> >> >  <profile name="dingaling_1">
> >> >
> >> I believe that this should be:
> >> <profile type="client">
> >>
> >> -MC
> >>
> > I
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Oct 27 16:02:39 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 27 Oct 2010 16:02:39 -0700
Subject: [Freeswitch-users] xml_mod_cdr
In-Reply-To: <AANLkTinoJUd7OwtZkTh7wj3oWW4OU06-GpjYW-Qfgg7K@mail.gmail.com>
References: <AANLkTi=NQ9HC98Q3QPDOkfvx45RiFWSmHuq7VWvnaKfZ@mail.gmail.com>
	<AANLkTinoJUd7OwtZkTh7wj3oWW4OU06-GpjYW-Qfgg7K@mail.gmail.com>
Message-ID: <AANLkTimwhL5FnNziu-RJWf3rk9LaZQbkvyHJj7v68fWT@mail.gmail.com>

Possible cruft in build system? I'd do make clean and try again.
-MC

On Wed, Oct 27, 2010 at 3:17 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> anyone?
>
> On Tue, Oct 26, 2010 at 5:34 PM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
> > Hello Guys,
> >
> > I'm trying to configure mod_xml_cdr and i keep on getting this error on
> startup:
> >
> > 2010-10-26 17:06:06.364270 [CRIT] switch_loadable_module.c:882 Error
> > Loading module /usr/local/freeswitch/mod/mod_xml_cdr.so
> > **/usr/local/freeswitch/mod/mod_xml_cdr.so: undefined symbol:
> > switch_channel_get_variable**
> >
> > my xml config is the following:
> >
> > <configuration name="xml_cdr.conf" description="XML CDR CURL logger">
> >  <settings>
> >    <param name="url" value="http://server_ip/cdrpost/post.php"/>
> >    <param name="retries" value="2"/>
> >    <param name="log-dir" value="/var/log/freeswitch"/>
> >    <param name="log-b-leg" value="true"/>
> >    <param name="prefix-a-leg" value="true"/>
> >    <param name="encode" value="true"/>
> >    <param name="err-log-dir" value="/var/log/freeswitch/errors"/>
> >  </settings>
> > </configuration>
> >
> > My FreeSWITCH version is:
> >
> > 2010-10-26 17:25:38.227987 [CONSOLE] switch_core.c:1605
> > FreeSWITCH Version 1.0.head (git-) Started.
> >
> >
> > Can anyone help me out?
> >
> >
> > Thanks!
> >
> >
> > David
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From moises.silva at gmail.com  Wed Oct 27 16:03:58 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 27 Oct 2010 19:03:58 -0400
Subject: [Freeswitch-users] issue with freetdm
In-Reply-To: <AANLkTimwOY+TcjacxuXXWJ=Da1k3MptOj=sOAhwvaLDR@mail.gmail.com>
References: <AANLkTimwOY+TcjacxuXXWJ=Da1k3MptOj=sOAhwvaLDR@mail.gmail.com>
Message-ID: <AANLkTin=isswKDbNWckxpYFo9SP1+VwGzCjd_KbQizWP@mail.gmail.com>

On Wed, Oct 27, 2010 at 9:23 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:
> Hello,
>
> I am using sangoma pri card A101 with freeSwitch.
> Today, I moved from openzap to freetdm.
> I have followed http://wiki.sangoma.com/wanpipe-freeswitch-ftdm-installation
> page to install freeSwitch and freedtm.
>
> After installing? freeSwitch and Sangoma, I am getting following error in
> fs_cli
>
> 2010-10-27 17:57:03.228457 [ERR] ftmod_sangoma_isdn_stack_rcv.c:883
> sng_isdn->s1:L1 Rx Error
> 2010-10-27 17:57:03.228457 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:871
> sng_isdn->s1: Resetting L1 link
>
>
> What is these mean?

Do you get that constantly or eventually the link comes up?

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From steveayre at gmail.com  Wed Oct 27 16:07:16 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 28 Oct 2010 00:07:16 +0100
Subject: [Freeswitch-users] xml_mod_cdr
In-Reply-To: <AANLkTinoJUd7OwtZkTh7wj3oWW4OU06-GpjYW-Qfgg7K@mail.gmail.com>
References: <AANLkTi=NQ9HC98Q3QPDOkfvx45RiFWSmHuq7VWvnaKfZ@mail.gmail.com>
	<AANLkTinoJUd7OwtZkTh7wj3oWW4OU06-GpjYW-Qfgg7K@mail.gmail.com>
Message-ID: <88EE93B6-C1D8-4DEA-9015-24AA374822AA@gmail.com>

Guessing that FS didn't compile properly. Try checking out a fresh copy of the git repository (clone not pull), bootstrap, configure and make.

Steve on iPhone

On 27 Oct 2010, at 23:17, David Villasmil <david.villasmil.work at gmail.com> wrote:

> anyone?
> 
> On Tue, Oct 26, 2010 at 5:34 PM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>> Hello Guys,
>> 
>> I'm trying to configure mod_xml_cdr and i keep on getting this error on startup:
>> 
>> 2010-10-26 17:06:06.364270 [CRIT] switch_loadable_module.c:882 Error
>> Loading module /usr/local/freeswitch/mod/mod_xml_cdr.so
>> **/usr/local/freeswitch/mod/mod_xml_cdr.so: undefined symbol:
>> switch_channel_get_variable**
>> 
>> my xml config is the following:
>> 
>> <configuration name="xml_cdr.conf" description="XML CDR CURL logger">
>>  <settings>
>>    <param name="url" value="http://server_ip/cdrpost/post.php"/>
>>    <param name="retries" value="2"/>
>>    <param name="log-dir" value="/var/log/freeswitch"/>
>>    <param name="log-b-leg" value="true"/>
>>    <param name="prefix-a-leg" value="true"/>
>>    <param name="encode" value="true"/>
>>    <param name="err-log-dir" value="/var/log/freeswitch/errors"/>
>>  </settings>
>> </configuration>
>> 
>> My FreeSWITCH version is:
>> 
>> 2010-10-26 17:25:38.227987 [CONSOLE] switch_core.c:1605
>> FreeSWITCH Version 1.0.head (git-) Started.
>> 
>> 
>> Can anyone help me out?
>> 
>> 
>> Thanks!
>> 
>> 
>> David
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Wed Oct 27 16:27:35 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 27 Oct 2010 16:27:35 -0700
Subject: [Freeswitch-users] FS incoming calls Queue ACD implement
In-Reply-To: <1288212978229-5680281.post@n2.nabble.com>
References: <1288212978229-5680281.post@n2.nabble.com>
Message-ID: <AANLkTinuCX6xrzXHEg1pA43_TXnauSr4EDpnQhmRdPe-@mail.gmail.com>

I'm afraid this question is a bit broad. I notice that you are trying to
avoid the use of mod_fifo - is there a reason for that? Alternatively you
can use Moc's newer ACD system: mod_callcenter. It is more of a traditional
ACD than mod_fifo. (Both get inbound calls sent to agents, they just differ
in the methodology used to accomplish that task.)

I've found the documentation on mod_callcenter to be refreshingly detailed
and complete:
http://wiki.freeswitch.org/wiki/Mod_callcenter

-MC

On Wed, Oct 27, 2010 at 1:56 PM, Johnson <johnsonw at eallway.com> wrote:

>
> Hi FS Group,
> I am a new vanilla. Just want to realize Queue ACD function in Freeswitch.
> Is any one know how to configure or create the script to process incoming
> calls put into defined Queues without using mod_FIFO. Any additional
> prompting is very appreciated.
>
> Johnson
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/FS-incoming-calls-Queue-ACD-implement-tp5680281p5680281.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Oct 27 16:32:32 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 27 Oct 2010 16:32:32 -0700
Subject: [Freeswitch-users] Problems making a receiveing calls with
	mod_jingling
In-Reply-To: <AANLkTi=Exfm08ZMfm7Eur8GQBViX6KuPt_ntRYJ48qW_@mail.gmail.com>
References: <201010131244.41086.justlikeef@gmail.com>
	<AANLkTim60-09hvL1=SBrb8R86_1qqPXE2-+42ZSxZ9Ja@mail.gmail.com>
	<201010131807.46035.justlikeef@gmail.com>
	<AANLkTi=K65YxFQAXudWiPTSAUSH2Xzjd4ks5_coSJ5K_@mail.gmail.com>
	<AANLkTi=Exfm08ZMfm7Eur8GQBViX6KuPt_ntRYJ48qW_@mail.gmail.com>
Message-ID: <AANLkTi=ZmQo6QHfEFoHPaNd_Tx2HrM+aZZAgfh569scz@mail.gmail.com>

First, update to the latest git head. Second, turn on dingaling debugging
and capture the output:
dl_debug on

Hopefully there will be clues.
-MC

On Wed, Oct 27, 2010 at 3:44 PM, budi wibowo <bwibowo at gmail.com> wrote:

> hi
> i use FreeSWITCH version: 1.0.head (git-cf5c1d2 2010-10-20 16-40-26 -0400)
> i have followed the suggestion but still found this error
>
>
>
> 2010-10-28 06:41:45.720905 [ERR] libdingaling.c:1205 NODE ERROR!
>
> 2010-10-28 06:41:45.934903 [ERR] libdingaling.c:1205 NODE ERROR!
>
> 2010-10-28 06:41:46.158907 [ERR] libdingaling.c:1205 NODE ERROR!
>
> 2010-10-28 06:41:46.379903 [ERR] libdingaling.c:1205 NODE ERROR!
>
> 2010-10-28 06:41:46.589904 [ERR] libdingaling.c:1209 DISCONNECTED!
>
> any help is highly appreciate
>
>
> On Thu, Oct 14, 2010 at 12:28 PM, Jeffrey Leung <curriegrad2004 at gmail.com>wrote:
>
>> This is my config inside the jingle_profiles:
>>  <profile type="client">
>> *    <param name="name" value="gtalk[Example]"/>*
>>    <param name="login" value="[EXAMPLE]@gmail.com/talk"/>
>>    <param name="password" value="[REDACTED]"/>
>>
>>    <param name="dialplan" value="XML"/>
>>    <param name="context" value="public"/>
>>    <param name="message" value="Online"/>
>>
>>    <param name="rtp-ip" value="$${bind_server_ip}"/>
>>    <!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
>>
>>    <param name="auto-login" value="true"/>
>>     <!-- SASL "plain" or "md5" -->
>>
>>    <param name="sasl" value="plain"/>
>>    <!-- if the server where the jabber is hosted is not the same as the
>> one in the jid -->
>>
>>    <param name="server" value="talk.google.com"/>
>>    <!-- Enable TLS or not -->
>>
>>    <param name="tls" value="true"/>
>>     <!-- disable to trade async for more calls -->
>>
>>    <param name="use-rtp-timer" value="true"/>
>>    <!-- default extension (if one cannot be determined) -->
>>    <param name="exten" value="[EXTENSION]"/>
>>    <!-- VAD choose one -->
>>    <!-- <param name="vad" value="in"/> -->
>>    <!-- <param name="vad" value="out"/> -->
>>
>>    <param name="vad" value="both"/>
>>     <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
>>
>>    <param name="candidate-acl" value="wan.auto"/>
>>    <param name="local-network-acl" value="localnet.auto"/>
>>   </profile>
>>
>> And dialplan to follow:
>>
>>  <extension name="gvoice_out">
>>    <condition field="destination_number" expression="^4664(\d{10})$">
>> *      <action application="bridge" data="dingaling/gtalk[EXAMPLE]/+1$
>> 1 at voice.google.com"/>*
>>    </condition>
>>  </extension>
>>
>> Pay special attention to the bolded areas of the configuration example
>> I've provided. Configuring Google talk to work with FreeSwitch is quite
>> straightforward with my configuration example I've provided above. I've ran
>> to this same configuration ambiguity when they first completed this feature
>> anyways, so I hope my configuration example does help you into resolving
>> that problem
>>
>>
>> On Wed, Oct 13, 2010 at 3:07 PM, Rob Hutton <justlikeef at gmail.com> wrote:
>> > Caught and changed that, but it does the same thing.  I also changed the
>> profile name to gtalk in case there was a case sensitivity issue or
>> something, but same result.
>> >
>> > I am told that the wiki is not quite correct either, but with everyone
>> away from their development PCs, it will be the weekend before the
>> corrections are available.
>> >
>> > --
>> > Thanks,
>> > Rob
>> > On Wednesday 13 October 2010 17:51:07 Michael Collins wrote:
>> >> > Here is the profile:
>> >> >
>> >> >  <profile name="dingaling_1">
>> >> >
>> >> I believe that this should be:
>> >> <profile type="client">
>> >>
>> >> -MC
>> >>
>> > I
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From moises.silva at gmail.com  Wed Oct 27 16:32:46 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 27 Oct 2010 19:32:46 -0400
Subject: [Freeswitch-users] FreeTDM - Screening Indicator Field
In-Reply-To: <AANLkTi=qfumFZKopgP9hxEA0EGMWgG=6SXPjaa7E8cXe@mail.gmail.com>
References: <AANLkTin-5ncn2NvGJo1Qmd5EkjUqf=xAaSr1_c_Kd79J@mail.gmail.com>
	<AANLkTimNdWANeBvzTLpCUbmLxApo47VMu3HZNkotiDoi@mail.gmail.com>
	<AANLkTi=qfumFZKopgP9hxEA0EGMWgG=6SXPjaa7E8cXe@mail.gmail.com>
Message-ID: <AANLkTi=GvtzB_DPMsSPodwnu0u1CTm45E-EoE3=n7H+s@mail.gmail.com>

On Wed, Oct 27, 2010 at 2:55 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> Hi,
> yes, I'm using ftmod_sangoma_isdn library.
> Thanks,

To check the value on incoming call, try screen_bit variable
${screen_bit} which will be either true or false.

To set it for outgoing calls, try using the privacy application that
FreeSWITCH has.

<application name="privacy" data="on" />

Let me know if you need anything else,

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From ovvenkatesan at gmail.com  Wed Oct 27 21:04:21 2010
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Thu, 28 Oct 2010 09:34:21 +0530
Subject: [Freeswitch-users] issue with freetdm
In-Reply-To: <AANLkTin=isswKDbNWckxpYFo9SP1+VwGzCjd_KbQizWP@mail.gmail.com>
References: <AANLkTimwOY+TcjacxuXXWJ=Da1k3MptOj=sOAhwvaLDR@mail.gmail.com>
	<AANLkTin=isswKDbNWckxpYFo9SP1+VwGzCjd_KbQizWP@mail.gmail.com>
Message-ID: <AANLkTinYb5P7oSm1z5u=WJCqHXDjP7sbcojgr3T95nqc@mail.gmail.com>

Hi moises,

I am getting this error every 20 seconds.

freeswitch at internal> 2010-10-28 09:32:55.366623 [ERR]
ftmod_sangoma_isdn_stack_rcv.c:883 sng_isdn->s1:L1 Rx Error
2010-10-28 09:32:55.366623 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:871
sng_isdn->s1: Resetting L1 link
2010-10-28 09:33:05.364851 [ERR] ftmod_sangoma_isdn_stack_rcv.c:883
sng_isdn->s1:L1 Rx Error
2010-10-28 09:33:05.364851 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:871
sng_isdn->s1: Resetting L1 link
2010-10-28 09:33:25.361287 [ERR] ftmod_sangoma_isdn_stack_rcv.c:883
sng_isdn->s1:L1 Rx Error
2010-10-28 09:33:25.361287 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:871
sng_isdn->s1: Resetting L1 link
2010-10-28 09:33:45.358910 [ERR] ftmod_sangoma_isdn_stack_rcv.c:883
sng_isdn->s1:L1 Rx Error
2010-10-28 09:33:45.358910 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:871
sng_isdn->s1: Resetting L1 link
2010-10-28 09:33:55.353722 [ERR] ftmod_sangoma_isdn_stack_rcv.c:883
sng_isdn->s1:L1 Rx Error
2010-10-28 09:33:55.353722 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:871
sng_isdn->s1: Resetting L1 link



On Thu, Oct 28, 2010 at 4:33 AM, Moises Silva <moises.silva at gmail.com>wrote:

> On Wed, Oct 27, 2010 at 9:23 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:
> > Hello,
> >
> > I am using sangoma pri card A101 with freeSwitch.
> > Today, I moved from openzap to freetdm.
> > I have followed
> http://wiki.sangoma.com/wanpipe-freeswitch-ftdm-installation
> > page to install freeSwitch and freedtm.
> >
> > After installing  freeSwitch and Sangoma, I am getting following error in
> > fs_cli
> >
> > 2010-10-27 17:57:03.228457 [ERR] ftmod_sangoma_isdn_stack_rcv.c:883
> > sng_isdn->s1:L1 Rx Error
> > 2010-10-27 17:57:03.228457 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:871
> > sng_isdn->s1: Resetting L1 link
> >
> >
> > What is these mean?
>
> Do you get that constantly or eventually the link comes up?
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
> L3R 9R6 Canada
> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From ovvenkatesan at gmail.com  Wed Oct 27 23:17:01 2010
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Thu, 28 Oct 2010 11:47:01 +0530
Subject: [Freeswitch-users] problem with ODBC connection in new Git version
Message-ID: <AANLkTinqy1wJWB2i=ORyYvy0AVauc8WW084VPZE+Xetz@mail.gmail.com>

Hi,

I cant connect to the ODBC using Lua via *freeswitch.Dbh*

I am using below script to connect to the database
*local dbh = assert(freeswitch.Dbh("dsn_name","venkat","venkat123"))*

I am getting below error while executing the script
*2010-10-28 11:45:50.174582 [CRIT] switch_core_sqldb.c:356 Failure! ODBC NOT
AVAILABLE!*
previously it was working fine. recently,
I have formatted my server and installed latest git version.
 After that I am getting this error


I can to connect to the DB using below isql comman
*isql dsn_name venkat venkat123*

I am using

*FreeSWITCH Version 1.0.head (git-b639aeb 2010-10-27 00-57-13 -0400)*
*CENT OS 5.3*


-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From ovvenkatesan at gmail.com  Wed Oct 27 23:22:10 2010
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Thu, 28 Oct 2010 11:52:10 +0530
Subject: [Freeswitch-users] issue with freetdm
In-Reply-To: <AANLkTinYb5P7oSm1z5u=WJCqHXDjP7sbcojgr3T95nqc@mail.gmail.com>
References: <AANLkTimwOY+TcjacxuXXWJ=Da1k3MptOj=sOAhwvaLDR@mail.gmail.com>
	<AANLkTin=isswKDbNWckxpYFo9SP1+VwGzCjd_KbQizWP@mail.gmail.com>
	<AANLkTinYb5P7oSm1z5u=WJCqHXDjP7sbcojgr3T95nqc@mail.gmail.com>
Message-ID: <AANLkTikxER-ADbewv-k9gbNskhu=C+enCXzdgd=OgOg5@mail.gmail.com>

Hi Moises,
For you information,
I have upgraded my sangoma A101 card to v37 and
Installed latest version of wanpipe (wanpipe-3.5.17.8)

Still, I am facing the same issue,
thanks in advance,

Regards,
Venkat.

On Thu, Oct 28, 2010 at 9:34 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:

> Hi moises,
>
> I am getting this error every 20 seconds.
>
> freeswitch at internal> 2010-10-28 09:32:55.366623 [ERR]
> ftmod_sangoma_isdn_stack_rcv.c:883 sng_isdn->s1:L1 Rx Error
> 2010-10-28 09:32:55.366623 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:871
> sng_isdn->s1: Resetting L1 link
> 2010-10-28 09:33:05.364851 [ERR] ftmod_sangoma_isdn_stack_rcv.c:883
> sng_isdn->s1:L1 Rx Error
> 2010-10-28 09:33:05.364851 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:871
> sng_isdn->s1: Resetting L1 link
> 2010-10-28 09:33:25.361287 [ERR] ftmod_sangoma_isdn_stack_rcv.c:883
> sng_isdn->s1:L1 Rx Error
> 2010-10-28 09:33:25.361287 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:871
> sng_isdn->s1: Resetting L1 link
> 2010-10-28 09:33:45.358910 [ERR] ftmod_sangoma_isdn_stack_rcv.c:883
> sng_isdn->s1:L1 Rx Error
> 2010-10-28 09:33:45.358910 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:871
> sng_isdn->s1: Resetting L1 link
> 2010-10-28 09:33:55.353722 [ERR] ftmod_sangoma_isdn_stack_rcv.c:883
> sng_isdn->s1:L1 Rx Error
> 2010-10-28 09:33:55.353722 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:871
> sng_isdn->s1: Resetting L1 link
>
>
>
>
> On Thu, Oct 28, 2010 at 4:33 AM, Moises Silva <moises.silva at gmail.com>wrote:
>
>> On Wed, Oct 27, 2010 at 9:23 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:
>> > Hello,
>> >
>> > I am using sangoma pri card A101 with freeSwitch.
>> > Today, I moved from openzap to freetdm.
>> > I have followed
>> http://wiki.sangoma.com/wanpipe-freeswitch-ftdm-installation
>> > page to install freeSwitch and freedtm.
>> >
>> > After installing  freeSwitch and Sangoma, I am getting following error
>> in
>> > fs_cli
>> >
>> > 2010-10-27 17:57:03.228457 [ERR] ftmod_sangoma_isdn_stack_rcv.c:883
>> > sng_isdn->s1:L1 Rx Error
>> > 2010-10-27 17:57:03.228457 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:871
>> > sng_isdn->s1: Resetting L1 link
>> >
>> >
>> > What is these mean?
>>
>> Do you get that constantly or eventually the link comes up?
>>
>> Moises Silva
>> Senior Software Engineer
>> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
>> L3R 9R6 Canada
>> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> If you have come to help me, you are wasting your time.
> If you have come to because your liberation is bound up in mine, we can
> work together.
>
>
> Regards
> Venkatesan OV.
>
>


-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From ovvenkatesan at gmail.com  Wed Oct 27 23:36:29 2010
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Thu, 28 Oct 2010 12:06:29 +0530
Subject: [Freeswitch-users] problem with ODBC connection in new Git version
In-Reply-To: <AANLkTinqy1wJWB2i=ORyYvy0AVauc8WW084VPZE+Xetz@mail.gmail.com>
References: <AANLkTinqy1wJWB2i=ORyYvy0AVauc8WW084VPZE+Xetz@mail.gmail.com>
Message-ID: <AANLkTimweTitWzUhQtkWRuYj0bLbFKJ05BmdDHVkPayy@mail.gmail.com>

Hi,

for your information,

I have copied odbc configuration files to */etc/* directory

thanks in advance,
Venkat.

---------- Forwarded message ----------
From: ovvenkat <ovvenkatesan at gmail.com>
Date: Thu, Oct 28, 2010 at 11:47 AM
Subject: problem with ODBC connection in new Git version
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>


Hi,

I cant connect to the ODBC using Lua via *freeswitch.Dbh*

I am using below script to connect to the database
*local dbh = assert(freeswitch.Dbh("dsn_name","venkat","venkat123"))*

I am getting below error while executing the script
*2010-10-28 11:45:50.174582 [CRIT] switch_core_sqldb.c:356 Failure! ODBC NOT
AVAILABLE!*
previously it was working fine. recently,
I have formatted my server and installed latest git version.
 After that I am getting this error


I can to connect to the DB using below isql comman
*isql dsn_name venkat venkat123*

I am using

*FreeSWITCH Version 1.0.head (git-b639aeb 2010-10-27 00-57-13 -0400)*
*CENT OS 5.3*


-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.




-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From steveayre at gmail.com  Thu Oct 28 00:40:53 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 28 Oct 2010 08:40:53 +0100
Subject: [Freeswitch-users] problem with ODBC connection in new Git
	version
In-Reply-To: <AANLkTinqy1wJWB2i=ORyYvy0AVauc8WW084VPZE+Xetz@mail.gmail.com>
References: <AANLkTinqy1wJWB2i=ORyYvy0AVauc8WW084VPZE+Xetz@mail.gmail.com>
Message-ID: <AANLkTi=CGHn=ZbR-AqcCa2ttjhK4mVLZcfefGNEdkAi3@mail.gmail.com>

You have compiled FS without ODBC support. Did you configure with
--enable-core-odbc-support?

-Steve


On 28 October 2010 07:17, ovvenkat <ovvenkatesan at gmail.com> wrote:
> Hi,
>
> I cant connect to the ODBC using Lua via freeswitch.Dbh
>
> I am using below script to connect to the database
> local dbh = assert(freeswitch.Dbh("dsn_name","venkat","venkat123"))
>
> I am getting below error while executing the script
> 2010-10-28 11:45:50.174582 [CRIT] switch_core_sqldb.c:356 Failure! ODBC NOT
> AVAILABLE!
> previously it was working fine. recently,
> I have formatted my server and installed latest git version.
> ?After that I am getting this error
>
>
> I can to connect to the DB using below isql comman
> isql dsn_name venkat venkat123
>
> I am using
>
> FreeSWITCH Version 1.0.head (git-b639aeb 2010-10-27 00-57-13 -0400)
> CENT OS 5.3
>
>
> --
>
> If you have come to help me, you are wasting your time.
> If you have come to because your liberation is bound up in mine, we can work
> together.
>
>
> Regards
> Venkatesan OV.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From wstephen80 at gmail.com  Thu Oct 28 02:48:04 2010
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Thu, 28 Oct 2010 11:48:04 +0200
Subject: [Freeswitch-users] FreeTDM - Screening Indicator Field
In-Reply-To: <AANLkTi=GvtzB_DPMsSPodwnu0u1CTm45E-EoE3=n7H+s@mail.gmail.com>
References: <AANLkTin-5ncn2NvGJo1Qmd5EkjUqf=xAaSr1_c_Kd79J@mail.gmail.com>
	<AANLkTimNdWANeBvzTLpCUbmLxApo47VMu3HZNkotiDoi@mail.gmail.com>
	<AANLkTi=qfumFZKopgP9hxEA0EGMWgG=6SXPjaa7E8cXe@mail.gmail.com>
	<AANLkTi=GvtzB_DPMsSPodwnu0u1CTm45E-EoE3=n7H+s@mail.gmail.com>
Message-ID: <AANLkTimSLxnDuJESWX6CDa4W4RFJEYMXY_3gPK4SW+tn@mail.gmail.com>

The row:

<action application="privacy" data="on" />

changes the information element "Presentation Indicator" to the value
"Presentation Restricted".

My problem is different: I want to change the "Screening Indicator"
information element that is set by FreeTDM to the value "User-provided,
verified and passed" (value = 1) and I have found no way to change it. I
want to set it to "Network provided" (value = 3).

Stephen


On Thu, Oct 28, 2010 at 1:32 AM, Moises Silva <moises.silva at gmail.com>wrote:

> On Wed, Oct 27, 2010 at 2:55 AM, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> > Hi,
> > yes, I'm using ftmod_sangoma_isdn library.
> > Thanks,
>
> To check the value on incoming call, try screen_bit variable
> ${screen_bit} which will be either true or false.
>
> To set it for outgoing calls, try using the privacy application that
> FreeSWITCH has.
>
> <application name="privacy" data="on" />
>
> Let me know if you need anything else,
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
> L3R 9R6 Canada
> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ovvenkatesan at gmail.com  Thu Oct 28 03:06:13 2010
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Thu, 28 Oct 2010 15:36:13 +0530
Subject: [Freeswitch-users] problem with ODBC connection in new Git
	version
In-Reply-To: <AANLkTi=CGHn=ZbR-AqcCa2ttjhK4mVLZcfefGNEdkAi3@mail.gmail.com>
References: <AANLkTinqy1wJWB2i=ORyYvy0AVauc8WW084VPZE+Xetz@mail.gmail.com>
	<AANLkTi=CGHn=ZbR-AqcCa2ttjhK4mVLZcfefGNEdkAi3@mail.gmail.com>
Message-ID: <AANLkTi=XGG5UFkhRN8k5z4DoRMWvqKPsD8PaBN0FmQPz@mail.gmail.com>

Hi Steve,

It works. I thought that, it has enabled by default.
thanks again,

Regards,
Venkat.

On Thu, Oct 28, 2010 at 1:10 PM, Steven Ayre <steveayre at gmail.com> wrote:

> You have compiled FS without ODBC support. Did you configure with
> --enable-core-odbc-support?
>
> -Steve
>
>
> On 28 October 2010 07:17, ovvenkat <ovvenkatesan at gmail.com> wrote:
> > Hi,
> >
> > I cant connect to the ODBC using Lua via freeswitch.Dbh
> >
> > I am using below script to connect to the database
> > local dbh = assert(freeswitch.Dbh("dsn_name","venkat","venkat123"))
> >
> > I am getting below error while executing the script
> > 2010-10-28 11:45:50.174582 [CRIT] switch_core_sqldb.c:356 Failure! ODBC
> NOT
> > AVAILABLE!
> > previously it was working fine. recently,
> > I have formatted my server and installed latest git version.
> >  After that I am getting this error
> >
> >
> > I can to connect to the DB using below isql comman
> > isql dsn_name venkat venkat123
> >
> > I am using
> >
> > FreeSWITCH Version 1.0.head (git-b639aeb 2010-10-27 00-57-13 -0400)
> > CENT OS 5.3
> >
> >
> > --
> >
> > If you have come to help me, you are wasting your time.
> > If you have come to because your liberation is bound up in mine, we can
> work
> > together.
> >
> >
> > Regards
> > Venkatesan OV.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From freeswitch at peely.com  Thu Oct 28 06:01:29 2010
From: freeswitch at peely.com (peely)
Date: Thu, 28 Oct 2010 06:01:29 -0700 (PDT)
Subject: [Freeswitch-users] FreeSWITCH Nagios Plugin
In-Reply-To: <2d9149cd0912141149o4b72cc07pdcf5224f17ae260d@mail.gmail.com>
References: <2d9149cd0912141149o4b72cc07pdcf5224f17ae260d@mail.gmail.com>
Message-ID: <30076321.post@talk.nabble.com>




Kristian Kielhofner-3 wrote:
> 
> Hello everyone,
> 
>   I've been looking for a FreeSWITCH Nagios plugin.  Ideally I'd like
> to connect to the event socket and run some api commands and return
> them (as opposed to checking SIP, for example).  I haven't found
> anything and I've started to write one in perl using ESL.  I'm sure
> whatever I come up with is going to be pretty ugly and I'd much rather
> use something else.  Has this been done?
> 
> 
> 


This may be a bit basic for your needs, but it does what I need.

It outputs something like:

Occupancy: 17/2000, Uptime: 0y0d 0:38:7.78287s 

check_freeswitch.sh:

#!/bin/bash
host="$1"
pass="$2"

set fsStatus=" "
/usr/lib/nagios/plugins/fs_cli -H $host -p $pass -x "show calls count" >
/tmp/fsStatus
if [ $? -ne 0 ]; then
        echo "Critical: Freeswitch not responding!"
        rm /tmp/fsStatus
        exit -1
fi

fsCalls=`grep "total."</tmp/fsStatus`
/usr/lib/nagios/plugins/fs_cli -H $host -p $pass -x "status" > /tmp/fsStatus
fsUptime=`grep "UP"</tmp/fsStatus`
fsMaxChannels=`grep "max"</tmp/fsStatus`

#Strip unwanted chars from the channels responses to get pure ints.
fsCalls=${fsCalls/ total./}
fsMaxChannels=${fsMaxChannels/ session(s) max/}

# Reformat the uptime statement for efficiency
fsUptime=${fsUptime/UP /}
fsUptime=${fsUptime/ years, /y}
fsUptime=${fsUptime/ year, /y}
fsUptime=${fsUptime/ days, /d }
fsUptime=${fsUptime/ day, /d }
fsUptime=${fsUptime/ hours, /:}
fsUptime=${fsUptime/ hour, /:}
fsUptime=${fsUptime/ minutes, /:}
fsUptime=${fsUptime/ minute, /:}
fsUptime=${fsUptime/ seconds, /.}
fsUptime=${fsUptime/ second, /.}
fsUptime=${fsUptime/ milliseconds, /}
fsUptime=${fsUptime/ millisecond, /}
fsUptime=${fsUptime/ microseconds/s}
fsUptime=${fsUptime/ microsecond/s}

# Build a message up
message="Occupancy: $fsCalls/$fsMaxChannels, Uptime: $fsUptime"
echo $message

rm /tmp/fsStatus
#exit $fsCalls
exit 0



commands.cfg:

#check freeswitch status
define command{
        command_name    check_freeswitch
        command_line    $USER1$/check_freeswitch.sh $HOSTADDRESS$
EslPassword << change this.
}



Services configuration:

define service{
        hostgroup_name                  mediaservers
        use                             generic-service         ; Name of
service template to use
        service_description             Freeswitch
        check_command                   check_freeswitch
        check_interval                  1
        retry_interval                  1
        max_check_attempts              3
        }




Hope it helps, or at least gives you a starting point.


Neil.




-- 
View this message in context: http://old.nabble.com/FreeSWITCH-Nagios-Plugin-tp26783791p30076321.html
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From johnsonw at eallway.com  Wed Oct 27 19:58:19 2010
From: johnsonw at eallway.com (Johnson)
Date: Wed, 27 Oct 2010 19:58:19 -0700 (PDT)
Subject: [Freeswitch-users] FS incoming calls Queue ACD implement
In-Reply-To: <AANLkTinuCX6xrzXHEg1pA43_TXnauSr4EDpnQhmRdPe-@mail.gmail.com>
References: <1288212978229-5680281.post@n2.nabble.com>
	<AANLkTinuCX6xrzXHEg1pA43_TXnauSr4EDpnQhmRdPe-@mail.gmail.com>
Message-ID: <AANLkTinuy+DhD0mZGj0BWOMx2Ha8sTXLetnQWLNK9PX6@mail.gmail.com>


Hi MC,

Thanks for your feedback quickly. Actually i have tested mod_fifo and met
the callerid can't transfer to agent's phone before the agent pickup.
Traditional phones almost don't support new features like getting sip
"UPDATE MESSAGE". As alternatively, i try to use Moc' mod_callcenter, but
unfortunitely, there is no mod_callcenter package in FS1.06, even i put in
dialplan
<action application="callcenter" data="support at default"/>, it show log:
 [ERR] switch_core_session.c:1731 Invalid Application callcenter
Is it meaning that the mod_callcenter is still going on coding? Any further
more helps would be much appreciated.

Thanks,

Johnson


On Thu, Oct 28, 2010 at 7:37 AM, mercutioviz [via freeswitch-users] <
ml-node+5680777-556671345-298848 at n2.nabble.com<ml-node%2B5680777-556671345-298848 at n2.nabble.com>
> wrote:

> I'm afraid this question is a bit broad. I notice that you are trying to
> avoid the use of mod_fifo - is there a reason for that? Alternatively you
> can use Moc's newer ACD system: mod_callcenter. It is more of a traditional
> ACD than mod_fifo. (Both get inbound calls sent to agents, they just differ
> in the methodology used to accomplish that task.)
>
> I've found the documentation on mod_callcenter to be refreshingly detailed
> and complete:
> http://wiki.freeswitch.org/wiki/Mod_callcenter
>
> -MC
>
> On Wed, Oct 27, 2010 at 1:56 PM, Johnson <[hidden email]<http://user/SendEmail.jtp?type=node&node=5680777&i=0>
> > wrote:
>
>>
>> Hi FS Group,
>> I am a new vanilla. Just want to realize Queue ACD function in Freeswitch.
>> Is any one know how to configure or create the script to process incoming
>> calls put into defined Queues without using mod_FIFO. Any additional
>> prompting is very appreciated.
>>
>> Johnson
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/FS-incoming-calls-Queue-ACD-implement-tp5680281p5680281.html<http://freeswitch-users.2379917.n2.nabble.com/FS-incoming-calls-Queue-ACD-implement-tp5680281p5680281.html?by-user=t>
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> [hidden email] <http://user/SendEmail.jtp?type=node&node=5680777&i=1>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org <http://www.freeswitch.org/?by-user=t>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> [hidden email] <http://user/SendEmail.jtp?type=node&node=5680777&i=2>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org <http://www.freeswitch.org/?by-user=t&by-user=t>
>
>
> ------------------------------
> View message @
> http://freeswitch-users.2379917.n2.nabble.com/FS-incoming-calls-Queue-ACD-implement-tp5680281p5680777.html
> To unsubscribe from FS incoming calls Queue ACD implement, click here<http://freeswitch-users.2379917.n2.nabble.com/template/TplServlet.jtp?tpl=unsubscribe_by_code&node=5680281&code=am9obnNvbndAZWFsbHdheS5jb218NTY4MDI4MXwtNTg0MTQ0ODU4>.
>
>
>

-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-incoming-calls-Queue-ACD-implement-tp5680281p5681149.html
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From msarro at gmail.com  Wed Oct 27 09:13:34 2010
From: msarro at gmail.com (Matty Sarro)
Date: Wed, 27 Oct 2010 12:13:34 -0400
Subject: [Freeswitch-users] Moose Penis? Wiki Installation question
Message-ID: <AANLkTi=-iw8PdxZ7a7MrK0u_C0TkLysgHh-kqTrqus5j@mail.gmail.com>

So after checking the wiki on installation (
http://wiki.freeswitch.org/wiki/Installation_Guide), I attempted to go to
the latest build version. The link includes a joke about moose penis, and
then a comment to just use 1.0.6. If we're just supposed to use 1.0.6, why
does the installation guide keep mentioning to use the latest build or the
latest git source to install? It seems to contradict itself.
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From mcampbellsmith at gmail.com  Wed Oct 27 17:45:11 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 28 Oct 2010 11:45:11 +1100
Subject: [Freeswitch-users] FS not detecting public IP address change
In-Reply-To: <AANLkTinvSh0DJJAWxv_A+=WgcVubu1u5EKQMo-kaKVDp@mail.gmail.com>
References: <AANLkTimBEwo+vsMe8CAEp6o3mJNzSL65rrVOtxgOcBrf@mail.gmail.com>
	<AANLkTinEoTcLmUVcpras8-m8pdkfpE=_r0spz1S50iMF@mail.gmail.com>
	<AANLkTi=MoV27FW0Ukd6K+mqmgnkyNgb9hjFx-vBYV1ak@mail.gmail.com>
	<AANLkTimCRfKSWG+5OdQY0=HTt2kOqxwk+0vKvaBNdmAd@mail.gmail.com>
	<AANLkTikfpRi9Shxiqd2w5nz3ZHsBnrHTt+=rHHnvoDJn@mail.gmail.com>
	<AANLkTikH3JJCCpL5s_eF2qjjttnEetE0i=nPn87zhBUL@mail.gmail.com>
	<AANLkTimzkb7qY8nW3M2afTPn18=OxY6jQye8D9TKr9Ts@mail.gmail.com>
	<AANLkTimR2x5AQPNiDDt4fZ8F=z_Wep3TBVR9Lob_72OA@mail.gmail.com>
	<AANLkTi=JgCME8u=27kUyLGU3cMP2nZpFpu7Zd5-4tpj0@mail.gmail.com>
	<AANLkTikL=kvbfLQx3E-vCaErRM-ARnU3_YSAvKo67Uvy@mail.gmail.com>
	<AF5D2E2E-F8A7-44AB-8EA0-D240A28A51F6@mgtech.com>
	<AANLkTikkFaJ-EWipGij2UAzMpmH8JGyhj+bUNKSDS8T5@mail.gmail.com>
	<AANLkTin=hRwNSBLxTRdJ6SqcPedbHiDgAJPCFESM9nCT@mail.gmail.com>
	<AANLkTi=+ggi1L7oAhKGFFR8f=QaXhoj_fMGPJ1pUFk+v@mail.gmail.com>
	<AANLkTin9X9vXL0Sg8BNPO9opuEzSq=iLChwVypuSec2x@mail.gmail.com>
	<AANLkTi=KL7w=ZeudXyEPu3S78CadmM88HfFFrzkEkcO3@mail.gmail.com>
	<BC3127AD-18EB-4A1E-9A80-9A2BF4C55FE5@mgtech.com>
	<AANLkTinvSh0DJJAWxv_A+=WgcVubu1u5EKQMo-kaKVDp@mail.gmail.com>
Message-ID: <AANLkTikN166nWqGbcqreqPxGR+HmMp9wZAEDYP_wcsqk@mail.gmail.com>

Thanks a lot Rupa,

Seems NAT-PMP was causing my problems.  I turned this off and now I get the
upnp keep alive debug traces as expected and the ip addresses are updated
correctly in sofia and reregistration occurs.

THE END!


On Thu, Oct 28, 2010 at 3:03 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> Mario, your issue is a different (unrelated) issue.
>
>
> On Wed, Oct 27, 2010 at 10:47 AM, Mario G <mario_fs at mgtech.com> wrote:
>
>> FYI, same thing I seeing on my R042 problem. sofia status shows correct
>> (static) external IP in internal and gateway profiles, but nat_map has the
>> dynamic IP.
>>
>> On Oct 26, 2010, at 10:13 PM, Mark Campbell-Smith wrote:
>>
>> That gives me quite a bit....
>>
>> Can there be the case that nat_map status has a different IP address to
>> what I would see in sofia status profile internal?  And does sofia
>> reregister if there is detected a public IP address change?
>>
>> Thanks
>>
>> freeswitch at internal> nat_map reinit
>> [DEBUG] libs/esl/src/esl.c:1141 esl_send() SEND
>> api nat_map reinit
>> console_execute: true
>>
>>
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type]
>> = [api/response]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER
>> [Content-Length] = [130]
>> Nat Type: NAT-PMP, ExtIP: 210.xx.xxx.xx
>> port,proto,proto_num,sticky
>> 5060,udp,0,0
>> 5060,tcp,1,0
>> 5080,udp,0,0
>> 5080,tcp,1,0
>>
>> 4 total.
>>
>> freeswitch at internal> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV
>> HEADER [Content-Type] = [log/data]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER
>> [Content-Length] = [68]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Level] =
>> [6]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Text-Channel]
>> = [0]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-File] =
>> [switch_nat.c]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Func] =
>> [switch_nat_init]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Line] =
>> [410]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [User-Data] =
>> []
>> [DEBUG] libs/esl/src/esl.c:1115 esl_recv_event() RECV MESSAGE
>> Event-Name: SOCKET_DATA
>> Content-Type: log/data
>> Content-Length: 68
>> Log-Level: 6
>> Text-Channel: 0
>> Log-File: switch_nat.c
>> Log-Func: switch_nat_init
>> Log-Line: 410
>> User-Data: _undef_
>>
>> 2010-10-27 16:10:07.003208 [INFO] switch_nat.c:410 Scanning for NAT
>>
>> 2010-10-27 16:10:07.003208 [INFO] switch_nat.c:410 Scanning for NAT
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type]
>> = [log/data]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER
>> [Content-Length] = [73]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Level] =
>> [7]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Text-Channel]
>> = [0]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-File] =
>> [switch_nat.c]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Func] =
>> [get_pmp_pubaddr]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Line] =
>> [167]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [User-Data] =
>> []
>> [DEBUG] libs/esl/src/esl.c:1115 esl_recv_event() RECV MESSAGE
>> Event-Name: SOCKET_DATA
>> Content-Type: log/data
>> Content-Length: 73
>> Log-Level: 7
>> Text-Channel: 0
>> Log-File: switch_nat.c
>> Log-Func: get_pmp_pubaddr
>> Log-Line: 167
>> User-Data: _undef_
>>
>> 2010-10-27 16:10:07.003208 [DEBUG] switch_nat.c:167 Checking for PMP 1/5
>>
>> 2010-10-27 16:10:07.003208 [DEBUG] switch_nat.c:167 Checking for PMP 1/5
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Content-Type]
>> = [log/data]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER
>> [Content-Length] = [98]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Level] =
>> [6]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Text-Channel]
>> = [0]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-File] =
>> [switch_nat.c]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Func] =
>> [switch_nat_init]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [Log-Line] =
>> [423]
>> [DEBUG] libs/esl/src/esl.c:989 esl_recv_event() RECV HEADER [User-Data] =
>> []
>> [DEBUG] libs/esl/src/esl.c:1115 esl_recv_event() RECV MESSAGE
>> Event-Name: SOCKET_DATA
>> Content-Type: log/data
>> Content-Length: 98
>> Log-Level: 6
>> Text-Channel: 0
>> Log-File: switch_nat.c
>> Log-Func: switch_nat_init
>> Log-Line: 423
>> User-Data: _undef_
>>
>> 2010-10-27 16:10:07.013797 [INFO] switch_nat.c:423 NAT detected type: pmp,
>> ExtIP: '210.xx.xxx.xx'
>>
>> 2010-10-27 16:10:07.013797 [INFO] switch_nat.c:423 NAT detected type: pmp,
>> ExtIP: '210.xx.xxx.xx'
>>
>>
>> On Wed, Oct 27, 2010 at 4:01 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>>> Then I have no idea why you aren't seeing upnp messages on the console.
>>>  What do you see when you do a "nat_map reinit" ?
>>>
>>>
>>> On Tue, Oct 26, 2010 at 8:09 PM, Mark Campbell-Smith <
>>> mcampbellsmith at gmail.com> wrote:
>>>
>>>> I see debug from sofia.c, switch_core_state_machine, mod_sofia, diaplan
>>>> parsing, sofia_glue, switch_rtp, switch_ivr_play_say etc.
>>>>
>>>>
>>>>
>>>> On Wed, Oct 27, 2010 at 11:41 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>>
>>>>> Do you see ANY debug messages on your console from any other FS module?
>>>>>  fsctl loglevel 7 sets the base log level, but not necessarily the log level
>>>>> on your screen.
>>>>>
>>>>> /log 7
>>>>>
>>>>> sets the log level in fs_cli.
>>>>>
>>>>> Your screen should get VERY busy especially when processing a call.
>>>>>
>>>>> On Tue, Oct 26, 2010 at 4:21 PM, Mark Campbell-Smith <
>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>
>>>>>> I have it set to debug in switch.conf.xml and enable fsctl loglevel 7
>>>>>> in fs_cli....
>>>>>>
>>>>>> :~$ cat /usr/local/freeswitch/conf/autoload_configs/switch.conf.xml |
>>>>>> egrep "param.*loglevel"
>>>>>>     <param name="loglevel" value="debug"/>
>>>>>>
>>>>>> With regards to sofia: how does it determine if the external IP has
>>>>>> changed?  And when it does, does sofia initiate a reregister of all external
>>>>>> sip providers?
>>>>>>
>>>>>> Thanks!
>>>>>>
>>>>>> On Wed, Oct 27, 2010 at 3:14 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>>
>>>>>>> What you said: "And nat_map status shows me which ports have been
>>>>>>> forwarded and the external ip address (which is sometimes wrong).  I also
>>>>>>> see this in the router gui." is also happening to me on a Linksys/Cisco
>>>>>>> RV042 router. Can't keep FS working for more than 2-5 hours without doing a
>>>>>>> restart or nat_map reinit. I think the routers upnp has a problem with dual
>>>>>>> wans, I have one static and one dynamic DSL.
>>>>>>>
>>>>>>> On Oct 26, 2010, at 6:43 AM, Rupa Schomaker wrote:
>>>>>>>
>>>>>>> What do you have set for loglevel in switch.conf.xml?  My guess is
>>>>>>> that you have it set to something other than debug.  If that is set to (say)
>>>>>>> info, then info is as low as you can go.  Setting the console log level to
>>>>>>> debug will not give you debug output.
>>>>>>>
>>>>>>> sofia is unaware of upnp so a keepalive packet should have no impact
>>>>>>> on it's operation.  I'm not familiar enough with sofia to tell what triggers
>>>>>>> that event.
>>>>>>>
>>>>>>> On Tue, Oct 26, 2010 at 6:24 AM, Mark Campbell-Smith <
>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>
>>>>>>>> OK..  tcpdump gave me this and I still do not see anything on FS
>>>>>>>> except for this line:
>>>>>>>>
>>>>>>>> 2010-10-26 22:02:51.235608 [DEBUG] sofia.c:957 nua_i_outbound:
>>>>>>>> unknown event 8: 102 NAT binding changed
>>>>>>>>
>>>>>>>> Hypertext Transfer Protocol
>>>>>>>>     NOTIFY * HTTP/1.1\r\n
>>>>>>>>     HOST:239.255.255.250:1900\r\n
>>>>>>>>     Cache-Control:max-age=120\r\n
>>>>>>>>     Location:http://192.168.1.1:1278/rootDesc.xml\r\n<http://192.168.1.1:1278/rootDesc.xml%5Cr%5Cn>
>>>>>>>>     Server: Tomato UPnP/1.0 MiniUPnPd/1.4\r\n
>>>>>>>>     NT:urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>>>>>>>>
>>>>>>>>  USN:uuid:882dbe81-c28a-4992-baf7-410c05f1caf4::urn:schemas-upnp-org:service:WANIPConnection:1\r\n
>>>>>>>>     NTS:ssdp:alive\r\n
>>>>>>>>     \r\n
>>>>>>>>
>>>>>>>> This looks the same as you posted before, so why do I get the
>>>>>>>> unknown event 8 (is this related)?
>>>>>>>>
>>>>>>>> Thanks!
>>>>>>>>
>>>>>>>> On Tue, Oct 26, 2010 at 3:32 PM, Mark Campbell-Smith <
>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> :)  that was the first thing I did, but I get nothing; no keep
>>>>>>>>> alive packets.
>>>>>>>>>
>>>>>>>>> in the logfile at startup I see:
>>>>>>>>>
>>>>>>>>> 2010-10-26 10:19:45.484258 [INFO] switch_nat.c:410 Scanning for NAT
>>>>>>>>> 2010-10-26 10:19:45.485991 [DEBUG] switch_nat.c:167 Checking for
>>>>>>>>> PMP 1/5
>>>>>>>>> 2010-10-26 10:19:45.488575 [INFO] switch_nat.c:423 NAT detected
>>>>>>>>> type: pmp, ExtIP: '203.xxx.xxx.xxx'
>>>>>>>>> 2010-10-26 10:19:45.490683 [DEBUG] switch_nat.c:256 NAT thread
>>>>>>>>> configured
>>>>>>>>> 2010-10-26 10:19:45.492180 [DEBUG] switch_nat.c:267 NAT thread
>>>>>>>>> started
>>>>>>>>>
>>>>>>>>> And nat_map status shows me which ports have been forwarded and the
>>>>>>>>> external ip address (which is sometimes wrong).  I also see this in the
>>>>>>>>> router gui.
>>>>>>>>>
>>>>>>>>> But I never see the keep alive packets.
>>>>>>>>>
>>>>>>>>> On Tue, Oct 26, 2010 at 3:13 PM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>>
>>>>>>>>>> Just turn up console logging to level 7.
>>>>>>>>>>
>>>>>>>>>> /log 7 if using fs_cli
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Mon, Oct 25, 2010 at 6:53 PM, Mark Campbell-Smith <
>>>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> Hi!
>>>>>>>>>>>
>>>>>>>>>>> How do I enable debugging to get the UPnP keep alive packets?
>>>>>>>>>>>
>>>>>>>>>>> Thanks!
>>>>>>>>>>>
>>>>>>>>>>> On Tue, Oct 26, 2010 at 1:15 AM, Rupa Schomaker <rupa at rupa.com>wrote:
>>>>>>>>>>>
>>>>>>>>>>>> I need the debug logging from the upnp stuff.  It should look
>>>>>>>>>>>> something like:
>>>>>>>>>>>>
>>>>>>>>>>>> +OK log level 7 [7]
>>>>>>>>>>>> freeswitch at internal> 2010-10-25 09:13:59.926861 [DEBUG]
>>>>>>>>>>>> switch_nat.c:299 got UPnP keep alive packet:
>>>>>>>>>>>> NOTIFY * HTTP/1.1
>>>>>>>>>>>> HOST:239.255.255.250:1900
>>>>>>>>>>>> Cache-Control:max-age=60
>>>>>>>>>>>> Location:http://192.168.1.1:5000/rootDesc.xml
>>>>>>>>>>>> Server: Tomato UPnP/1.0 MiniUPnPd/1.4
>>>>>>>>>>>> NT:urn:schemas-upnp-org:service:WANIPConnection:1
>>>>>>>>>>>>
>>>>>>>>>>>> USN:uuid:00000000-0000-0000-0000-000000000000::urn:schemas-upnp-org:service:WANIPConnection:1
>>>>>>>>>>>> NTS:ssdp:alive
>>>>>>>>>>>>
>>>>>>>>>>>> I also run tomato and haven't had trouble with the upnp support
>>>>>>>>>>>> so at least we have that part working.
>>>>>>>>>>>>
>>>>>>>>>>>> You should see the above keepalive appear periodically every 30s
>>>>>>>>>>>> or so.  You should see another set of messages when you terminate the DSL
>>>>>>>>>>>> connection and when that dsl connection comes back online.
>>>>>>>>>>>>
>>>>>>>>>>>> On Sat, Oct 23, 2010 at 5:36 PM, Mark Campbell-Smith <
>>>>>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> Hi!
>>>>>>>>>>>>>
>>>>>>>>>>>>> The router is a ASUS router but it is running the tomato
>>>>>>>>>>>>> firmware which runs miniupnd.
>>>>>>>>>>>>>
>>>>>>>>>>>>> In the debug I see the message: 2010-10-24 08:02:59.178918
>>>>>>>>>>>>> [DEBUG] sofia.c:956 nua_i_outbound: unknown event 8: 101 NAT detected
>>>>>>>>>>>>>
>>>>>>>>>>>>> This is what I see below. I hope I enabled all debug messages.
>>>>>>>>>>>>>  I thought I saw some xml type messages earlier, but not sure how I enabled
>>>>>>>>>>>>> them.
>>>>>>>>>>>>>
>>>>>>>>>>>>> nta_outgoing: RTT is 67.958 ms
>>>>>>>>>>>>> outbound(0xb6e07c00): NAT binding changed: [210.xx.xxx.xx]:5080
>>>>>>>>>>>>> != [58.xxx.xx.xx]:5080
>>>>>>>>>>>>> nua(0xb6e07c00): event i_outbound 102 NAT binding changed
>>>>>>>>>>>>> nua: nua_application_event: entering
>>>>>>>>>>>>> 2010-10-24 09:05:08.849525 [DEBUG] sofia.c:956 nua_i_outbound:
>>>>>>>>>>>>> unknown event 8: 102 NAT binding changed
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Sun, Oct 24, 2010 at 1:08 AM, Rupa Schomaker <rupa at rupa.com
>>>>>>>>>>>>> > wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>> btw: what router are you using?
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Sat, Oct 23, 2010 at 9:06 AM, Rupa Schomaker <
>>>>>>>>>>>>>> rupa at rupa.com> wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> In a upnp config, FS is depending on the router to notify it
>>>>>>>>>>>>>>> of the ip address change.  It is possible the upnp message(s) aren't
>>>>>>>>>>>>>>> formatted as expected.  If you turn on debug logging you'll get the upnp
>>>>>>>>>>>>>>> messages on the console.  If you can pastebin the capture of those while
>>>>>>>>>>>>>>> dropping and restarting the dsl connection I can maybe see if there is
>>>>>>>>>>>>>>> something obvious going on.  The debug should also have messages related to
>>>>>>>>>>>>>>> the processing of those upnp messages.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> On Sat, Oct 23, 2010 at 7:04 AM, Mark Campbell-Smith <
>>>>>>>>>>>>>>> mcampbellsmith at gmail.com> wrote:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> HI!
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> I know this question must have been answered 100's of
>>>>>>>>>>>>>>>> times....
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> My adsl is a bit dodgy at the moment and tends to go up and
>>>>>>>>>>>>>>>> down; which means I get assigned a new ip address from my ISP.  FS is nat'd
>>>>>>>>>>>>>>>> behind a upnp capable router.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> The problem is that FS does not seem to be detecting the
>>>>>>>>>>>>>>>> change.   For both the internal and external profile, I have auto-nat setup,
>>>>>>>>>>>>>>>> and I have tried stun and host settings in vars.conf.  When I issue
>>>>>>>>>>>>>>>> a nat_map status I see my old IP address, and if I then issue a nat_map
>>>>>>>>>>>>>>>> reinit, I see the new public IP address.  How can I get this to be
>>>>>>>>>>>>>>>> automatic?
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Also the public IP address shown in the nat_map status is
>>>>>>>>>>>>>>>> not the same as shown in the sofia profile internal/external printouts.  Why
>>>>>>>>>>>>>>>> is this?
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> I'm sure this is configuration, just not sure what to
>>>>>>>>>>>>>>>> change.
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Thanks!
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>> -Rupa
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> --
>>>>>>>>>>>>>> -Rupa
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> --
>>>>>>>>>>>> -Rupa
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> --
>>>>>>>>>> -Rupa
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> -Rupa
>>>>>>>  _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> -Rupa
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From babstar99 at gmail.com  Wed Oct 27 22:04:38 2010
From: babstar99 at gmail.com (Babstar)
Date: Thu, 28 Oct 2010 16:04:38 +1100
Subject: [Freeswitch-users] Memory problems
In-Reply-To: <AANLkTimHXVGdoBurEAw6NtFN8PnJDa649wib8_9qTTzC@mail.gmail.com>
References: <AANLkTina_J3XMLcrTweeKVw6AaVjJaCsvytFGSo+iPfv@mail.gmail.com>
	<AANLkTimHXVGdoBurEAw6NtFN8PnJDa649wib8_9qTTzC@mail.gmail.com>
Message-ID: <AANLkTi=g9ySTLppUg-urCRThPea=+8G0=AwQkieYNwt4@mail.gmail.com>

On Tue, Oct 26, 2010 at 6:23 AM, Michael Collins <msc at freeswitch.org> wrote:

> Look at the *.db files. Are any of them getting really large?
> -MC
>
> Michael,
               just had the latest failure, nothing unusually large with the
*.db files:
16K     /usr/local/freeswitch/db/call_limit.db
148K    /usr/local/freeswitch/db/core.db
8.0K    /usr/local/freeswitch/db/fifo.db
76K     /usr/local/freeswitch/db/sofia_reg_external.db
504K    /usr/local/freeswitch/db/sofia_reg_internal.db
8.0K    /usr/local/freeswitch/db/sofia_reg_internal.db-journal
76K     /usr/local/freeswitch/db/sofia_reg_internal-ipv6.db
16K     /usr/local/freeswitch/db/voicemail_default.db


When I restarted the daemon, the usage over a few minutes jump up to 24%,
 only a reboot could  fix it.

--
Babstar
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From thisjoy0528 at gmail.com  Thu Oct 28 04:07:21 2010
From: thisjoy0528 at gmail.com (joy this)
Date: Thu, 28 Oct 2010 19:07:21 +0800
Subject: [Freeswitch-users]  problem with jingaling (may cause by TLS)
Message-ID: <AANLkTik87R0ZLuM0zhY8F4nsV3eiWkHHPKB0qNdvf3RC@mail.gmail.com>

Dear all:



I am confused with dingaling for several days. I hope someone could help me
please.

Below are my config steps:

1.          Download ?gnutls-2.9.9.exe? from ?
http://josefsson.org/gnutls4win/? and install.

2.          Build ?libgnutls-26.lib? (refer to ?
http://wiki.freeswitch.org/wiki/Ixemel_MSVS_project_example?). I have
rebuilt the FreeSwitch.

3.          Copy the TLS dlls to ?C:\FreeSWITCH\Debug?.

4.          Set the conf in ?\conf\jingle_profiles\client.xml? (refer to ?
http://wiki.freeswitch.org/wiki/Mod_dingaling#TLS? and the letter
?[Freeswitch-users] Problems making a receiveing calls with mod_jingling? ).

5.          Start FreeSwitch.


Below are my config in step 3:

<include>

  <profile type="client">

    <param name="name" value="thisjoy"/>

    <param name="login" value="account at gmail.com/gtalk"/>

    <param name="password" value="pwd"/>

    <param name="dialplan" value="XML"/>

    <param name="context" value="public"/>

    <param name="message" value="Jingle all the way"/>

    <param name="rtp-ip" value="$${bind_server_ip}"/>

    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>

    <param name="auto-login" value="true"/>

    <param name="sasl" value="plain"/>

    <param name="server" value="talk.google.com"/>

    <param name="tls" value="true"/>

    <param name="use-rtp-timer" value="true"/>

    <param name="exten" value="888"/>

    <param name="vad" value="both"/>

    <param name="candidate-acl" value="wan.auto"/>

    <param name="local-network-acl" value="localnet.auto"/>

  </profile>

</include>



I input the command ?dingaling status? and found the account was
?UNCONNECTED?. Besides, I saw a message in ?freeswitch.log?, it is:

I. 2010-10-28 18:46:56.468750 [DEBUG] libdingaling.c:1237 TLS NOT SUPPORTED
IN THIS BUILD!

II. 2010-10-28 18:47:16.453125 [DEBUG] libdingaling.c:1607 io error 2 7
retry in 20 second(s)


My OS is Windows XP and the FreeSwitch is 1.0.head. Thank you for helping.



Sincerely yours,

Thisjoy.
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From grant at mrit.co.za  Thu Oct 28 06:03:54 2010
From: grant at mrit.co.za (Grant Gray)
Date: Thu, 28 Oct 2010 15:03:54 +0200
Subject: [Freeswitch-users] Freeswitch and Digium
Message-ID: <mailman.0.1288272139.19149.freeswitch-users@lists.freeswitch.org>

 

Hi All,

 

I am trying to get a digium TDM410 card to work. I have looked at the
following tutorials:

 

http://wiki.sangoma.com/wanpipe-linux-drivers

http://wiki.sangoma.com/wanpipe-api-freetdm-linux

 

and a few others.

 

The problem I am having is that this command does nothing:

 

export FREETDM_MOD_DIR=/usr/local/freetdm

 

it does not create the directory as required. Is this normal on Centos or is
there another command?

 

I also tried this command of one of the forum posts:

 

make dahdi DAHDI_DIR=/usr/src/dahdi
(http://wiki.sangoma.com/Topic-1273333936389)

 

I poked around further and found that the export -p command shows that the
command is there:

 

declare -x FREETDM_MOD_DIR="/usr/local/freetdm"

 

So, I am wondering why its not working. I also cannot find a command like
/sbin/export so i am lost.

 

what I am really trying to achieve is the installation of the tdm410 card on
freeswitch. Any tutorials would be really great.

 

Regards,

 

Gigg

 

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From freeswitch at peely.com  Thu Oct 28 06:27:44 2010
From: freeswitch at peely.com (peely)
Date: Thu, 28 Oct 2010 06:27:44 -0700 (PDT)
Subject: [Freeswitch-users] Problem with T.38 fax receive using ESL
Message-ID: <1288272464502-5682561.post@n2.nabble.com>


Hi,

I'm having problems receiving faxes through the Event Socket API using T.38
and SpanDSP. If I use the example dialplan entry:

    <extension name="NTS TESTfax">
        <condition field="destination_number" expression="^\+(testfax)">
            <action application="answer" />
            <action application="playback" data="silence_stream://2000"/>
            <action application="set" data="fax_enable_t38_request=true"/>
            <action application="set" data="fax_enable_t38=true"/>
            <action application="rxfax" data="/tmp/faxes/FAX-${uuid}.tif"/>
            <action application="hangup"/>
        </condition>
    </extension>


Then everything is OK and a fax is received nicely whereas if I use an ESL
command on an answered call like:

SendMsg 5716720c-e291-11df-939c-75e5ad3b2de7
call-command: execute
execute-app-name: rxfax
execute-app-arg:
{fax_enable_t38_request=true,fax_enable_t38=true,fax_header=testheader,fax_ident=testident}/tmp/faxes/myfax.tif

Then there's lots of T.38 mucking about but the fax terminates with a
communications error. I've tried sleeping for 2 seconds, broadcasting
silence_stream://2000 etc but to no avail.

Has anyone successfully got T.38 fax receive working through the ESL, and if
so, what chain of commands do you execute?

I'm using a GIT trunk circa Oct 15th BTW.



Many thanks,



Neil.

-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-T-38-fax-receive-using-ESL-tp5682561p5682561.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From testeador01 at gmail.com  Thu Oct 28 06:29:12 2010
From: testeador01 at gmail.com (Milena)
Date: Thu, 28 Oct 2010 08:29:12 -0500
Subject: [Freeswitch-users] Moose Penis? Wiki Installation question
In-Reply-To: <AANLkTi=-iw8PdxZ7a7MrK0u_C0TkLysgHh-kqTrqus5j@mail.gmail.com>
References: <AANLkTi=-iw8PdxZ7a7MrK0u_C0TkLysgHh-kqTrqus5j@mail.gmail.com>
Message-ID: <AANLkTikgkyLdpFHmg5Lw5BpSbu=s0p45T5uTg8HH358+@mail.gmail.com>

The recommended download is the one specified under the "Recommended
Download" title in that same page which is the latest git head.

1.0.6 is really old, can you please edit the parts of the wiki that confused
you into thinking you were meant to download 1.0.6?

On Wed, Oct 27, 2010 at 11:13 AM, Matty Sarro <msarro at gmail.com> wrote:

> So after checking the wiki on installation (
> http://wiki.freeswitch.org/wiki/Installation_Guide), I attempted to go to
> the latest build version. The link includes a joke about moose penis, and
> then a comment to just use 1.0.6. If we're just supposed to use 1.0.6, why
> does the installation guide keep mentioning to use the latest build or the
> latest git source to install? It seems to contradict itself.
>
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From gmaruzz at gmail.com  Thu Oct 28 06:34:32 2010
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 28 Oct 2010 15:34:32 +0200
Subject: [Freeswitch-users] Moose Penis? Wiki Installation question
In-Reply-To: <AANLkTi=-iw8PdxZ7a7MrK0u_C0TkLysgHh-kqTrqus5j@mail.gmail.com>
References: <AANLkTi=-iw8PdxZ7a7MrK0u_C0TkLysgHh-kqTrqus5j@mail.gmail.com>
Message-ID: <AANLkTimYeQOveEp1NHP-wWeW_dGCMgshafS9HA_oxCBV@mail.gmail.com>

On Wed, Oct 27, 2010 at 6:13 PM, Matty Sarro <msarro at gmail.com> wrote:
> So after checking the wiki on installation
> (http://wiki.freeswitch.org/wiki/Installation_Guide), I attempted to go to
> the latest build version. The link includes a joke about moose penis, and
> then a comment to just use 1.0.6. If we're just supposed to use 1.0.6, why
> does the installation guide keep mentioning to use the latest build or the
> latest git source to install? It seems to contradict itself.

Hey, me too want the 1.0.9 version !

That's the one with the latest features!

Where did you find the link to 1.0.9, codename "moose penis"?

-giovanni



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



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From dujinfang at gmail.com  Thu Oct 28 06:41:03 2010
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 28 Oct 2010 21:41:03 +0800
Subject: [Freeswitch-users] FS incoming calls Queue ACD implement
In-Reply-To: <AANLkTinuy+DhD0mZGj0BWOMx2Ha8sTXLetnQWLNK9PX6@mail.gmail.com>
References: <1288212978229-5680281.post@n2.nabble.com>
	<AANLkTinuCX6xrzXHEg1pA43_TXnauSr4EDpnQhmRdPe-@mail.gmail.com>
	<AANLkTinuy+DhD0mZGj0BWOMx2Ha8sTXLetnQWLNK9PX6@mail.gmail.com>
Message-ID: <AANLkTikEf2AhP6n4Lg+jS2t-oG-fxhZU9aMfjn80gxMa@mail.gmail.com>

You shout get the git HEAD to use mod_callcenter. Also you need to
uncomment it in modules to enable build.

On Thu, Oct 28, 2010 at 10:58 AM, Johnson <johnsonw at eallway.com> wrote:
> Hi MC,
>
> Thanks for your feedback quickly. Actually i have tested mod_fifo and met
> the callerid can't transfer to agent's phone before the agent pickup.
> Traditional phones almost don't support new features like getting sip
> "UPDATE MESSAGE". As alternatively, i try to use Moc' mod_callcenter, but
> unfortunitely, there is no mod_callcenter package in FS1.06, even i put in
> dialplan
> <action application="callcenter" data="support at default"/>, it show log:
> ?[ERR] switch_core_session.c:1731 Invalid Application callcenter
> Is it meaning that the mod_callcenter is still going on coding? Any further
> more helps would be much appreciated.
>
> Thanks,
>
> Johnson
>
>
> On Thu, Oct 28, 2010 at 7:37 AM, mercutioviz [via freeswitch-users] <[hidden
> email]> wrote:
>>
>> I'm afraid this question is a bit broad. I notice that you are trying to
>> avoid the use of mod_fifo - is there a reason for that? Alternatively you
>> can use Moc's newer ACD system: mod_callcenter. It is more of a traditional
>> ACD than mod_fifo. (Both get inbound calls sent to agents, they just differ
>> in the methodology used to accomplish that task.)
>>
>> I've found the documentation on mod_callcenter to be refreshingly detailed
>> and complete:
>> http://wiki.freeswitch.org/wiki/Mod_callcenter
>>
>> -MC
>>
>> On Wed, Oct 27, 2010 at 1:56 PM, Johnson <[hidden email]> wrote:
>>>
>>> Hi FS Group,
>>> I am a new vanilla. Just want to realize Queue ACD function in
>>> Freeswitch.
>>> Is any one know how to configure or create the script to process incoming
>>> calls put into defined Queues without using mod_FIFO. Any additional
>>> prompting is very appreciated.
>>>
>>> Johnson
>>> --
>>> View this message in context:
>>> http://freeswitch-users.2379917.n2.nabble.com/FS-incoming-calls-Queue-ACD-implement-tp5680281p5680281.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> [hidden email]
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> [hidden email]
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> ________________________________
>> View message @
>> http://freeswitch-users.2379917.n2.nabble.com/FS-incoming-calls-Queue-ACD-implement-tp5680281p5680777.html
>> To unsubscribe from FS incoming calls Queue ACD implement, click here.
>
>
> ________________________________
> View this message in context: Re: FS incoming calls Queue ACD implement
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From steveayre at gmail.com  Thu Oct 28 06:51:51 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 28 Oct 2010 14:51:51 +0100
Subject: [Freeswitch-users] FS incoming calls Queue ACD implement
In-Reply-To: <AANLkTikEf2AhP6n4Lg+jS2t-oG-fxhZU9aMfjn80gxMa@mail.gmail.com>
References: <1288212978229-5680281.post@n2.nabble.com>
	<AANLkTinuCX6xrzXHEg1pA43_TXnauSr4EDpnQhmRdPe-@mail.gmail.com>
	<AANLkTinuy+DhD0mZGj0BWOMx2Ha8sTXLetnQWLNK9PX6@mail.gmail.com>
	<AANLkTikEf2AhP6n4Lg+jS2t-oG-fxhZU9aMfjn80gxMa@mail.gmail.com>
Message-ID: <AANLkTinxF6O_P2qHrTRcaqVLwrO4zkaedzHrfOJQ0aKa@mail.gmail.com>

Git head also has many bugfixes since 1.0.6 so it's a good idea to
update anyway.

-Steve



On 28 October 2010 14:41, Seven Du <dujinfang at gmail.com> wrote:
> You shout get the git HEAD to use mod_callcenter. Also you need to
> uncomment it in modules to enable build.
>
> On Thu, Oct 28, 2010 at 10:58 AM, Johnson <johnsonw at eallway.com> wrote:
>> Hi MC,
>>
>> Thanks for your feedback quickly. Actually i have tested mod_fifo and met
>> the callerid can't transfer to agent's phone before the agent pickup.
>> Traditional phones almost don't support new features like getting sip
>> "UPDATE MESSAGE". As alternatively, i try to use Moc' mod_callcenter, but
>> unfortunitely, there is no mod_callcenter package in FS1.06, even i put in
>> dialplan
>> <action application="callcenter" data="support at default"/>, it show log:
>> ?[ERR] switch_core_session.c:1731 Invalid Application callcenter
>> Is it meaning that the mod_callcenter is still going on coding? Any further
>> more helps would be much appreciated.
>>
>> Thanks,
>>
>> Johnson
>>
>>
>> On Thu, Oct 28, 2010 at 7:37 AM, mercutioviz [via freeswitch-users] <[hidden
>> email]> wrote:
>>>
>>> I'm afraid this question is a bit broad. I notice that you are trying to
>>> avoid the use of mod_fifo - is there a reason for that? Alternatively you
>>> can use Moc's newer ACD system: mod_callcenter. It is more of a traditional
>>> ACD than mod_fifo. (Both get inbound calls sent to agents, they just differ
>>> in the methodology used to accomplish that task.)
>>>
>>> I've found the documentation on mod_callcenter to be refreshingly detailed
>>> and complete:
>>> http://wiki.freeswitch.org/wiki/Mod_callcenter
>>>
>>> -MC
>>>
>>> On Wed, Oct 27, 2010 at 1:56 PM, Johnson <[hidden email]> wrote:
>>>>
>>>> Hi FS Group,
>>>> I am a new vanilla. Just want to realize Queue ACD function in
>>>> Freeswitch.
>>>> Is any one know how to configure or create the script to process incoming
>>>> calls put into defined Queues without using mod_FIFO. Any additional
>>>> prompting is very appreciated.
>>>>
>>>> Johnson
>>>> --
>>>> View this message in context:
>>>> http://freeswitch-users.2379917.n2.nabble.com/FS-incoming-calls-Queue-ACD-implement-tp5680281p5680281.html
>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> [hidden email]
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> [hidden email]
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> ________________________________
>>> View message @
>>> http://freeswitch-users.2379917.n2.nabble.com/FS-incoming-calls-Queue-ACD-implement-tp5680281p5680777.html
>>> To unsubscribe from FS incoming calls Queue ACD implement, click here.
>>
>>
>> ________________________________
>> View this message in context: Re: FS incoming calls Queue ACD implement
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Blog: http://www.dujinfang.com
> Proj:? http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From renjian at gmail.com  Thu Oct 28 07:40:47 2010
From: renjian at gmail.com (Jian Ren)
Date: Thu, 28 Oct 2010 10:40:47 -0400
Subject: [Freeswitch-users] Running FS on Segate dockstar
Message-ID: <AANLkTi=8HFMCc_PSkMPGhC4sph0C-vU3Vb0J=J+5mTCM@mail.gmail.com>

Hi,
Did anyone compile FS successfully on dockstar? What about skypopen?
Thanks!
Jian
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From jeff at jefflenk.com  Thu Oct 28 07:56:41 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 28 Oct 2010 07:56:41 -0700 (PDT)
Subject: [Freeswitch-users] problem with jingaling (may cause by TLS)
In-Reply-To: <AANLkTik87R0ZLuM0zhY8F4nsV3eiWkHHPKB0qNdvf3RC@mail.gmail.com>
References: <AANLkTik87R0ZLuM0zhY8F4nsV3eiWkHHPKB0qNdvf3RC@mail.gmail.com>
Message-ID: <1288277801895-5682888.post@n2.nabble.com>


This is not currently supported in the current windows iksemel project. The
required code for tls has been conditionally compiled out.

Do a search for WIN32 in the project!

If you are able to make this work please post your recipe to the Wiki so we
can correct the docs there.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/problem-with-jingaling-may-cause-by-TLS-tp5682569p5682888.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Thu Oct 28 07:59:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 28 Oct 2010 09:59:23 -0500
Subject: [Freeswitch-users] Loss of first second of media
In-Reply-To: <AANLkTikiVxFAOwftrmfqd3LtD4H__UUytsM117WBhZy0@mail.gmail.com>
References: <AANLkTimECwtVmnbFsBJRiN3YSaGyXbvh-sKtKYPey8xx@mail.gmail.com>
	<AANLkTik5H74MupY8iyx6Ao4=Rv=wCm8jv-w=xCtSnm0Z@mail.gmail.com>
	<AANLkTikpVH-s6AvJ7fQoZu2PE4dduTg+d-ygyBWSfr4R@mail.gmail.com>
	<AANLkTi=FZDqev9R1aTMpdvTk8t9NfxiyeisQSRBxfTEb@mail.gmail.com>
	<AANLkTikY68ovuiSO9xkbZZPSskxCNsjrpTSdb-42bwJB@mail.gmail.com>
	<AANLkTikiG3xPKDVkwU4GDk5t6KDe67AQmM-dkheMCSu7@mail.gmail.com>
	<AANLkTinJRcpRx_BQyKifECcurYb+vG2P8Nct1xT_X3XO@mail.gmail.com>
	<AANLkTinpaRivNHteDp4i6WJjmgsPAE6otVx5LjgJcvWr@mail.gmail.com>
	<AANLkTikiVxFAOwftrmfqd3LtD4H__UUytsM117WBhZy0@mail.gmail.com>
Message-ID: <AANLkTikLtKhBiCZeMY+GH6X7jb4wVdybVF=OiMePK8Rm@mail.gmail.com>

Can you do this trace with debug level logging in addition to the sip trace
console loglevel debug

you also may want to get a pcap of it

tshark udp and port 5060 -w test.pcap



On Sat, Oct 23, 2010 at 11:44 AM, Fraser Redmond
<fraserredmond at gmail.com> wrote:
> Thanks Anthony,
>
> Finally managed to get a sip trace - could you do me a favor and take a look
> and/or give me some ideas of what to look for?
>
> http://pastebin.freeswitch.org/14300
>
> I've highlighted lines 168 and 193. In between these lines is where the
> number is dialed and rings once, then picks up, then theres silence for a
> second or two, and that second SIP message is when I start hearing audio.
>
> Thanks,
> Fraser
>
>
>
>
> On Thu, Oct 21, 2010 at 6:44 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> its a blue message on cli
>>
>> It could also be the other side expecting us to send media first or
>> something silly.
>> try getting a sip trace and figure out when the rtp starts arriving.
>>
>>
>> On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond
>> <fraserredmond at gmail.com> wrote:
>> > Sorry, yes, I am setting ignore_early_media=true in the first area. (Or
>> > are
>> > you saying that should be off? I forget now why I needed it on, but
>> > there
>> > was a reason I added it.)
>> >
>> > Yes, the bridge doesn't start until after the A-leg has answered.
>> >
>> > Thanks for the suggestion about nat/auto-changing port, I'll have a look
>> > into that - would that be in the cli output or in a sip trace? I've
>> > already
>> > looked and it's not appearing in the CLI output (with loglevel=debug),
>> > haven't looked in the sip trace yet.
>> >
>> > Cheers,
>> > Fraser
>> >
>> >
>> >
>> >
>> > On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> are you setting ignore_early_media=true in the first vars=values area?
>> >>
>> >> This looks like you could be calling one leg who is still not answered
>> >> and then bridging it to another dest. ?The bridge app will wait for
>> >> the first leg to answer before bridging.
>> >>
>> >> Also if you have any NAT anywhere, look for an "auto-changing port"
>> >> type message which can also be attributed to this due to a detection
>> >> period for incorrect ports.
>> >>
>> >>
>> >>
>> >> On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond
>> >> <fraserredmond at gmail.com> wrote:
>> >> > event_socket:
>> >> > api originate {vars=values}user/$fromExtn at Domain
>> >> > 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline
>> >> >
>> >> > then
>> >> >
>> >> > dialplan:
>> >> > <action application="set"
>> >> > data="effective_caller_id_number=+1800number"/>
>> >> > <action application="set" data="effective_caller_id_name="/>
>> >> > (set and/or export a bunch of other vars too)
>> >> > <action application="set"
>> >> >
>> >> >
>> >> > data="dial_string=sofia/gateway/gatewayname.com/00${destination_number}"/>
>> >> > <action application="bridge" data="${dial_string}"/>
>> >> >
>> >> > Cheers,
>> >> > Fraser
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale
>> >> > <anthony.minessale at gmail.com> wrote:
>> >> >>
>> >> >> how are you accomplishing that? by which technique?
>> >> >>
>> >> >> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond
>> >> >> <fraserredmond at gmail.com> wrote:
>> >> >> > The call is originated from Freeswitch (via CLI) to a softphone,
>> >> >> > then
>> >> >> > when
>> >> >> > that is connected it bridges out to the gateway.
>> >> >> >
>> >> >> > Cheers,
>> >> >> > Fraser
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale
>> >> >> > <anthony.minessale at gmail.com> wrote:
>> >> >> >>
>> >> >> >> Where is the other side of this call coming from?
>> >> >> >>
>> >> >> >> [ ( ? ) ] -> FS -> (PSTN via SIP)
>> >> >> >>
>> >> >> >> What goes in the empty space above?
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > 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
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From dyatsin at sangoma.com  Thu Oct 28 08:57:14 2010
From: dyatsin at sangoma.com (David Yat Sin)
Date: Thu, 28 Oct 2010 11:57:14 -0400
Subject: [Freeswitch-users] issue with freetdm
In-Reply-To: <AANLkTinYb5P7oSm1z5u=WJCqHXDjP7sbcojgr3T95nqc@mail.gmail.com>
References: <AANLkTimwOY+TcjacxuXXWJ=Da1k3MptOj=sOAhwvaLDR@mail.gmail.com>	<AANLkTin=isswKDbNWckxpYFo9SP1+VwGzCjd_KbQizWP@mail.gmail.com>
	<AANLkTinYb5P7oSm1z5u=WJCqHXDjP7sbcojgr3T95nqc@mail.gmail.com>
Message-ID: <4CC99D5A.3010107@sangoma.com>

Hi,
Can you check what type of L1 errors you are seeing:

from FS cli:
ftdm sangoma_isdn l1_stats wp1

Can you also check if wanpipemon is reporting errors:

wanpipemon -i w1g1 -c Ta

(repeat a couple times to see if any of the alarms are continuously 
toggling or the number of errors at the bottom keep incrementing)


*David Yat Sin, BEng*
Software Engineer
Sangoma Technologies
100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada 	

t.   +1 800 388 2475 x119
t.   +1 905 474 1990 x119
f.   +1 905 474 9223 	SANGOMA 
<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 

Products 
<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 
| Solutions 
<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 
| Events 
<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 
| Contact 
<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 
| Wiki 
<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 
| Facebook 
<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 
| Twitter 
<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> 



On 10/28/2010 12:04 AM, ovvenkat wrote:
> Hi moises,
>
> I am getting this error every 20 seconds.
>
> freeswitch at internal> 2010-10-28 09:32:55.366623 [ERR] 
> ftmod_sangoma_isdn_stack_rcv.c:883 sng_isdn->s1:L1 Rx Error
> 2010-10-28 09:32:55.366623 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:871 
> sng_isdn->s1: Resetting L1 link
> 2010-10-28 09:33:05.364851 [ERR] ftmod_sangoma_isdn_stack_rcv.c:883 
> sng_isdn->s1:L1 Rx Error
> 2010-10-28 09:33:05.364851 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:871 
> sng_isdn->s1: Resetting L1 link
> 2010-10-28 09:33:25.361287 [ERR] ftmod_sangoma_isdn_stack_rcv.c:883 
> sng_isdn->s1:L1 Rx Error
> 2010-10-28 09:33:25.361287 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:871 
> sng_isdn->s1: Resetting L1 link
> 2010-10-28 09:33:45.358910 [ERR] ftmod_sangoma_isdn_stack_rcv.c:883 
> sng_isdn->s1:L1 Rx Error
> 2010-10-28 09:33:45.358910 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:871 
> sng_isdn->s1: Resetting L1 link
> 2010-10-28 09:33:55.353722 [ERR] ftmod_sangoma_isdn_stack_rcv.c:883 
> sng_isdn->s1:L1 Rx Error
> 2010-10-28 09:33:55.353722 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:871 
> sng_isdn->s1: Resetting L1 link
>
>
>
> On Thu, Oct 28, 2010 at 4:33 AM, Moises Silva <moises.silva at gmail.com 
> <mailto:moises.silva at gmail.com>> wrote:
>
>     On Wed, Oct 27, 2010 at 9:23 AM, ovvenkat <ovvenkatesan at gmail.com
>     <mailto:ovvenkatesan at gmail.com>> wrote:
>     > Hello,
>     >
>     > I am using sangoma pri card A101 with freeSwitch.
>     > Today, I moved from openzap to freetdm.
>     > I have followed
>     http://wiki.sangoma.com/wanpipe-freeswitch-ftdm-installation
>     > page to install freeSwitch and freedtm.
>     >
>     > After installing  freeSwitch and Sangoma, I am getting following
>     error in
>     > fs_cli
>     >
>     > 2010-10-27 17:57:03.228457 [ERR] ftmod_sangoma_isdn_stack_rcv.c:883
>     > sng_isdn->s1:L1 Rx Error
>     > 2010-10-27 17:57:03.228457 [DEBUG]
>     ftmod_sangoma_isdn_stack_rcv.c:871
>     > sng_isdn->s1: Resetting L1 link
>     >
>     >
>     > What is these mean?
>
>     Do you get that constantly or eventually the link comes up?
>
>     Moises Silva
>     Senior Software Engineer
>     Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
>     L3R 9R6 Canada
>     t. 1 905 474 1990 x128 | e. moy at sangoma.com <mailto:moy at sangoma.com>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
>
> If you have come to help me, you are wasting your time.
> If you have come to because your liberation is bound up in mine, we 
> can work together.
>
>
> Regards
> Venkatesan OV.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From Nabble at slickdeals.endjunk.com  Thu Oct 28 09:09:22 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 28 Oct 2010 09:09:22 -0700 (PDT)
Subject: [Freeswitch-users] Running FS on Segate dockstar
In-Reply-To: <AANLkTi=8HFMCc_PSkMPGhC4sph0C-vU3Vb0J=J+5mTCM@mail.gmail.com>
References: <AANLkTi=8HFMCc_PSkMPGhC4sph0C-vU3Vb0J=J+5mTCM@mail.gmail.com>
Message-ID: <1288282162397-5683224.post@n2.nabble.com>


FS runs just fine on a Seagate DockStar. I have my FS v1.0.6 hosted on my
Seagate DockStar running on an OpenWRT Linux distro for almost three months
now. The only problem I have is sometime my FS crashes after a call has
ended. I haven't got any time to take a look at this yet.

W.r.t SkypeOpen, I believe it requires a Skype client running on the same
host. Those who are using SkypOpen module with FS are on an Intel/AMD
platform with an Intel/AMD Skype client. These days, it seems Skype has
supported many more different CPU platforms. If you can find a Skype client
for an ARM5 platform that can run on a DockStar along with FS, then there
may be a chance SkypOpen module will work. Until then, I don't know how to
make a SkypOpen module to work with FS on a Seagate DockStar.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Running-FS-on-Segate-dockstar-tp5682856p5683224.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From renjian at gmail.com  Thu Oct 28 09:13:09 2010
From: renjian at gmail.com (Jian Ren)
Date: Thu, 28 Oct 2010 12:13:09 -0400
Subject: [Freeswitch-users] Running FS on Segate dockstar
In-Reply-To: <1288282162397-5683224.post@n2.nabble.com>
References: <AANLkTi=8HFMCc_PSkMPGhC4sph0C-vU3Vb0J=J+5mTCM@mail.gmail.com>
	<1288282162397-5683224.post@n2.nabble.com>
Message-ID: <AANLkTimoqcbCS0Vvys6erUtp8Amn08TR2+Z50Ztg_6g+@mail.gmail.com>

I see. I installed debian on dockstar and always got errors when compiling
FS. It seems people only used OpenWRT.
Thanks!
Jian

On Thu, Oct 28, 2010 at 12:09 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
> FS runs just fine on a Seagate DockStar. I have my FS v1.0.6 hosted on my
> Seagate DockStar running on an OpenWRT Linux distro for almost three months
> now. The only problem I have is sometime my FS crashes after a call has
> ended. I haven't got any time to take a look at this yet.
>
> W.r.t SkypeOpen, I believe it requires a Skype client running on the same
> host. Those who are using SkypOpen module with FS are on an Intel/AMD
> platform with an Intel/AMD Skype client. These days, it seems Skype has
> supported many more different CPU platforms. If you can find a Skype client
> for an ARM5 platform that can run on a DockStar along with FS, then there
> may be a chance SkypOpen module will work. Until then, I don't know how to
> make a SkypOpen module to work with FS on a Seagate DockStar.
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Running-FS-on-Segate-dockstar-tp5682856p5683224.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Oct 28 10:11:49 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 28 Oct 2010 10:11:49 -0700
Subject: [Freeswitch-users] Memory problems
In-Reply-To: <AANLkTi=g9ySTLppUg-urCRThPea=+8G0=AwQkieYNwt4@mail.gmail.com>
References: <AANLkTina_J3XMLcrTweeKVw6AaVjJaCsvytFGSo+iPfv@mail.gmail.com>
	<AANLkTimHXVGdoBurEAw6NtFN8PnJDa649wib8_9qTTzC@mail.gmail.com>
	<AANLkTi=g9ySTLppUg-urCRThPea=+8G0=AwQkieYNwt4@mail.gmail.com>
Message-ID: <AANLkTimHMnYVPUF8gvSeh9NBFU8V4=aEC6qJSEWbNMiu@mail.gmail.com>

Is it consistently driving the CPU usage up after a restart? You might want
to do top -H to see if you can narrow down the exact process that is sucking
up all the resources. Also, make sure that you are not suffering a SIP
attack. You may want to consider setting up fail2ban:
http://wiki.freeswitch.org/wiki/Fail2ban

Make sure you update to the latest as soon as possible as Tony has fixed
some recent bugs.

-MC

On Wed, Oct 27, 2010 at 10:04 PM, Babstar <babstar99 at gmail.com> wrote:

>
>
> On Tue, Oct 26, 2010 at 6:23 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Look at the *.db files. Are any of them getting really large?
>> -MC
>>
>> Michael,
>                just had the latest failure, nothing unusually large with
> the *.db files:
> 16K     /usr/local/freeswitch/db/call_limit.db
> 148K    /usr/local/freeswitch/db/core.db
> 8.0K    /usr/local/freeswitch/db/fifo.db
> 76K     /usr/local/freeswitch/db/sofia_reg_external.db
> 504K    /usr/local/freeswitch/db/sofia_reg_internal.db
> 8.0K    /usr/local/freeswitch/db/sofia_reg_internal.db-journal
> 76K     /usr/local/freeswitch/db/sofia_reg_internal-ipv6.db
> 16K     /usr/local/freeswitch/db/voicemail_default.db
>
>
> When I restarted the daemon, the usage over a few minutes jump up to 24%,
>  only a reboot could  fix it.
>
> --
> Babstar
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Oct 28 10:19:45 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 28 Oct 2010 10:19:45 -0700
Subject: [Freeswitch-users] Problem with T.38 fax receive using ESL
In-Reply-To: <1288272464502-5682561.post@n2.nabble.com>
References: <1288272464502-5682561.post@n2.nabble.com>
Message-ID: <AANLkTinzujxqm7gT8k20+yeSKR+nYSRzjmdUSg_BGLZT@mail.gmail.com>

I don't see any indication in the docs that this syntax is doing what you
think it is:
{fax_enable_t38_request=true,fax_enable_t38=true,fax_header=testheader,fax_ident=testident}/tmp/faxes/myfax.tif

You could confirm this by removing the two "set" apps and adding those to
the "rxfax" call in the dialplan:
<action application="rxfax"
data="{fax_enable_t38_request=true,fax_enable_t38=true,fax_header=testheader,fax_ident=testident}/tmp/faxes/FAX-${uuid}.tif"/>

My guess is that the above will not work. I would recommend trying the
SendMsg to set the vars:

SendMsg 5716720c-e291-11df-939c-75e5ad3b2de7
call-command: execute
execute-app-name: set
execute-app-arg: fax_enable_t38_request=true

Do that for each var and then do the rxfax. Let us know what happens.
-MC

On Thu, Oct 28, 2010 at 6:27 AM, peely <freeswitch at peely.com> wrote:

>
> Hi,
>
> I'm having problems receiving faxes through the Event Socket API using T.38
> and SpanDSP. If I use the example dialplan entry:
>
>    <extension name="NTS TESTfax">
>        <condition field="destination_number" expression="^\+(testfax)">
>            <action application="answer" />
>            <action application="playback" data="silence_stream://2000"/>
>            <action application="set" data="fax_enable_t38_request=true"/>
>            <action application="set" data="fax_enable_t38=true"/>
>            <action application="rxfax" data="/tmp/faxes/FAX-${uuid}.tif"/>
>            <action application="hangup"/>
>        </condition>
>    </extension>
>
>
> Then everything is OK and a fax is received nicely whereas if I use an ESL
> command on an answered call like:
>
> SendMsg 5716720c-e291-11df-939c-75e5ad3b2de7
> call-command: execute
> execute-app-name: rxfax
> execute-app-arg:
>
> {fax_enable_t38_request=true,fax_enable_t38=true,fax_header=testheader,fax_ident=testident}/tmp/faxes/myfax.tif
>
> Then there's lots of T.38 mucking about but the fax terminates with a
> communications error. I've tried sleeping for 2 seconds, broadcasting
> silence_stream://2000 etc but to no avail.
>
> Has anyone successfully got T.38 fax receive working through the ESL, and
> if
> so, what chain of commands do you execute?
>
> I'm using a GIT trunk circa Oct 15th BTW.
>
>
>
> Many thanks,
>
>
>
> Neil.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Problem-with-T-38-fax-receive-using-ESL-tp5682561p5682561.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Oct 28 10:24:03 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 28 Oct 2010 10:24:03 -0700
Subject: [Freeswitch-users] Freeswitch and Digium
In-Reply-To: <4cc979a1.977adc0a.2bfe.ffff826eSMTPIN_ADDED@mx.google.com>
References: <4cc979a1.977adc0a.2bfe.ffff826eSMTPIN_ADDED@mx.google.com>
Message-ID: <AANLkTi=qZua8GyEgBR+a=4jDiT7_=XHrnR8xp_Vwk1fC@mail.gmail.com>

Grant,

I believe you might be looking in the wrong place for information. I would
start here:
http://wiki.freeswitch.org/wiki/FreeTDM

Note that if you are using a Digium-compatible card then you will need to
install the DAHDI drivers first and make sure that part is working. (If you
are able to run dahdi_tool and see your ports then you've got the DAHDI
stuff done and can move on to the FreeTDM setup.)

-MC

On Thu, Oct 28, 2010 at 6:03 AM, Grant Gray <grant at mrit.co.za> wrote:

>
>
> Hi All,
>
>
>
> I am trying to get a digium TDM410 card to work. I have looked at the
> following tutorials:
>
>
>
> http://wiki.sangoma.com/wanpipe-linux-drivers
>
> http://wiki.sangoma.com/wanpipe-api-freetdm-linux
>
>
>
> and a few others.
>
>
>
> The problem I am having is that this command does nothing:
>
>
>
> *export FREETDM_MOD_DIR=/usr/local/freetdm*
>
> * *
>
> it does not create the directory as required. Is this normal on Centos or
> is there another command?
>
>
>
> I also tried this command of one of the forum posts:
>
>
>
> make dahdi DAHDI_DIR=/usr/src/dahdi (
> http://wiki.sangoma.com/Topic-1273333936389)
>
>
>
> I poked around further and found that the export ?p command shows that the
> command is there:
>
>
>
> declare -x FREETDM_MOD_DIR="/usr/local/freetdm"
>
>
>
> So, I am wondering why its not working. I also cannot find a command like
> /sbin/export so i am lost.
>
>
>
> what I am really trying to achieve is the installation of the tdm410 card
> on freeswitch. Any tutorials would be really great.
>
>
>
> Regards,
>
>
>
> Gigg
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sos at sokhapkin.dyndns.org  Thu Oct 28 10:26:37 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 28 Oct 2010 13:26:37 -0400
Subject: [Freeswitch-users] Memory problems
In-Reply-To: <AANLkTimHMnYVPUF8gvSeh9NBFU8V4=aEC6qJSEWbNMiu@mail.gmail.com>
References: <AANLkTina_J3XMLcrTweeKVw6AaVjJaCsvytFGSo+iPfv@mail.gmail.com>
	<AANLkTi=g9ySTLppUg-urCRThPea=+8G0=AwQkieYNwt4@mail.gmail.com>
	<AANLkTimHMnYVPUF8gvSeh9NBFU8V4=aEC6qJSEWbNMiu@mail.gmail.com>
Message-ID: <201010281326.37641.sos@sokhapkin.dyndns.org>

AFAIK the original poster wrote about memory usage, but not CPU. I have 
exactly the same problem - after few days of run FS RSS grows to more than 
1GB, after FS restart RSS is about 50M. Version git-fe638ad 2010-10-13. Boxes 
running older FS version git-828960a 2010-09-25 have RSS in 60-100M range for 
weeks.

On Thursday 28 October 2010, Michael Collins wrote:
> Is it consistently driving the CPU usage up after a restart? You might want
> to do top -H to see if you can narrow down the exact process that is
>  sucking up all the resources. Also, make sure that you are not suffering a
>  SIP attack. You may want to consider setting up fail2ban:
> http://wiki.freeswitch.org/wiki/Fail2ban
> 
> Make sure you update to the latest as soon as possible as Tony has fixed
> some recent bugs.
> 
> -MC
> 
> On Wed, Oct 27, 2010 at 10:04 PM, Babstar <babstar99 at gmail.com> wrote:
> > On Tue, Oct 26, 2010 at 6:23 AM, Michael Collins 
<msc at freeswitch.org>wrote:
> >> Look at the *.db files. Are any of them getting really large?
> >> -MC
> >>
> >> Michael,
> >
> >                just had the latest failure, nothing unusually large with
> > the *.db files:
> > 16K     /usr/local/freeswitch/db/call_limit.db
> > 148K    /usr/local/freeswitch/db/core.db
> > 8.0K    /usr/local/freeswitch/db/fifo.db
> > 76K     /usr/local/freeswitch/db/sofia_reg_external.db
> > 504K    /usr/local/freeswitch/db/sofia_reg_internal.db
> > 8.0K    /usr/local/freeswitch/db/sofia_reg_internal.db-journal
> > 76K     /usr/local/freeswitch/db/sofia_reg_internal-ipv6.db
> > 16K     /usr/local/freeswitch/db/voicemail_default.db
> >
> >
> > When I restarted the daemon, the usage over a few minutes jump up to 24%,
> >  only a reboot could  fix it.
> >
> > --
> > Babstar
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 



From Nabble at slickdeals.endjunk.com  Thu Oct 28 10:53:11 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 28 Oct 2010 10:53:11 -0700 (PDT)
Subject: [Freeswitch-users] Running FS on Segate dockstar
In-Reply-To: <AANLkTimoqcbCS0Vvys6erUtp8Amn08TR2+Z50Ztg_6g+@mail.gmail.com>
References: <AANLkTi=8HFMCc_PSkMPGhC4sph0C-vU3Vb0J=J+5mTCM@mail.gmail.com>
	<1288282162397-5683224.post@n2.nabble.com>
	<AANLkTimoqcbCS0Vvys6erUtp8Amn08TR2+Z50Ztg_6g+@mail.gmail.com>
Message-ID: <1288288391296-5683644.post@n2.nabble.com>



Jian Ren wrote:
> I see. I installed debian on dockstar and always got errors when compiling
> FS.
Perhaps, you will need to fully install the Debian Linux distro for an ARM
platform on your DockStar in order to compile FS. Otherwise, you can
cross-compile FS for DockStar on you Linux desktop computer and then install
it on your DockStar along with the needed libraries used to cross-compile
the FS (unless the lbiraries are statically  cross-compiled into your FS
codes).


It seems people only used OpenWRT.
For my DockStar with OpenWRT, everything is cross-compiled on my AMD
PhenomII X3 desktop computer running on an OpenSuSE v11.3 Linux distro to
produce an OpenWRT firmware for a Seagate DockStar.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Running-FS-on-Segate-dockstar-tp5682856p5683644.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Thu Oct 28 12:11:18 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 28 Oct 2010 12:11:18 -0700
Subject: [Freeswitch-users] FS Conference Call Recordings
Message-ID: <AANLkTimpd1hbB0WwMgLbhn1YwZwb4QOfDjBPpBT5jV2C@mail.gmail.com>

Hello all!

I just wanted to let everyone know that I have started putting the
recordings for the conference calls on the main conference page:
http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls

I've got them in mp3, ogg, and wave formats.

Enjoy!

-MC
-------------- next part --------------
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From anthony.minessale at gmail.com  Thu Oct 28 13:38:18 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 28 Oct 2010 15:38:18 -0500
Subject: [Freeswitch-users] Memory problems
In-Reply-To: <201010281326.37641.sos@sokhapkin.dyndns.org>
References: <AANLkTina_J3XMLcrTweeKVw6AaVjJaCsvytFGSo+iPfv@mail.gmail.com>
	<AANLkTi=g9ySTLppUg-urCRThPea=+8G0=AwQkieYNwt4@mail.gmail.com>
	<AANLkTimHMnYVPUF8gvSeh9NBFU8V4=aEC6qJSEWbNMiu@mail.gmail.com>
	<201010281326.37641.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTi=pUif+Rzdd18B4vBAvq+YZRg6QFch6DRp24FhP@mail.gmail.com>

install valgrind

valgrind --tool=memcheck --log-file=vg.log --leak-check=full
--leak-resolution=high --show-reachable=yes .libs/lt-freeswitch -vg

make several minutes of typical calls
shutdown FS cleanly and send me vg.log

Also you could consider posting a jira in the future....





On Thu, Oct 28, 2010 at 12:26 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
> AFAIK the original poster wrote about memory usage, but not CPU. I have
> exactly the same problem - after few days of run FS RSS grows to more than
> 1GB, after FS restart RSS is about 50M. Version git-fe638ad 2010-10-13. Boxes
> running older FS version git-828960a 2010-09-25 have RSS in 60-100M range for
> weeks.
>
> On Thursday 28 October 2010, Michael Collins wrote:
>> Is it consistently driving the CPU usage up after a restart? You might want
>> to do top -H to see if you can narrow down the exact process that is
>> ?sucking up all the resources. Also, make sure that you are not suffering a
>> ?SIP attack. You may want to consider setting up fail2ban:
>> http://wiki.freeswitch.org/wiki/Fail2ban
>>
>> Make sure you update to the latest as soon as possible as Tony has fixed
>> some recent bugs.
>>
>> -MC
>>
>> On Wed, Oct 27, 2010 at 10:04 PM, Babstar <babstar99 at gmail.com> wrote:
>> > On Tue, Oct 26, 2010 at 6:23 AM, Michael Collins
> <msc at freeswitch.org>wrote:
>> >> Look at the *.db files. Are any of them getting really large?
>> >> -MC
>> >>
>> >> Michael,
>> >
>> > ? ? ? ? ? ? ? ?just had the latest failure, nothing unusually large with
>> > the *.db files:
>> > 16K ? ? /usr/local/freeswitch/db/call_limit.db
>> > 148K ? ?/usr/local/freeswitch/db/core.db
>> > 8.0K ? ?/usr/local/freeswitch/db/fifo.db
>> > 76K ? ? /usr/local/freeswitch/db/sofia_reg_external.db
>> > 504K ? ?/usr/local/freeswitch/db/sofia_reg_internal.db
>> > 8.0K ? ?/usr/local/freeswitch/db/sofia_reg_internal.db-journal
>> > 76K ? ? /usr/local/freeswitch/db/sofia_reg_internal-ipv6.db
>> > 16K ? ? /usr/local/freeswitch/db/voicemail_default.db
>> >
>> >
>> > When I restarted the daemon, the usage over a few minutes jump up to 24%,
>> > ?only a reboot could ?fix it.
>> >
>> > --
>> > Babstar
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From anthony.minessale at gmail.com  Thu Oct 28 13:39:04 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 28 Oct 2010 15:39:04 -0500
Subject: [Freeswitch-users] Memory problems
In-Reply-To: <AANLkTi=pUif+Rzdd18B4vBAvq+YZRg6QFch6DRp24FhP@mail.gmail.com>
References: <AANLkTina_J3XMLcrTweeKVw6AaVjJaCsvytFGSo+iPfv@mail.gmail.com>
	<AANLkTi=g9ySTLppUg-urCRThPea=+8G0=AwQkieYNwt4@mail.gmail.com>
	<AANLkTimHMnYVPUF8gvSeh9NBFU8V4=aEC6qJSEWbNMiu@mail.gmail.com>
	<201010281326.37641.sos@sokhapkin.dyndns.org>
	<AANLkTi=pUif+Rzdd18B4vBAvq+YZRg6QFch6DRp24FhP@mail.gmail.com>
Message-ID: <AANLkTi=xspoih5w6toHnRJT6hbGXtSfxQ2cvNzGWCy2Y@mail.gmail.com>

oh, and update to current first

git pull
make update-clean
make modwipe
make install


On Thu, Oct 28, 2010 at 3:38 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> install valgrind
>
> valgrind --tool=memcheck --log-file=vg.log --leak-check=full
> --leak-resolution=high --show-reachable=yes .libs/lt-freeswitch -vg
>
> make several minutes of typical calls
> shutdown FS cleanly and send me vg.log
>
> Also you could consider posting a jira in the future....
>
>
>
>
>
> On Thu, Oct 28, 2010 at 12:26 PM, Sergey Okhapkin
> <sos at sokhapkin.dyndns.org> wrote:
>> AFAIK the original poster wrote about memory usage, but not CPU. I have
>> exactly the same problem - after few days of run FS RSS grows to more than
>> 1GB, after FS restart RSS is about 50M. Version git-fe638ad 2010-10-13. Boxes
>> running older FS version git-828960a 2010-09-25 have RSS in 60-100M range for
>> weeks.
>>
>> On Thursday 28 October 2010, Michael Collins wrote:
>>> Is it consistently driving the CPU usage up after a restart? You might want
>>> to do top -H to see if you can narrow down the exact process that is
>>> ?sucking up all the resources. Also, make sure that you are not suffering a
>>> ?SIP attack. You may want to consider setting up fail2ban:
>>> http://wiki.freeswitch.org/wiki/Fail2ban
>>>
>>> Make sure you update to the latest as soon as possible as Tony has fixed
>>> some recent bugs.
>>>
>>> -MC
>>>
>>> On Wed, Oct 27, 2010 at 10:04 PM, Babstar <babstar99 at gmail.com> wrote:
>>> > On Tue, Oct 26, 2010 at 6:23 AM, Michael Collins
>> <msc at freeswitch.org>wrote:
>>> >> Look at the *.db files. Are any of them getting really large?
>>> >> -MC
>>> >>
>>> >> Michael,
>>> >
>>> > ? ? ? ? ? ? ? ?just had the latest failure, nothing unusually large with
>>> > the *.db files:
>>> > 16K ? ? /usr/local/freeswitch/db/call_limit.db
>>> > 148K ? ?/usr/local/freeswitch/db/core.db
>>> > 8.0K ? ?/usr/local/freeswitch/db/fifo.db
>>> > 76K ? ? /usr/local/freeswitch/db/sofia_reg_external.db
>>> > 504K ? ?/usr/local/freeswitch/db/sofia_reg_internal.db
>>> > 8.0K ? ?/usr/local/freeswitch/db/sofia_reg_internal.db-journal
>>> > 76K ? ? /usr/local/freeswitch/db/sofia_reg_internal-ipv6.db
>>> > 16K ? ? /usr/local/freeswitch/db/voicemail_default.db
>>> >
>>> >
>>> > When I restarted the daemon, the usage over a few minutes jump up to 24%,
>>> > ?only a reboot could ?fix it.
>>> >
>>> > --
>>> > Babstar
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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
>



-- 
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


From sos at sokhapkin.dyndns.org  Thu Oct 28 13:52:31 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 28 Oct 2010 16:52:31 -0400
Subject: [Freeswitch-users] Memory problems
In-Reply-To: <AANLkTi=pUif+Rzdd18B4vBAvq+YZRg6QFch6DRp24FhP@mail.gmail.com>
References: <AANLkTina_J3XMLcrTweeKVw6AaVjJaCsvytFGSo+iPfv@mail.gmail.com>
	<201010281326.37641.sos@sokhapkin.dyndns.org>
	<AANLkTi=pUif+Rzdd18B4vBAvq+YZRg6QFch6DRp24FhP@mail.gmail.com>
Message-ID: <201010281652.31959.sos@sokhapkin.dyndns.org>

I can't run valgrind on the production server (you know why)...

On Thursday 28 October 2010, Anthony Minessale wrote:
> install valgrind
> 
> valgrind --tool=memcheck --log-file=vg.log --leak-check=full
> --leak-resolution=high --show-reachable=yes .libs/lt-freeswitch -vg
> 
> make several minutes of typical calls
> shutdown FS cleanly and send me vg.log
> 
> Also you could consider posting a jira in the future....
> 
> 
> 
> 
> 
> On Thu, Oct 28, 2010 at 12:26 PM, Sergey Okhapkin
> 
> <sos at sokhapkin.dyndns.org> wrote:
> > AFAIK the original poster wrote about memory usage, but not CPU. I have
> > exactly the same problem - after few days of run FS RSS grows to more
> > than 1GB, after FS restart RSS is about 50M. Version git-fe638ad
> > 2010-10-13. Boxes running older FS version git-828960a 2010-09-25 have
> > RSS in 60-100M range for weeks.
> >
> > On Thursday 28 October 2010, Michael Collins wrote:
> >> Is it consistently driving the CPU usage up after a restart? You might
> >> want to do top -H to see if you can narrow down the exact process that
> >> is sucking up all the resources. Also, make sure that you are not
> >> suffering a SIP attack. You may want to consider setting up fail2ban:
> >> http://wiki.freeswitch.org/wiki/Fail2ban
> >>
> >> Make sure you update to the latest as soon as possible as Tony has fixed
> >> some recent bugs.
> >>
> >> -MC
> >>
> >> On Wed, Oct 27, 2010 at 10:04 PM, Babstar <babstar99 at gmail.com> wrote:
> >> > On Tue, Oct 26, 2010 at 6:23 AM, Michael Collins
> >
> > <msc at freeswitch.org>wrote:
> >> >> Look at the *.db files. Are any of them getting really large?
> >> >> -MC
> >> >>
> >> >> Michael,
> >> >
> >> >                just had the latest failure, nothing unusually large
> >> > with the *.db files:
> >> > 16K     /usr/local/freeswitch/db/call_limit.db
> >> > 148K    /usr/local/freeswitch/db/core.db
> >> > 8.0K    /usr/local/freeswitch/db/fifo.db
> >> > 76K     /usr/local/freeswitch/db/sofia_reg_external.db
> >> > 504K    /usr/local/freeswitch/db/sofia_reg_internal.db
> >> > 8.0K    /usr/local/freeswitch/db/sofia_reg_internal.db-journal
> >> > 76K     /usr/local/freeswitch/db/sofia_reg_internal-ipv6.db
> >> > 16K     /usr/local/freeswitch/db/voicemail_default.db
> >> >
> >> >
> >> > When I restarted the daemon, the usage over a few minutes jump up to
> >> > 24%, only a reboot could  fix it.
> >> >
> >> > --
> >> > Babstar
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> >> >rs http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 



From sos at sokhapkin.dyndns.org  Thu Oct 28 14:01:12 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 28 Oct 2010 17:01:12 -0400
Subject: [Freeswitch-users] Memory problems
In-Reply-To: <201010281652.31959.sos@sokhapkin.dyndns.org>
References: <AANLkTina_J3XMLcrTweeKVw6AaVjJaCsvytFGSo+iPfv@mail.gmail.com>
	<AANLkTi=pUif+Rzdd18B4vBAvq+YZRg6QFch6DRp24FhP@mail.gmail.com>
	<201010281652.31959.sos@sokhapkin.dyndns.org>
Message-ID: <201010281701.12754.sos@sokhapkin.dyndns.org>

Many month ago we had similar discussion about memory leak in mod_nibblebill. 
Valgring didn't find the cause of the leak.

On Thursday 28 October 2010, Sergey Okhapkin wrote:
> I can't run valgrind on the production server (you know why)...
> 
> On Thursday 28 October 2010, Anthony Minessale wrote:
> > install valgrind
> >
> > valgrind --tool=memcheck --log-file=vg.log --leak-check=full
> > --leak-resolution=high --show-reachable=yes .libs/lt-freeswitch -vg
> >
> > make several minutes of typical calls
> > shutdown FS cleanly and send me vg.log
> >
> > Also you could consider posting a jira in the future....
> >
> >
> >
> >
> >
> > On Thu, Oct 28, 2010 at 12:26 PM, Sergey Okhapkin
> >
> > <sos at sokhapkin.dyndns.org> wrote:
> > > AFAIK the original poster wrote about memory usage, but not CPU. I have
> > > exactly the same problem - after few days of run FS RSS grows to more
> > > than 1GB, after FS restart RSS is about 50M. Version git-fe638ad
> > > 2010-10-13. Boxes running older FS version git-828960a 2010-09-25 have
> > > RSS in 60-100M range for weeks.
> > >
> > > On Thursday 28 October 2010, Michael Collins wrote:
> > >> Is it consistently driving the CPU usage up after a restart? You might
> > >> want to do top -H to see if you can narrow down the exact process that
> > >> is sucking up all the resources. Also, make sure that you are not
> > >> suffering a SIP attack. You may want to consider setting up fail2ban:
> > >> http://wiki.freeswitch.org/wiki/Fail2ban
> > >>
> > >> Make sure you update to the latest as soon as possible as Tony has
> > >> fixed some recent bugs.
> > >>
> > >> -MC
> > >>
> > >> On Wed, Oct 27, 2010 at 10:04 PM, Babstar <babstar99 at gmail.com> wrote:
> > >> > On Tue, Oct 26, 2010 at 6:23 AM, Michael Collins
> > >
> > > <msc at freeswitch.org>wrote:
> > >> >> Look at the *.db files. Are any of them getting really large?
> > >> >> -MC
> > >> >>
> > >> >> Michael,
> > >> >
> > >> >                just had the latest failure, nothing unusually large
> > >> > with the *.db files:
> > >> > 16K     /usr/local/freeswitch/db/call_limit.db
> > >> > 148K    /usr/local/freeswitch/db/core.db
> > >> > 8.0K    /usr/local/freeswitch/db/fifo.db
> > >> > 76K     /usr/local/freeswitch/db/sofia_reg_external.db
> > >> > 504K    /usr/local/freeswitch/db/sofia_reg_internal.db
> > >> > 8.0K    /usr/local/freeswitch/db/sofia_reg_internal.db-journal
> > >> > 76K     /usr/local/freeswitch/db/sofia_reg_internal-ipv6.db
> > >> > 16K     /usr/local/freeswitch/db/voicemail_default.db
> > >> >
> > >> >
> > >> > When I restarted the daemon, the usage over a few minutes jump up to
> > >> > 24%, only a reboot could  fix it.
> > >> >
> > >> > --
> > >> > Babstar
> > >> >
> > >> > _______________________________________________
> > >> > FreeSWITCH-users mailing list
> > >> > FreeSWITCH-users at lists.freeswitch.org
> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u
> > >> >se rs http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >s http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From anthony.minessale at gmail.com  Thu Oct 28 14:12:25 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 28 Oct 2010 16:12:25 -0500
Subject: [Freeswitch-users] FS incoming calls Queue ACD implement
In-Reply-To: <AANLkTinuy+DhD0mZGj0BWOMx2Ha8sTXLetnQWLNK9PX6@mail.gmail.com>
References: <1288212978229-5680281.post@n2.nabble.com>
	<AANLkTinuCX6xrzXHEg1pA43_TXnauSr4EDpnQhmRdPe-@mail.gmail.com>
	<AANLkTinuy+DhD0mZGj0BWOMx2Ha8sTXLetnQWLNK9PX6@mail.gmail.com>
Message-ID: <AANLkTim+tWdjghr3AhP-eFX9nu4rmk291y2y0AxPHAB4@mail.gmail.com>

btw, i rewrote hundereds of lines of code in mod_fifo just to get that
callerid thing to work which it now does.


On Wed, Oct 27, 2010 at 9:58 PM, Johnson <johnsonw at eallway.com> wrote:
> Hi MC,
>
> Thanks for your feedback quickly. Actually i have tested mod_fifo and met
> the callerid can't transfer to agent's phone before the agent pickup.
> Traditional phones almost don't support new features like getting sip
> "UPDATE MESSAGE". As alternatively, i try to use Moc' mod_callcenter, but
> unfortunitely, there is no mod_callcenter package in FS1.06, even i put in
> dialplan
> <action application="callcenter" data="support at default"/>, it show log:
> ?[ERR] switch_core_session.c:1731 Invalid Application callcenter
> Is it meaning that the mod_callcenter is still going on coding? Any further
> more helps would be much appreciated.
>
> Thanks,
>
> Johnson
>
>
> On Thu, Oct 28, 2010 at 7:37 AM, mercutioviz [via freeswitch-users] <[hidden
> email]> wrote:
>>
>> I'm afraid this question is a bit broad. I notice that you are trying to
>> avoid the use of mod_fifo - is there a reason for that? Alternatively you
>> can use Moc's newer ACD system: mod_callcenter. It is more of a traditional
>> ACD than mod_fifo. (Both get inbound calls sent to agents, they just differ
>> in the methodology used to accomplish that task.)
>>
>> I've found the documentation on mod_callcenter to be refreshingly detailed
>> and complete:
>> http://wiki.freeswitch.org/wiki/Mod_callcenter
>>
>> -MC
>>
>> On Wed, Oct 27, 2010 at 1:56 PM, Johnson <[hidden email]> wrote:
>>>
>>> Hi FS Group,
>>> I am a new vanilla. Just want to realize Queue ACD function in
>>> Freeswitch.
>>> Is any one know how to configure or create the script to process incoming
>>> calls put into defined Queues without using mod_FIFO. Any additional
>>> prompting is very appreciated.
>>>
>>> Johnson
>>> --
>>> View this message in context:
>>> http://freeswitch-users.2379917.n2.nabble.com/FS-incoming-calls-Queue-ACD-implement-tp5680281p5680281.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> [hidden email]
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> [hidden email]
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> ________________________________
>> View message @
>> http://freeswitch-users.2379917.n2.nabble.com/FS-incoming-calls-Queue-ACD-implement-tp5680281p5680777.html
>> To unsubscribe from FS incoming calls Queue ACD implement, click here.
>
>
> ________________________________
> View this message in context: Re: FS incoming calls Queue ACD implement
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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
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FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From msc at freeswitch.org  Thu Oct 28 16:37:44 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 28 Oct 2010 16:37:44 -0700
Subject: [Freeswitch-users] Language files - French, German, Spanish, etc.
Message-ID: <AANLkTim4Wc+hV_aCufsa1VkiBheCEWJGC16OhzRQ7fi8@mail.gmail.com>

Hello all!

I have heard about several of our intrepid community members creating sound
sets in various languages. If you have created a sound set in any language
other than US English please contact me off list. I want to help get this
sound prompts organized so that the global community can use them.

Thanks!
-Michael
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From willbelair at yahoo.com  Thu Oct 28 14:14:26 2010
From: willbelair at yahoo.com (Will Smith)
Date: Thu, 28 Oct 2010 14:14:26 -0700 (PDT)
Subject: [Freeswitch-users] Mod Event Socket
Message-ID: <12229.75246.qm@web55701.mail.re3.yahoo.com>

Hi,I am new to FS, and trying to get the mod event socket installed and running. I have FS running, with SIP account, can dial in/out via gateway.Now I want to dial in, FS will send some info to client browser, here is one question, does this work over the internet, or just local ? Info sent could be the uuid, so that client browser could decide to bridge the call, send to IVR or transfer ...Please give me some guide line how to set this up.I added to dialplan/default.xml----- <extension name="socketTest">
      <condition field="destination_number" expression="^444$">
        <action application="socket" data="127.0.0.1:9090 async full"/>
      </condition>
 </extension>-----
got the php sample file: <?php

/**
 * Based loosely on the NET_Server code in PEAR.
 * This is only an example - considerable additional work is needed
 * Specifically, the code in the handleConnection method should be 
 * handled in a subclass
 *
 * 
 */

class Message
{
  var $properties = array();
  var $content = null;
}

class EventSocketListener
{

  var $host;
  var $port;
  var $sock;
  var $is_parent = true;
  var $clientInfo;
  var $clientFD;

  var $connectionContext = array();


  function &create($port, $host = "localhost")
      {
	$esl = new EventSocketListener;
	$esl->port = $port;
	$esl->host = $host;
        if (!function_exists('socket_create')) {
            return PEAR::raiseError('Sockets extension not available.');
        }
	return $esl;
      }

  function start()
  {
	if (($this->sock = socket_create(AF_INET, SOCK_STREAM, SOL_TCP)) === false) {
	  echo "socket_create() failed: reason: " . socket_strerror(socket_last_error()) . "\n";
	}

	if (!socket_set_option($this->sock, SOL_SOCKET, SO_REUSEADDR, 1)) {
	  echo 'Unable to set option on socket: '. socket_strerror(socket_last_error()) . PHP_EOL;
	}

	if (socket_bind($this->sock, $this->host, $this->port) === false) {
	  echo "socket_bind() failed: reason: " . socket_strerror(socket_last_error($this->sock)) . "\n";
	}

	if (socket_listen($this->sock, 5) === false) {
	  echo "socket_listen() failed: reason: " . socket_strerror(socket_last_error($this->sock)) . "\n";
	}

        // Dear children, please do not become zombies
        pcntl_signal(SIGCHLD, SIG_IGN);
        
        // wait for incmoning connections
        while (true)
        {
            // new connection
            if(($fd = socket_accept($this->sock)))
            {
	      $pid = pcntl_fork();
	      if($pid == -1) {
		  return  PEAR::raiseError('Could not fork child process.');
                }
                // This is the child => handle the request
                elseif($pid == 0) {
                    // this is not the parent
                    $this->_isParent = false;
                    // store the new file descriptor
                    $this->clientFD = $fd;

                    $peer_host    =    "";
                    $peer_port    =    "";
                    socket_getpeername($this->clientFD, $peer_host, $peer_port);
                    $this->clientInfo    =    array(
                                                      "host"        =>    $peer_host,
                                                      "port"        =>    $peer_port,
                                                      "connectOn"   =>    time()
                                                   );
		    $this->handleConnection();
		    socket_shutdown($this->clientFD, 2);
		    socket_close($this->clientFD);
		}
		else /* Parent does nothing */
		  {
		  }
	    }
	}
  }

  function handleConnection()
  {
    $fd = $this->clientFD;
    //first, read headers & setup a state for this connection
    $line = "";
    socket_write($fd, "CONNECT\n\n");
    do 
      {
	$line = socket_read($fd, 2048, PHP_NORMAL_READ);
	if (trim($line) == "")
	  break;
	//we got a header, we need to add it to the 
	list($key, $value) = explode(":", $line);
	$key = trim($key);
	$value = trim(urldecode($value));
	$this->connectionContext[$key] = $value;
      }
    while ($line != "");
    
    //    print_r($this->connectionContext);
    $this->callConnected();

    exit();
    
  }

  function processMessages($returnOnReply = false)
  {
    $fd = $this->clientFD;
    $result = new Message();
    $props = array();
    while (true)
      {
	do 
	  {
	    $line = @socket_read($fd, 2048, PHP_NORMAL_READ);
	    if (socket_last_error($fd) == 104)
	      return null;
	    if ($line == null || $line == FALSE || trim($line) == "")
	      break;
	    //we got a header, we need to add it to the message
	    list($key, $value) = explode(":", $line);
	    $key = trim($key);
	    $value = trim(urldecode($value));
	    $props[$key] = $value;
	  }
	while ($line != "");
	$result->properties = $props;

	if (isset($props['Content-Length']))
	  {
	    $length = $props['Content-Length'];
	    print("Reading content - $length\n");
	    $data = socket_read($fd, $length);
	    $result->content = $data;
	  }
	if (isset($props['Content-Type']))
	  {
	    $type = $props['Content-Type'];
	    if ($returnOnReply && 
		($type == "command/reply" || $type == "api/response"))
	      {
		return $result;
	      }
	    else if ($type == "text/event-plain") //only plain events for now
	      {
		$this->handleEvent($result);
	      }
	  }
	else
	  {
	    print("UNKNOWN MESSAGE: \n");
	    print_r($result);
	  }
      }    
  }
    


  function invokeCommand($command)
  {
    //Send the command
    print("Invoking: $command\n");
    $this->sendCommand($command);
    // Wait for the response
    $result = $this->processMessages(true);
    return $result;
  }

  function sendCommand($command)
  {
    $fd = $this->clientFD;
    socket_write($fd, trim($command) . "\n\n"); 
  }

/*-----------------------------------------------------*/
  /* Abstract Methods - should move to subclass*/

  function callConnected()
  {
    print_r($this->connectionContext);
    print("----------------\n");
    $result = $this->invokeCommand("log DEBUG");
    print_r($result);
    $result = $this->invokeCommand("event plain ALL");
    print_r($result);

    $this->processMessages(false);
    print("DONE PROCESSING MESSAGES");
    print_r($this->connectionContext);
  }

  function handleCommandResponse($response)
  {
    print("Recieved Unhandled Response:\n");
    print_r($response);
  }

  function handleEvent($event)
  {
    print("Recieved Unhandled Event:\n");
    print_r($event);
  }
}

  

  
// create a server that forks new processes
$server  = &EventSocketListener::create(9090);

// start the server
$server->start();
?>--------( this is the original file, not perfect sample) I tried to run this, and got error with auth.Also, I modify the even_socket_conf.xml  in autoload_configs/  change listen-ip to  0.0.0.0 , port = 9090 , disable password
What did I miss? Thankyou for your helpWill


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From msc at freeswitch.org  Thu Oct 28 17:57:49 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 28 Oct 2010 17:57:49 -0700
Subject: [Freeswitch-users] Mod Event Socket
In-Reply-To: <12229.75246.qm@web55701.mail.re3.yahoo.com>
References: <12229.75246.qm@web55701.mail.re3.yahoo.com>
Message-ID: <AANLkTi=6HgNczOBs110tX-ofAO50FbapWZ1P3ascMy0h@mail.gmail.com>

Be sure to learn the difference between inbound and outbound event socket.
In your case you are doing outbound event socket. The dialplan calls the
socket app which does an outbound socket connection to port 9090 on the
localhost. Make sure that your app is listening on port 9090.

The event_socket.conf.xml file is for configuring inbound socket
connections, e.g. when you have a script that attempts to connect to FS
where FS is listening on port 8021. So change your event_socket.conf.xml
file back to the default and retry.

Report back here if you have trouble. Also, buy or borrow the FreeSWITCH
book and check out chapter 9. Lots of good stuff there on how the even
system works.

-MC

On Thu, Oct 28, 2010 at 2:14 PM, Will Smith <willbelair at yahoo.com> wrote:

> Hi,
> I am new to FS, and trying to get the mod event socket installed and
> running. I have FS running, with SIP account, can dial in/out via gateway.
> Now I want to dial in, FS will send some info to client browser, here is
> one question, does this work over the internet, or just local ? Info sent
> could be the uuid, so that client browser could decide to bridge the call,
> send to IVR or transfer ...
> Please give me some guide line how to set this up.
> I added to dialplan/default.xml
> -----
>
>  <extension name="socketTest">
>       <condition field="destination_number" expression="^444$">
>         <action application="socket" data="127.0.0.1:9090 async full"/>
>       </condition>
>  </extension>
>
> -----
>
>
> got the php sample file:
>
> <?php
>
> /**
>  * Based loosely on the NET_Server code in PEAR.
>  * This is only an example - considerable additional work is needed
>  * Specifically, the code in the handleConnection method should be
>  * handled in a subclass
>  *
>  *
>  */
>
> class Message
> {
>   var $properties = array();
>   var $content = null;
> }
>
> class EventSocketListener
> {
>
>   var $host;
>   var $port;
>   var $sock;
>   var $is_parent = true;
>   var $clientInfo;
>   var $clientFD;
>
>   var $connectionContext = array();
>
>
>   function &create($port, $host = "localhost")
>       {
> 	$esl = new EventSocketListener;
> 	$esl->port = $port;
> 	$esl->host = $host;
>         if (!function_exists('socket_create')) {
>             return PEAR::raiseError('Sockets extension not available.');
>         }
> 	return $esl;
>       }
>
>   function start()
>   {
> 	if (($this->sock = socket_create(AF_INET, SOCK_STREAM, SOL_TCP)) === false) {
> 	  echo "socket_create() failed: reason: " . socket_strerror(socket_last_error()) . "\n";
> 	}
>
> 	if (!socket_set_option($this->sock, SOL_SOCKET, SO_REUSEADDR, 1)) {
> 	  echo 'Unable to set option on socket: '. socket_strerror(socket_last_error()) . PHP_EOL;
> 	}
>
> 	if (socket_bind($this->sock, $this->host, $this->port) === false) {
> 	  echo "socket_bind() failed: reason: " . socket_strerror(socket_last_error($this->sock)) . "\n";
> 	}
>
> 	if (socket_listen($this->sock, 5) === false) {
> 	  echo "socket_listen() failed: reason: " . socket_strerror(socket_last_error($this->sock)) . "\n";
> 	}
>
>         // Dear children, please do not become zombies
>         pcntl_signal(SIGCHLD, SIG_IGN);
>
>         // wait for incmoning connections
>         while (true)
>         {
>             // new connection
>             if(($fd = socket_accept($this->sock)))
>             {
> 	      $pid = pcntl_fork();
> 	      if($pid == -1) {
> 		  return  PEAR::raiseError('Could not fork child process.');
>                 }
>                 // This is the child => handle the request
>                 elseif($pid == 0) {
>                     // this is not the parent
>                     $this->_isParent = false;
>                     // store the new file descriptor
>                     $this->clientFD = $fd;
>
>                     $peer_host    =    "";
>                     $peer_port    =    "";
>                     socket_getpeername($this->clientFD, $peer_host, $peer_port);
>                     $this->clientInfo    =    array(
>                                                       "host"        =>    $peer_host,
>                                                       "port"        =>    $peer_port,
>                                                       "connectOn"   =>    time()
>                                                    );
> 		    $this->handleConnection();
> 		    socket_shutdown($this->clientFD, 2);
> 		    socket_close($this->clientFD);
> 		}
> 		else /* Parent does nothing */
> 		  {
> 		  }
> 	    }
> 	}
>   }
>
>   function handleConnection()
>   {
>     $fd = $this->clientFD;
>     //first, read headers & setup a state for this connection
>     $line = "";
>     socket_write($fd, "CONNECT\n\n");
>     do
>       {
> 	$line = socket_read($fd, 2048, PHP_NORMAL_READ);
> 	if (trim($line) == "")
> 	  break;
> 	//we got a header, we need to add it to the
> 	list($key, $value) = explode(":", $line);
> 	$key = trim($key);
> 	$value = trim(urldecode($value));
> 	$this->connectionContext[$key] = $value;
>       }
>     while ($line != "");
>
>     //    print_r($this->connectionContext);
>     $this->callConnected();
>
>     exit();
>
>   }
>
>   function processMessages($returnOnReply = false)
>   {
>     $fd = $this->clientFD;
>     $result = new Message();
>     $props = array();
>     while (true)
>       {
> 	do
> 	  {
> 	    $line = @socket_read($fd, 2048, PHP_NORMAL_READ);
> 	    if (socket_last_error($fd) == 104)
> 	      return null;
> 	    if ($line == null || $line == FALSE || trim($line) == "")
> 	      break;
> 	    //we got a header, we need to add it to the message
> 	    list($key, $value) = explode(":", $line);
> 	    $key = trim($key);
> 	    $value = trim(urldecode($value));
> 	    $props[$key] = $value;
> 	  }
> 	while ($line != "");
> 	$result->properties = $props;
>
> 	if (isset($props['Content-Length']))
> 	  {
> 	    $length = $props['Content-Length'];
> 	    print("Reading content - $length\n");
> 	    $data = socket_read($fd, $length);
> 	    $result->content = $data;
> 	  }
> 	if (isset($props['Content-Type']))
> 	  {
> 	    $type = $props['Content-Type'];
> 	    if ($returnOnReply &&
> 		($type == "command/reply" || $type == "api/response"))
> 	      {
> 		return $result;
> 	      }
> 	    else if ($type == "text/event-plain") //only plain events for now
> 	      {
> 		$this->handleEvent($result);
> 	      }
> 	  }
> 	else
> 	  {
> 	    print("UNKNOWN MESSAGE: \n");
> 	    print_r($result);
> 	  }
>       }
>   }
>
>
>
>   function invokeCommand($command)
>   {
>     //Send the command
>     print("Invoking: $command\n");
>     $this->sendCommand($command);
>     // Wait for the response
>     $result = $this->processMessages(true);
>     return $result;
>   }
>
>   function sendCommand($command)
>   {
>     $fd = $this->clientFD;
>     socket_write($fd, trim($command) . "\n\n");
>   }
>
> /*-----------------------------------------------------*/
>   /* Abstract Methods - should move to subclass*/
>
>   function callConnected()
>   {
>     print_r($this->connectionContext);
>     print("----------------\n");
>     $result = $this->invokeCommand("log DEBUG");
>     print_r($result);
>     $result = $this->invokeCommand("event plain ALL");
>     print_r($result);
>
>     $this->processMessages(false);
>     print("DONE PROCESSING MESSAGES");
>     print_r($this->connectionContext);
>   }
>
>   function handleCommandResponse($response)
>   {
>     print("Recieved Unhandled Response:\n");
>     print_r($response);
>   }
>
>   function handleEvent($event)
>   {
>     print("Recieved Unhandled Event:\n");
>     print_r($event);
>   }
> }
>
>
>
>
> // create a server that forks new processes
> $server  = &EventSocketListener::create(9090);
>
> // start the server
> $server->start();
> ?>
>
> --------
>
> ( this is the original file, not perfect sample) I tried to run this, and got error with auth.
>
> Also, I modify the even_socket_conf.xml  in autoload_configs/
>
> change listen-ip to  0.0.0.0 , port = 9090 , disable password
>
>
> What did I miss?
>
> Thankyou for your help
>
> Will
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bwibowo at gmail.com  Thu Oct 28 21:19:51 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Fri, 29 Oct 2010 11:19:51 +0700
Subject: [Freeswitch-users] Problems making a receiveing calls with
	mod_jingling
In-Reply-To: <AANLkTi=ZmQo6QHfEFoHPaNd_Tx2HrM+aZZAgfh569scz@mail.gmail.com>
References: <201010131244.41086.justlikeef@gmail.com>
	<AANLkTim60-09hvL1=SBrb8R86_1qqPXE2-+42ZSxZ9Ja@mail.gmail.com>
	<201010131807.46035.justlikeef@gmail.com>
	<AANLkTi=K65YxFQAXudWiPTSAUSH2Xzjd4ks5_coSJ5K_@mail.gmail.com>
	<AANLkTi=Exfm08ZMfm7Eur8GQBViX6KuPt_ntRYJ48qW_@mail.gmail.com>
	<AANLkTi=ZmQo6QHfEFoHPaNd_Tx2HrM+aZZAgfh569scz@mail.gmail.com>
Message-ID: <AANLkTin_0+04eA98dXTNZztLZfuCa8SFubaJr921j8JH@mail.gmail.com>

thx fixed after upgrade.
client.xml already configured as wiki said.
for dialplan i put
in usr/local/freeswitch/conf/dialplan/default/02_gtalk.xml containing
<include>
    <extension name="gtalk-out">
      <condition field="destination_number" expression="^1(1\d{10})$">
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="bridge" data="dingaling/gtalk/+$
1 at voice.google.com"/>
       </condition>
   </extension>
</include>

and got this error
2010-10-29 12:17:27.332766 [DEBUG] switch_core_state_machine.c:53
sofia/internal/1000 at 202.122.99.99 Standard REPORTING, cause:
RECOVERY_ON_TIMER_EXPIRE
2010-10-29 12:17:27.332766 [DEBUG] switch_core_state_machine.c:599
(sofia/internal/1000 at 202.122.99.99) State REPORTING going to sleep
2010-10-29 12:17:27.332766 [DEBUG] switch_core_state_machine.c:331
(sofia/internal/1000 at 202.122.99.99) State Change CS_REPORTING -> CS_DESTROY
2010-10-29 12:17:27.332766 [DEBUG] switch_core_session.c:1057 Send signal
sofia/internal/1000 at 202.122.99.99 [BREAK]
2010-10-29 12:17:27.332766 [DEBUG] switch_core_session.c:1224 Session 5
(sofia/internal/1000 at 202.122.99.99) Locked, Waiting on external entities

what is the trigger for recovery on timer expiry?
what codec should i choose for gtalk, i use acrobits on iphone with GSM
codec.


regards

budi




On Thu, Oct 28, 2010 at 6:32 AM, Michael Collins <msc at freeswitch.org> wrote:

> First, update to the latest git head. Second, turn on dingaling debugging
> and capture the output:
> dl_debug on
>
> Hopefully there will be clues.
> -MC
>
>
> On Wed, Oct 27, 2010 at 3:44 PM, budi wibowo <bwibowo at gmail.com> wrote:
>
>> hi
>> i use FreeSWITCH version: 1.0.head (git-cf5c1d2 2010-10-20 16-40-26 -0400)
>> i have followed the suggestion but still found this error
>>
>>
>>
>> 2010-10-28 06:41:45.720905 [ERR] libdingaling.c:1205 NODE ERROR!
>>
>> 2010-10-28 06:41:45.934903 [ERR] libdingaling.c:1205 NODE ERROR!
>>
>> 2010-10-28 06:41:46.158907 [ERR] libdingaling.c:1205 NODE ERROR!
>>
>> 2010-10-28 06:41:46.379903 [ERR] libdingaling.c:1205 NODE ERROR!
>>
>> 2010-10-28 06:41:46.589904 [ERR] libdingaling.c:1209 DISCONNECTED!
>>
>> any help is highly appreciate
>>
>>
>> On Thu, Oct 14, 2010 at 12:28 PM, Jeffrey Leung <curriegrad2004 at gmail.com
>> > wrote:
>>
>>> This is my config inside the jingle_profiles:
>>>  <profile type="client">
>>> *    <param name="name" value="gtalk[Example]"/>*
>>>    <param name="login" value="[EXAMPLE]@gmail.com/talk"/>
>>>    <param name="password" value="[REDACTED]"/>
>>>
>>>    <param name="dialplan" value="XML"/>
>>>    <param name="context" value="public"/>
>>>    <param name="message" value="Online"/>
>>>
>>>    <param name="rtp-ip" value="$${bind_server_ip}"/>
>>>    <!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
>>>
>>>    <param name="auto-login" value="true"/>
>>>     <!-- SASL "plain" or "md5" -->
>>>
>>>    <param name="sasl" value="plain"/>
>>>    <!-- if the server where the jabber is hosted is not the same as the
>>> one in the jid -->
>>>
>>>    <param name="server" value="talk.google.com"/>
>>>    <!-- Enable TLS or not -->
>>>
>>>    <param name="tls" value="true"/>
>>>     <!-- disable to trade async for more calls -->
>>>
>>>    <param name="use-rtp-timer" value="true"/>
>>>    <!-- default extension (if one cannot be determined) -->
>>>    <param name="exten" value="[EXTENSION]"/>
>>>    <!-- VAD choose one -->
>>>    <!-- <param name="vad" value="in"/> -->
>>>    <!-- <param name="vad" value="out"/> -->
>>>
>>>    <param name="vad" value="both"/>
>>>     <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
>>>
>>>    <param name="candidate-acl" value="wan.auto"/>
>>>    <param name="local-network-acl" value="localnet.auto"/>
>>>   </profile>
>>>
>>> And dialplan to follow:
>>>
>>>  <extension name="gvoice_out">
>>>    <condition field="destination_number" expression="^4664(\d{10})$">
>>> *      <action application="bridge" data="dingaling/gtalk[EXAMPLE]/+1$
>>> 1 at voice.google.com"/>*
>>>    </condition>
>>>  </extension>
>>>
>>> Pay special attention to the bolded areas of the configuration example
>>> I've provided. Configuring Google talk to work with FreeSwitch is quite
>>> straightforward with my configuration example I've provided above. I've ran
>>> to this same configuration ambiguity when they first completed this feature
>>> anyways, so I hope my configuration example does help you into resolving
>>> that problem
>>>
>>>
>>> On Wed, Oct 13, 2010 at 3:07 PM, Rob Hutton <justlikeef at gmail.com>
>>> wrote:
>>> > Caught and changed that, but it does the same thing.  I also changed
>>> the profile name to gtalk in case there was a case sensitivity issue or
>>> something, but same result.
>>> >
>>> > I am told that the wiki is not quite correct either, but with everyone
>>> away from their development PCs, it will be the weekend before the
>>> corrections are available.
>>> >
>>> > --
>>> > Thanks,
>>> > Rob
>>> > On Wednesday 13 October 2010 17:51:07 Michael Collins wrote:
>>> >> > Here is the profile:
>>> >> >
>>> >> >  <profile name="dingaling_1">
>>> >> >
>>> >> I believe that this should be:
>>> >> <profile type="client">
>>> >>
>>> >> -MC
>>> >>
>>> > I
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From abubacker at bksystems.co.in  Thu Oct 28 22:43:08 2010
From: abubacker at bksystems.co.in (abubacker)
Date: Fri, 29 Oct 2010 11:13:08 +0530
Subject: [Freeswitch-users] Same caller id comes in freeswitch
Message-ID: <4CCA5EEC.3040508@bksys.co.in>

Dear all ,
     I faced an issue which comes rarely , when the freeswitch process 
the calls in a queue
it would give the same customer caller id to the agents


$ fifo list
<bridge fifo_name="test_fifo" bridge_start="2010-10-29 11:03:51" 
bridge_start_epoch="1288330431">
<caller uuid="161fc7e8-e31e-11df-9a13-f1c4ac37d22d" caller_id_name="" 
caller_id_number="1"></caller>
<consumer>
<uuid>1b4c7090-e31e-11df-9a28-f1c4ac37d22d</uuid>
<outgoing_uuid>c21108c2f348bea840a8bab0dbc3458a</outgoing_uuid>
</consumer>
</bridge>
<bridge fifo_name="test_fifo" bridge_start="2010-10-29 11:03:58" 
bridge_start_epoch="1288330438">
<caller uuid="164f03b4-e31e-11df-9a14-f1c4ac37d22d" caller_id_name="" 
caller_id_number="2"></caller>
<consumer>
<uuid>1c7e4614-e31e-11df-9a2c-f1c4ac37d22d</uuid>
<outgoing_uuid>906dab93f19c508ba53ef7628ee7ce3b</outgoing_uuid>
</consumer>
</bridge>

above output  says that the caller_id_number 1 and 2 are connected with 
the agent 1000 and 1001
but in the twinkle it says  both 1 and 1 are connected with the agent 
1000 and agent 1001
why it is showing bad caller id

Note: Don't know whether attachment is allowed ( sorry if it has been 
prohibited )

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

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

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From johnsonw at eallway.com  Thu Oct 28 17:58:36 2010
From: johnsonw at eallway.com (Johnson)
Date: Thu, 28 Oct 2010 17:58:36 -0700 (PDT)
Subject: [Freeswitch-users] FS incoming calls Queue ACD implement
In-Reply-To: <AANLkTim+tWdjghr3AhP-eFX9nu4rmk291y2y0AxPHAB4@mail.gmail.com>
References: <1288212978229-5680281.post@n2.nabble.com>
	<AANLkTinuCX6xrzXHEg1pA43_TXnauSr4EDpnQhmRdPe-@mail.gmail.com>
	<AANLkTinuy+DhD0mZGj0BWOMx2Ha8sTXLetnQWLNK9PX6@mail.gmail.com>
	<AANLkTim+tWdjghr3AhP-eFX9nu4rmk291y2y0AxPHAB4@mail.gmail.com>
Message-ID: <1288313916151-5684872.post@n2.nabble.com>


Hi Anthony,

IT is much great news for this community. If inputing 'git pull', could i
get the new mod_fifo version? I can't wait for my taste. 

Thank you for you excited message!


Johnson

 
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-incoming-calls-Queue-ACD-implement-tp5680281p5684872.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From johnsonw at eallway.com  Thu Oct 28 18:01:05 2010
From: johnsonw at eallway.com (Johnson)
Date: Thu, 28 Oct 2010 18:01:05 -0700 (PDT)
Subject: [Freeswitch-users] FS incoming calls Queue ACD implement
In-Reply-To: <AANLkTinxF6O_P2qHrTRcaqVLwrO4zkaedzHrfOJQ0aKa@mail.gmail.com>
References: <1288212978229-5680281.post@n2.nabble.com>
	<AANLkTinuCX6xrzXHEg1pA43_TXnauSr4EDpnQhmRdPe-@mail.gmail.com>
	<AANLkTinuy+DhD0mZGj0BWOMx2Ha8sTXLetnQWLNK9PX6@mail.gmail.com>
	<AANLkTikEf2AhP6n4Lg+jS2t-oG-fxhZU9aMfjn80gxMa@mail.gmail.com>
	<AANLkTinxF6O_P2qHrTRcaqVLwrO4zkaedzHrfOJQ0aKa@mail.gmail.com>
Message-ID: <1288314065753-5684879.post@n2.nabble.com>


Steve, you are right. The call callcenter version still had issues need to
fiigure.  
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-incoming-calls-Queue-ACD-implement-tp5680281p5684879.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From thisjoy0528 at gmail.com  Fri Oct 29 04:06:19 2010
From: thisjoy0528 at gmail.com (joy this)
Date: Fri, 29 Oct 2010 19:06:19 +0800
Subject: [Freeswitch-users] problem with jingaling (may cause by TLS)
In-Reply-To: <1288277801895-5682888.post@n2.nabble.com>
References: <AANLkTik87R0ZLuM0zhY8F4nsV3eiWkHHPKB0qNdvf3RC@mail.gmail.com>
	<1288277801895-5682888.post@n2.nabble.com>
Message-ID: <AANLkTinEpQZY315pRNWNsMc1u4ayLec6f97ry+_1WpY+@mail.gmail.com>

OK. I will keep trying. Thank you for suggestions.

2010/10/28 Jeff Lenk <jeff at jefflenk.com>

>
> This is not currently supported in the current windows iksemel project. The
> required code for tls has been conditionally compiled out.
>
> Do a search for WIN32 in the project!
>
> If you are able to make this work please post your recipe to the Wiki so we
> can correct the docs there.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/problem-with-jingaling-may-cause-by-TLS-tp5682569p5682888.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch at peely.com  Fri Oct 29 06:21:23 2010
From: freeswitch at peely.com (peely)
Date: Fri, 29 Oct 2010 06:21:23 -0700 (PDT)
Subject: [Freeswitch-users] Problem with T.38 fax receive using ESL
In-Reply-To: <AANLkTinzujxqm7gT8k20+yeSKR+nYSRzjmdUSg_BGLZT@mail.gmail.com>
References: <1288272464502-5682561.post@n2.nabble.com>
	<AANLkTinzujxqm7gT8k20+yeSKR+nYSRzjmdUSg_BGLZT@mail.gmail.com>
Message-ID: <1288358483496-5686430.post@n2.nabble.com>


Thanks.

I'll test this when I have a combination of fax machine and compiler in the
same place, but note that I have tried setting the variables in the dialplan
entry before calling the ASL application.

If I get stuck I'll post some more detailed logs / traces etc.


Cheers,



Neil.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-T-38-fax-receive-using-ESL-tp5682561p5686430.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Fri Oct 29 07:26:16 2010
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 29 Oct 2010 07:26:16 -0700 (PDT)
Subject: [Freeswitch-users] PinDr0p: A GaTec System to Trace Call Paths
 Across Multiple Networks
Message-ID: <1288362376370-5686679.post@n2.nabble.com>


Someone recently posted this 
http://www.gatech.edu/newsroom/release.html?nid=61428 linky  on a 
http://www.dslreports.com/forum/r25000639-General-PinDr0p DSLR VoIP Forum .
The project is called PinDr0p and it uses Single-Ended Audio Features to
Determine Call Provenance through a series of algorithms to detect and
analyze call artifacts. From an engineering point of view, the algorithms
employed perhaps use some sort of orthogonal transformations which may
require a lot of CPU power to transform/analyze segments of audio streams in
a time domain to a frequency domain. Nevertheless, I posted this information
here to see if it may interest FS developers to incorporate and implement
such a scheme into the next releases of FS if the source codes are made
available publicly. After all, the project is funded by NSF.

-----
don't and stop are the ONLY two 4-letter words considered offensive to men,
but not when used together.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/PinDr0p-A-GaTec-System-to-Trace-Call-Paths-Across-Multiple-Networks-tp5686679p5686679.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Fri Oct 29 07:28:54 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 29 Oct 2010 09:28:54 -0500
Subject: [Freeswitch-users] FS incoming calls Queue ACD implement
In-Reply-To: <1288313916151-5684872.post@n2.nabble.com>
References: <1288212978229-5680281.post@n2.nabble.com>
	<AANLkTinuCX6xrzXHEg1pA43_TXnauSr4EDpnQhmRdPe-@mail.gmail.com>
	<AANLkTinuy+DhD0mZGj0BWOMx2Ha8sTXLetnQWLNK9PX6@mail.gmail.com>
	<AANLkTim+tWdjghr3AhP-eFX9nu4rmk291y2y0AxPHAB4@mail.gmail.com>
	<1288313916151-5684872.post@n2.nabble.com>
Message-ID: <AANLkTi=7+L5ZnE21_Bm_Jb-p=ib0UMUqsuronFrqk5=n@mail.gmail.com>

yes it's been in git for months now =D

On Thu, Oct 28, 2010 at 7:58 PM, Johnson <johnsonw at eallway.com> wrote:
>
> Hi Anthony,
>
> IT is much great news for this community. If inputing 'git pull', could i
> get the new mod_fifo version? I can't wait for my taste.
>
> Thank you for you excited message!
>
>
> Johnson
>
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-incoming-calls-Queue-ACD-implement-tp5680281p5684872.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From fdelawarde at wirelessmundi.com  Fri Oct 29 08:30:34 2010
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Fri, 29 Oct 2010 17:30:34 +0200
Subject: [Freeswitch-users] microsecond timestamps
Message-ID: <1288366234.6138.64.camel@luna.tc.commsmundi.com>

I've just seen that there are some channel variables
(Caller-Channel-Created-Time, ...) having timestamps in micro-seconds.

It is quite cool to see such resolution, but I was just wondering if
there was a practical use for it.

Does anyone actually uses such fine granularity in their CDRs or
elsewhere?
Do carrier environments have such requirements (and why?)?

Fran?ois.



From jeff at jefflenk.com  Fri Oct 29 09:22:51 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 29 Oct 2010 09:22:51 -0700 (PDT)
Subject: [Freeswitch-users] Weekly Binaries for Windows are available
Message-ID: <1288369371030-5687130.post@n2.nabble.com>


These evaluation binaries are built with VS2010(release mode) and are
packaged in a MSI for easy distribution and installation. They are available
in both X86 and X64 versions and install a base version of FreeSWITCH
including all default modules along with 8Khz sounds.

mod_managed can be used also but you will need to install the DotNet 4
Framework -
http://www.microsoft.com/downloads/en/details.aspx?FamilyID=9cfb2d51-5ff4-4491-b0e5-b386f32c0992&displaylang=en

Then enable the module in modules.conf.

NOTICE!! Make sure if you install a newer version of the MSI you create
backup copies of your /conf folder so you dont lose any configuration
settings.

New versions of the MSI will be available every week at approx 2am(GMT-6)
Sunday morning barring any unforeseen problems.
http://files.freeswitch.org/windows/installer/

The MSI will install v9 of the MSCRT libraries to your computer but
otherwise makes no modifications other than placing files in your program
files\FreeSWITCH folder and adding a "FreeSWITCH" menu entry in your start
menu.

This is a work in progress and will have frequent modifications if needed to
correct any problems.

You can also download and build the source code for yourself here.
http://wiki.freeswitch.org/wiki/Installation_Guide#Obtaining_the_Source_Code

Please enjoy and give feedback!
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Weekly-Binaries-for-Windows-are-available-tp5687130p5687130.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Fri Oct 29 09:29:07 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 29 Oct 2010 17:29:07 +0100
Subject: [Freeswitch-users] microsecond timestamps
In-Reply-To: <1288366234.6138.64.camel@luna.tc.commsmundi.com>
References: <1288366234.6138.64.camel@luna.tc.commsmundi.com>
Message-ID: <AANLkTin33EDuN2DKmo9gLHM_+i9xH2kttG1r75rbZCqO@mail.gmail.com>

I make use of it to measure PDD.

-Steve


On 29 October 2010 16:30, Fran?ois Delawarde
<fdelawarde at wirelessmundi.com> wrote:
> I've just seen that there are some channel variables
> (Caller-Channel-Created-Time, ...) having timestamps in micro-seconds.
>
> It is quite cool to see such resolution, but I was just wondering if
> there was a practical use for it.
>
> Does anyone actually uses such fine granularity in their CDRs or
> elsewhere?
> Do carrier environments have such requirements (and why?)?
>
> Fran?ois.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From Prometheus001 at gmx.net  Fri Oct 29 10:07:16 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 29 Oct 2010 19:07:16 +0200
Subject: [Freeswitch-users] progressinband=yes in freeswitch?
Message-ID: <4CCAFF44.2000402@gmx.net>

Hello,

we have some delays with media setup with Aastra SIP over DECT phones.
Aastra recommends using "progressinband=yes" for Asterisk so that the
media to the phone is already setup during ringing phase.
Are there any parameter in Freeswitch to support that?

If yes: As we have registered more than one phone (Aastra SIP/DECT +
Snom) to one extension, do we expect any problems in that case when
media is already setup to both phones during ringing time?

Best regards
Peter



From fraserredmond at gmail.com  Fri Oct 29 10:28:01 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 29 Oct 2010 18:28:01 +0100
Subject: [Freeswitch-users] Loss of first second of media
In-Reply-To: <AANLkTikLtKhBiCZeMY+GH6X7jb4wVdybVF=OiMePK8Rm@mail.gmail.com>
References: <AANLkTimECwtVmnbFsBJRiN3YSaGyXbvh-sKtKYPey8xx@mail.gmail.com>
	<AANLkTik5H74MupY8iyx6Ao4=Rv=wCm8jv-w=xCtSnm0Z@mail.gmail.com>
	<AANLkTikpVH-s6AvJ7fQoZu2PE4dduTg+d-ygyBWSfr4R@mail.gmail.com>
	<AANLkTi=FZDqev9R1aTMpdvTk8t9NfxiyeisQSRBxfTEb@mail.gmail.com>
	<AANLkTikY68ovuiSO9xkbZZPSskxCNsjrpTSdb-42bwJB@mail.gmail.com>
	<AANLkTikiG3xPKDVkwU4GDk5t6KDe67AQmM-dkheMCSu7@mail.gmail.com>
	<AANLkTinJRcpRx_BQyKifECcurYb+vG2P8Nct1xT_X3XO@mail.gmail.com>
	<AANLkTinpaRivNHteDp4i6WJjmgsPAE6otVx5LjgJcvWr@mail.gmail.com>
	<AANLkTikiVxFAOwftrmfqd3LtD4H__UUytsM117WBhZy0@mail.gmail.com>
	<AANLkTikLtKhBiCZeMY+GH6X7jb4wVdybVF=OiMePK8Rm@mail.gmail.com>
Message-ID: <AANLkTimyRrtba8nD7jkWJY2DysebRaycpvRXY3b_A3r-@mail.gmail.com>

Thanks Anthony, it's here:
http://pastebin.freeswitch.org/14350

And pcap is attached.

The call connects around (or just before) the 16:58:35 mark (line 558 is
what I see in the terminal while waiting for it to connect - both
early-media and the missing start of the media)

Cheers,
Fraser




On Thu, Oct 28, 2010 at 3:59 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Can you do this trace with debug level logging in addition to the sip trace
> console loglevel debug
>
> you also may want to get a pcap of it
>
> tshark udp and port 5060 -w test.pcap
>
>
>
> On Sat, Oct 23, 2010 at 11:44 AM, Fraser Redmond
> <fraserredmond at gmail.com> wrote:
> > Thanks Anthony,
> >
> > Finally managed to get a sip trace - could you do me a favor and take a
> look
> > and/or give me some ideas of what to look for?
> >
> > http://pastebin.freeswitch.org/14300
> >
> > I've highlighted lines 168 and 193. In between these lines is where the
> > number is dialed and rings once, then picks up, then theres silence for a
> > second or two, and that second SIP message is when I start hearing audio.
> >
> > Thanks,
> > Fraser
> >
> >
> >
> >
> > On Thu, Oct 21, 2010 at 6:44 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> its a blue message on cli
> >>
> >> It could also be the other side expecting us to send media first or
> >> something silly.
> >> try getting a sip trace and figure out when the rtp starts arriving.
> >>
> >>
> >> On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond
> >> <fraserredmond at gmail.com> wrote:
> >> > Sorry, yes, I am setting ignore_early_media=true in the first area.
> (Or
> >> > are
> >> > you saying that should be off? I forget now why I needed it on, but
> >> > there
> >> > was a reason I added it.)
> >> >
> >> > Yes, the bridge doesn't start until after the A-leg has answered.
> >> >
> >> > Thanks for the suggestion about nat/auto-changing port, I'll have a
> look
> >> > into that - would that be in the cli output or in a sip trace? I've
> >> > already
> >> > looked and it's not appearing in the CLI output (with loglevel=debug),
> >> > haven't looked in the sip trace yet.
> >> >
> >> > Cheers,
> >> > Fraser
> >> >
> >> >
> >> >
> >> >
> >> > On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> are you setting ignore_early_media=true in the first vars=values
> area?
> >> >>
> >> >> This looks like you could be calling one leg who is still not
> answered
> >> >> and then bridging it to another dest.  The bridge app will wait for
> >> >> the first leg to answer before bridging.
> >> >>
> >> >> Also if you have any NAT anywhere, look for an "auto-changing port"
> >> >> type message which can also be attributed to this due to a detection
> >> >> period for incorrect ports.
> >> >>
> >> >>
> >> >>
> >> >> On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond
> >> >> <fraserredmond at gmail.com> wrote:
> >> >> > event_socket:
> >> >> > api originate {vars=values}user/$fromExtn at Domain
> >> >> > 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline
> >> >> >
> >> >> > then
> >> >> >
> >> >> > dialplan:
> >> >> > <action application="set"
> >> >> > data="effective_caller_id_number=+1800number"/>
> >> >> > <action application="set" data="effective_caller_id_name="/>
> >> >> > (set and/or export a bunch of other vars too)
> >> >> > <action application="set"
> >> >> >
> >> >> >
> >> >> > data="dial_string=sofia/gateway/
> gatewayname.com/00${destination_number}<http://gatewayname.com/00$%7Bdestination_number%7D>
> "/>
> >> >> > <action application="bridge" data="${dial_string}"/>
> >> >> >
> >> >> > Cheers,
> >> >> > Fraser
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> > On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale
> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >>
> >> >> >> how are you accomplishing that? by which technique?
> >> >> >>
> >> >> >> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond
> >> >> >> <fraserredmond at gmail.com> wrote:
> >> >> >> > The call is originated from Freeswitch (via CLI) to a softphone,
> >> >> >> > then
> >> >> >> > when
> >> >> >> > that is connected it bridges out to the gateway.
> >> >> >> >
> >> >> >> > Cheers,
> >> >> >> > Fraser
> >> >> >> >
> >> >> >> >
> >> >> >> >
> >> >> >> >
> >> >> >> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale
> >> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >> >>
> >> >> >> >> Where is the other side of this call coming from?
> >> >> >> >>
> >> >> >> >> [ (   ) ] -> FS -> (PSTN via SIP)
> >> >> >> >>
> >> >> >> >> What goes in the empty space above?
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > 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<MSN%3Aanthony_minessale at hotmail.com>
> >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >> IRC: irc.freenode.net #freeswitch
> >> >>
> >> >> FreeSWITCH Developer Conference
> >> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >> pstn:+19193869900
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From curriegrad2004 at gmail.com  Fri Oct 29 10:44:37 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Fri, 29 Oct 2010 10:44:37 -0700
Subject: [Freeswitch-users] problem with jingaling (may cause by TLS)
In-Reply-To: <AANLkTinEpQZY315pRNWNsMc1u4ayLec6f97ry+_1WpY+@mail.gmail.com>
References: <AANLkTik87R0ZLuM0zhY8F4nsV3eiWkHHPKB0qNdvf3RC@mail.gmail.com>
	<1288277801895-5682888.post@n2.nabble.com>
	<AANLkTinEpQZY315pRNWNsMc1u4ayLec6f97ry+_1WpY+@mail.gmail.com>
Message-ID: <AANLkTine9VkkCLus-+QHZkzEE7HAHvC_Di=yNX2oqV6g@mail.gmail.com>

You'll need to patch the code to some extent for mod_dingaling with
TLS support on Windows to work. For now, run FreeSwitch under
CentOS/RHEL 5.x for now. Fedora 13's libgnutls is broken, so expect
some segfaults when you try to load mod_dingaling under that distro.

On Fri, Oct 29, 2010 at 4:06 AM, joy this <thisjoy0528 at gmail.com> wrote:
> OK. I will keep trying. Thank you for suggestions.
>
> 2010/10/28 Jeff Lenk <jeff at jefflenk.com>
>>
>> This is not currently supported in the current windows iksemel project.
>> The
>> required code for tls has been conditionally compiled out.
>>
>> Do a search for WIN32 in the project!
>>
>> If you are able to make this work please post your recipe to the Wiki so
>> we
>> can correct the docs there.
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/problem-with-jingaling-may-cause-by-TLS-tp5682569p5682888.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From curriegrad2004 at gmail.com  Fri Oct 29 10:47:21 2010
From: curriegrad2004 at gmail.com (Jeffrey Leung)
Date: Fri, 29 Oct 2010 10:47:21 -0700
Subject: [Freeswitch-users] Weekly Binaries for Windows are available
In-Reply-To: <1288369371030-5687130.post@n2.nabble.com>
References: <1288369371030-5687130.post@n2.nabble.com>
Message-ID: <AANLkTimicPsVeTU2z-wyhn2v3SM3ODtZWcaEomsP7SWr@mail.gmail.com>

Thanks for releasing these builds of FreeSwitch under this new MSI
based installer. However, I'd like to know why VS2010 is used for the
weekly builds instead of VS2008 SP1. At the moment, VS2008 seems more
stable and mature than VS2010, in my opinion.

On Fri, Oct 29, 2010 at 9:22 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
>
> These evaluation binaries are built with VS2010(release mode) and are
> packaged in a MSI for easy distribution and installation. They are available
> in both X86 and X64 versions and install a base version of FreeSWITCH
> including all default modules along with 8Khz sounds.
>
> mod_managed can be used also but you will need to install the DotNet 4
> Framework -
> http://www.microsoft.com/downloads/en/details.aspx?FamilyID=9cfb2d51-5ff4-4491-b0e5-b386f32c0992&displaylang=en
>
> Then enable the module in modules.conf.
>
> NOTICE!! Make sure if you install a newer version of the MSI you create
> backup copies of your /conf folder so you dont lose any configuration
> settings.
>
> New versions of the MSI will be available every week at approx 2am(GMT-6)
> Sunday morning barring any unforeseen problems.
> http://files.freeswitch.org/windows/installer/
>
> The MSI will install v9 of the MSCRT libraries to your computer but
> otherwise makes no modifications other than placing files in your program
> files\FreeSWITCH folder and adding a "FreeSWITCH" menu entry in your start
> menu.
>
> This is a work in progress and will have frequent modifications if needed to
> correct any problems.
>
> You can also download and build the source code for yourself here.
> http://wiki.freeswitch.org/wiki/Installation_Guide#Obtaining_the_Source_Code
>
> Please enjoy and give feedback!
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Weekly-Binaries-for-Windows-are-available-tp5687130p5687130.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From pjintheusa at gmail.com  Fri Oct 29 11:14:17 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 29 Oct 2010 14:14:17 -0400
Subject: [Freeswitch-users] Weekly Binaries for Windows are available
In-Reply-To: <AANLkTimicPsVeTU2z-wyhn2v3SM3ODtZWcaEomsP7SWr@mail.gmail.com>
References: <1288369371030-5687130.post@n2.nabble.com>
	<AANLkTimicPsVeTU2z-wyhn2v3SM3ODtZWcaEomsP7SWr@mail.gmail.com>
Message-ID: <AANLkTinBWo77RMPf=+Ttk8+j=npwhMB8f_WzfRrukrPF@mail.gmail.com>

Jeffrey - I am confused? Why should it matter what dev env Jeff used if he
is providing an MSI installable build? Surely you can do the same with
VS2008 if that is what you prefer to use? The sln is there.

I am not knocking you here. I am genuinely interested in why if makes a
difference.

Thanks Jeff for your efforts. These are very helpful.

On Fri, Oct 29, 2010 at 1:47 PM, Jeffrey Leung <curriegrad2004 at gmail.com>wrote:

> Thanks for releasing these builds of FreeSwitch under this new MSI
> based installer. However, I'd like to know why VS2010 is used for the
> weekly builds instead of VS2008 SP1. At the moment, VS2008 seems more
> stable and mature than VS2010, in my opinion.
>
> On Fri, Oct 29, 2010 at 9:22 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
> >
> > These evaluation binaries are built with VS2010(release mode) and are
> > packaged in a MSI for easy distribution and installation. They are
> available
> > in both X86 and X64 versions and install a base version of FreeSWITCH
> > including all default modules along with 8Khz sounds.
> >
> > mod_managed can be used also but you will need to install the DotNet 4
> > Framework -
> >
> http://www.microsoft.com/downloads/en/details.aspx?FamilyID=9cfb2d51-5ff4-4491-b0e5-b386f32c0992&displaylang=en
> >
> > Then enable the module in modules.conf.
> >
> > NOTICE!! Make sure if you install a newer version of the MSI you create
> > backup copies of your /conf folder so you dont lose any configuration
> > settings.
> >
> > New versions of the MSI will be available every week at approx 2am(GMT-6)
> > Sunday morning barring any unforeseen problems.
> > http://files.freeswitch.org/windows/installer/
> >
> > The MSI will install v9 of the MSCRT libraries to your computer but
> > otherwise makes no modifications other than placing files in your program
> > files\FreeSWITCH folder and adding a "FreeSWITCH" menu entry in your
> start
> > menu.
> >
> > This is a work in progress and will have frequent modifications if needed
> to
> > correct any problems.
> >
> > You can also download and build the source code for yourself here.
> >
> http://wiki.freeswitch.org/wiki/Installation_Guide#Obtaining_the_Source_Code
> >
> > Please enjoy and give feedback!
> > --
> > View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Weekly-Binaries-for-Windows-are-available-tp5687130p5687130.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Fri Oct 29 11:22:36 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 29 Oct 2010 13:22:36 -0500
Subject: [Freeswitch-users] progressinband=yes in freeswitch?
In-Reply-To: <4CCAFF44.2000402@gmx.net>
References: <4CCAFF44.2000402@gmx.net>
Message-ID: <AANLkTimUUZZnWMT3+stuS-bKMZNCaDHxjNW5eJOvumF=@mail.gmail.com>

that's the equiv of

{ignore_early_media=true}
and set ringback var before bridge

<action application="set" data="ringback=${us-ring}"/>


On Fri, Oct 29, 2010 at 12:07 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Hello,
>
> we have some delays with media setup with Aastra SIP over DECT phones.
> Aastra recommends using "progressinband=yes" for Asterisk so that the
> media to the phone is already setup during ringing phase.
> Are there any parameter in Freeswitch to support that?
>
> If yes: As we have registered more than one phone (Aastra SIP/DECT +
> Snom) to one extension, do we expect any problems in that case when
> media is already setup to both phones during ringing time?
>
> Best regards
> Peter
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From jeff at jefflenk.com  Fri Oct 29 11:30:48 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 29 Oct 2010 11:30:48 -0700 (PDT)
Subject: [Freeswitch-users] Weekly Binaries for Windows are available
In-Reply-To: <AANLkTimicPsVeTU2z-wyhn2v3SM3ODtZWcaEomsP7SWr@mail.gmail.com>
References: <1288369371030-5687130.post@n2.nabble.com>
	<AANLkTimicPsVeTU2z-wyhn2v3SM3ODtZWcaEomsP7SWr@mail.gmail.com>
Message-ID: <1288377048233-5687599.post@n2.nabble.com>


Sure,
VS2010 has much better support for msbuild enabled C projects (Project files
are msbuild versus the old format) and also supports the V4 managed
framework that will be at the forefront of managed development for years to
come. 

With that said I have not had any problems with VS2010 that stand out in my
mind. Lots of nice features including better support of Intelisense -much
better performance of lookups and dependency scanning which is very nice for
a solution of FS's size.

The VS2008 solution and projects will still be supported for those who wish
to build with it for quite a while I'm sure.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Weekly-Binaries-for-Windows-are-available-tp5687130p5687599.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Fri Oct 29 11:35:50 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 29 Oct 2010 13:35:50 -0500
Subject: [Freeswitch-users] Loss of first second of media
In-Reply-To: <AANLkTimyRrtba8nD7jkWJY2DysebRaycpvRXY3b_A3r-@mail.gmail.com>
References: <AANLkTimECwtVmnbFsBJRiN3YSaGyXbvh-sKtKYPey8xx@mail.gmail.com>
	<AANLkTik5H74MupY8iyx6Ao4=Rv=wCm8jv-w=xCtSnm0Z@mail.gmail.com>
	<AANLkTikpVH-s6AvJ7fQoZu2PE4dduTg+d-ygyBWSfr4R@mail.gmail.com>
	<AANLkTi=FZDqev9R1aTMpdvTk8t9NfxiyeisQSRBxfTEb@mail.gmail.com>
	<AANLkTikY68ovuiSO9xkbZZPSskxCNsjrpTSdb-42bwJB@mail.gmail.com>
	<AANLkTikiG3xPKDVkwU4GDk5t6KDe67AQmM-dkheMCSu7@mail.gmail.com>
	<AANLkTinJRcpRx_BQyKifECcurYb+vG2P8Nct1xT_X3XO@mail.gmail.com>
	<AANLkTinpaRivNHteDp4i6WJjmgsPAE6otVx5LjgJcvWr@mail.gmail.com>
	<AANLkTikiVxFAOwftrmfqd3LtD4H__UUytsM117WBhZy0@mail.gmail.com>
	<AANLkTikLtKhBiCZeMY+GH6X7jb4wVdybVF=OiMePK8Rm@mail.gmail.com>
	<AANLkTimyRrtba8nD7jkWJY2DysebRaycpvRXY3b_A3r-@mail.gmail.com>
Message-ID: <AANLkTinrSNUr49b3LE6c6ZP7NZ9d13t1CK2bPW2x+anD@mail.gmail.com>

can you try another one with just udp and not "port 5060"
so I can see the rtp too


On Fri, Oct 29, 2010 at 12:28 PM, Fraser Redmond
<fraserredmond at gmail.com> wrote:
> Thanks Anthony, it's here:
> http://pastebin.freeswitch.org/14350
>
> And pcap is attached.
>
> The call connects around (or just before) the 16:58:35 mark (line 558 is
> what I see in the terminal while waiting for it to connect - both
> early-media and the missing start of the media)
>
> Cheers,
> Fraser
>
>
>
>
> On Thu, Oct 28, 2010 at 3:59 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> Can you do this trace with debug level logging in addition to the sip
>> trace
>> console loglevel debug
>>
>> you also may want to get a pcap of it
>>
>> tshark udp and port 5060 -w test.pcap
>>
>>
>>
>> On Sat, Oct 23, 2010 at 11:44 AM, Fraser Redmond
>> <fraserredmond at gmail.com> wrote:
>> > Thanks Anthony,
>> >
>> > Finally managed to get a sip trace - could you do me a favor and take a
>> > look
>> > and/or give me some ideas of what to look for?
>> >
>> > http://pastebin.freeswitch.org/14300
>> >
>> > I've highlighted lines 168 and 193. In between these lines is where the
>> > number is dialed and rings once, then picks up, then theres silence for
>> > a
>> > second or two, and that second SIP message is when I start hearing
>> > audio.
>> >
>> > Thanks,
>> > Fraser
>> >
>> >
>> >
>> >
>> > On Thu, Oct 21, 2010 at 6:44 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> its a blue message on cli
>> >>
>> >> It could also be the other side expecting us to send media first or
>> >> something silly.
>> >> try getting a sip trace and figure out when the rtp starts arriving.
>> >>
>> >>
>> >> On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond
>> >> <fraserredmond at gmail.com> wrote:
>> >> > Sorry, yes, I am setting ignore_early_media=true in the first area.
>> >> > (Or
>> >> > are
>> >> > you saying that should be off? I forget now why I needed it on, but
>> >> > there
>> >> > was a reason I added it.)
>> >> >
>> >> > Yes, the bridge doesn't start until after the A-leg has answered.
>> >> >
>> >> > Thanks for the suggestion about nat/auto-changing port, I'll have a
>> >> > look
>> >> > into that - would that be in the cli output or in a sip trace? I've
>> >> > already
>> >> > looked and it's not appearing in the CLI output (with
>> >> > loglevel=debug),
>> >> > haven't looked in the sip trace yet.
>> >> >
>> >> > Cheers,
>> >> > Fraser
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale
>> >> > <anthony.minessale at gmail.com> wrote:
>> >> >>
>> >> >> are you setting ignore_early_media=true in the first vars=values
>> >> >> area?
>> >> >>
>> >> >> This looks like you could be calling one leg who is still not
>> >> >> answered
>> >> >> and then bridging it to another dest. ?The bridge app will wait for
>> >> >> the first leg to answer before bridging.
>> >> >>
>> >> >> Also if you have any NAT anywhere, look for an "auto-changing port"
>> >> >> type message which can also be attributed to this due to a detection
>> >> >> period for incorrect ports.
>> >> >>
>> >> >>
>> >> >>
>> >> >> On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond
>> >> >> <fraserredmond at gmail.com> wrote:
>> >> >> > event_socket:
>> >> >> > api originate {vars=values}user/$fromExtn at Domain
>> >> >> > 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline
>> >> >> >
>> >> >> > then
>> >> >> >
>> >> >> > dialplan:
>> >> >> > <action application="set"
>> >> >> > data="effective_caller_id_number=+1800number"/>
>> >> >> > <action application="set" data="effective_caller_id_name="/>
>> >> >> > (set and/or export a bunch of other vars too)
>> >> >> > <action application="set"
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > data="dial_string=sofia/gateway/gatewayname.com/00${destination_number}"/>
>> >> >> > <action application="bridge" data="${dial_string}"/>
>> >> >> >
>> >> >> > Cheers,
>> >> >> > Fraser
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale
>> >> >> > <anthony.minessale at gmail.com> wrote:
>> >> >> >>
>> >> >> >> how are you accomplishing that? by which technique?
>> >> >> >>
>> >> >> >> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond
>> >> >> >> <fraserredmond at gmail.com> wrote:
>> >> >> >> > The call is originated from Freeswitch (via CLI) to a
>> >> >> >> > softphone,
>> >> >> >> > then
>> >> >> >> > when
>> >> >> >> > that is connected it bridges out to the gateway.
>> >> >> >> >
>> >> >> >> > Cheers,
>> >> >> >> > Fraser
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale
>> >> >> >> > <anthony.minessale at gmail.com> wrote:
>> >> >> >> >>
>> >> >> >> >> Where is the other side of this call coming from?
>> >> >> >> >>
>> >> >> >> >> [ ( ? ) ] -> FS -> (PSTN via SIP)
>> >> >> >> >>
>> >> >> >> >> What goes in the empty space above?
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > 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
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > 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
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From mario_fs at mgtech.com  Fri Oct 29 11:56:48 2010
From: mario_fs at mgtech.com (Mario G)
Date: Fri, 29 Oct 2010 11:56:48 -0700
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <AANLkTi=3Kd-bCHJSED7VEh-cvtpdXMs_EkWZL=s8J4Ux@mail.gmail.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
	<407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>
	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
	<AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
	<6D9FD5E2-21FA-4630-8CF4-A87ADB30AC87@mgtech.com>
	<AANLkTi=BiDiXLgEoj_MfwmZbGjvXoOuNJgUYdetciNCj@mail.gmail.com>
	<9BAFDAF8-1E66-444C-A024-A88FF65D8BCA@mgtech.com>
	<3447E946-0BF8-4F7E-B5E9-E28F53DD0968@mgtech.com>
	<AANLkTi=LRbz6ADhLvBKkjGxOywAVLf--GTkzXEpUoDbi@mail.gmail.com>
	<BE754258-BE27-454F-B740-527A35DBDFFE@mgtech.com>
	<AANLkTi=3Kd-bCHJSED7VEh-cvtpdXMs_EkWZL=s8J4Ux@mail.gmail.com>
Message-ID: <14205291-75B3-470F-ABA1-EE7F08A695E9@mgtech.com>

Oh my...  looks it was not NAT after all? Please help! I changed to all profiles to static per instructions below and still have the problem:
2010-10-29 11:15:18.536446 [NOTICE] sofia_reg.c:342 Registering uuid1
2010-10-29 11:15:34.313150 [WARNING] sofia_reg.c:387 uuid1 Failed Registration, setting retry to 15 seconds.

sofia global siptrace on did not show any activity for this gateway in or out, others were fine but eventually fail. I setup static:
1. set the params ext-sip-ip and ext-rtp-ip to my external static IP
2. map the sip ports (5060-5080) and all of the rtp ports (16384-32767)  to FS lan addr.
3. set sip-ip and rtp-ip to the lan addr of FS
4. start FS with -nonat

I don't know what to try next. BTW, the sofia status for the profiles shows stun enabled but I did not set it up anywhere:

Name             	uuid1
Domain Name      	N/A
Auto-NAT         	false
DBName           	sofia_reg_mvvyl
Pres Hosts       	
Dialplan         	XML
Context          	public
Challenge Realm  	auto_to
RTP-IP           	10.x.x.20
Ext-RTP-IP       	210.x.x.100
SIP-IP           	10.x.x.20
Ext-SIP-IP       	210.x.x.100
URL              	sip:mod_sofia at 210.x.x.100:5068
BIND-URL         	sip:mod_sofia at 210.x.x.100:5068;maddr=10.x.x.20
HOLD-MUSIC       	local_stream://moh
OUTBOUND-PROXY   	N/A
CODECS IN        	G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
CODECS OUT       	PCMU,PCMA,GSM
TEL-EVENT        	101
DTMF-MODE        	rfc2833
CNG              	13
SESSION-TO       	0
MAX-DIALOG       	0
NOMEDIA          	false
LATE-NEG         	false
PROXY-MEDIA      	false
AGGRESSIVENAT    	false
STUN-ENABLED     	true
STUN-AUTO-DISABLE	false
CALLS-IN         	0
FAILED-CALLS-IN  	0
CALLS-OUT        	0
FAILED-CALLS-OUT 	0


On Oct 27, 2010, at 10:04 AM, Anthony Minessale wrote:

> if you map it or not, a scanner would penetrate it.
> There are lot of sip scanners out there now, you just need to beware of them.
> 
> 
> On Wed, Oct 27, 2010 at 11:50 AM, Mario G <mario_fs at mgtech.com> wrote:
>> Thanks so much! I am sure many others will find this info invaluable. I will try the static route again but have one question: When I started with FS I found a "sip scanner"  in FS and someone on this group said not to use port mapping since it was a security risk. Is that true?
>> 
>> On Oct 27, 2010, at 9:10 AM, Anthony Minessale wrote:
>> 
>>> you are completely guessing at things.
>>> I want you to understand that the only reason you are having problems
>>> with this is because you don't understand how it works enough to know
>>> what you are doing 100%
>>> 
>>> Its a given that the pnp stuff is only for your dynamic IP.
>>> aggressive-nat-detection and sip-force-expires are all related to
>>> inbound calls when the things who are registering to you may be behind
>>> nat.
>>> 
>>> You need to learn the difference between which nat tools are
>>> *) designed for your FS to run behind nat
>>> *) designed for FS to run public and accept connections from devices behind nat.
>>> 
>>> If you have a static IP, you don't need the pnp stuff so -nonat is fine
>>> What you need to do is set

>>> 1) set the params ext-sip-ip and ext-rtp-ip to your external static IP
>>> 2) map the sip ports and all of the rtp ports from your static IP to FS lan addr
>>> 3) set sip-ip and rtp-ip to the lan addr you forwarded through.
>> 
>>> 
>>> If you don't do this: your outbound registration will use NAT to your
>>> provider and if there is no activity for the expire time on your NAT
>>> mapping the reverse port mapping from your provider back to you is
>>> lost.  This is why you set your register expires to a very low number,
>>> (you need to make sure the provider does not turn the expires back up
>>> in the reply because it will beat your choice *see sip trace) if this
>>> is the case then you need the "ping" option set to 30, to continuously
>>> send an options to your provider.
>>> 
>>> The static mapping is obviously the better, easier and more reliable solution.
>>> 
>>> So I want you to understand that the only way to keep a nat mapped
>>> port alive is to continuously send traffic, all the other methods that
>>> you are mentioning are to detect that phones registered to your are
>>> behind nat, I gave you that force-expires option before because your
>>> trace was full of inbound reg so I thought that is what you wanted
>>> help with.
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> On Wed, Oct 27, 2010 at 10:43 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>> I should mention that I did not have this problem with an SPA9000 PBX
>>>> (asterisk based) for over two years so FS may be pickier about upnp and/or
>>>> nat, or just better at it exposing a problem in the router.
>>>> I made different changes to the gateways to test different things. One
>>>> failed after 17 hours, the other two stayed up.  What did not work:
>>>> added <variable name="sip-force-expires" value="30"/> to the directory
>>>> entries as suggested.
>>>> set the gateway expire times to 30 seconds.
>>>> What worked (could be coincidental) for the two gateways that stayed up:
>>>> I Added <param name="aggressive-nat-detection" value="true"/>
>>>> I originally setup FS to use the static ip by setting external sip/rtp to
>>>> just the static ip (no autonat:) and ran with -nonat but I could not get
>>>> incoming calls. The only way it worked was to use autonat:1.2.3.4. The
>>>> router has 1 static public address and 1 dynamic external IP, this is the
>>>> root of the problem, upnp only tells FS about the dynamic ip  Will keep this
>>>> thread up-to-date for anyone who may be in the same boat someday. Thanks
>>>> again for looking at the trace.
>>>> Mario
>>>> 
>>>> You should be setting the req freq to a low number on the outbound gateways
>>>> 
>>>> The examples you showed had a series of inbound reg
>>>> 
>>>> also set expire-seconds to 30 in your gateway xml
>>>> 
>>>> 
>>>> The problem is if you are not constantly sending traffic to the box
>>>> 
>>>> the nat mapping will go away.
>>>> 
>>>> If you are in production you should be using a static ip with a static
>>>> 
>>>> mapping, any trouble you are having is your own fault for playing with
>>>> 
>>>> fire.  The best we can do is tell you how to keep it contained.
>>>> 
>>>> 
>>>> 
>>>> 
>>>> On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>> 
>>>> I made the change. I had no idea the settings for the inside phones effected
>>>> nat for the outside sip accounts. I was looking into aggressive-nat-
>>>> detection since the internal profile status always shows the right external
>>>> static IP but the nat_ap status always shows the dynamic ip. Crossing
>>>> fingers/etc since this problem is 85% of time (weeks!) into FS changeover.
>>>> Thanks!
>>>> 
>>>> Mario
>>>> 
>>>> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
>>>> 
>>>> add
>>>> 
>>>> <variable name="sip-force-expires" value="30"/>
>>>> 
>>>> to the <variables> section of your <user>
>>>> 
>>>> you have it at 600 and the nat mapping is timing out while the 600
>>>> 
>>>> seconds is ticking away
>>>> 
>>>> 
>>>> 
>>>> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>> 
>>>> From the TSP:
>>>> 
>>>> "I have enabled the SIP trace on your account. We are not currently seeing
>>>> 
>>>> any registration attempts to your account within the last 15 minutes. Please
>>>> 
>>>> restart FreeSwitch so that registration attempts begin again. Thank you. ".
>>>> 
>>>> So FS is not getting past router.
>>>> 
>>>> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>>>> 
>>>> I ran the global trace during the problem and it is
>>>> 
>>>> at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
>>>> 
>>>> "accttwo", "acct3". The trace includes phones since it was global. I am
>>>> 
>>>> using:
>>>> 
>>>>    <param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>>>> 
>>>>    <param name="ext-sip-ip" value="autonat:my-static.ip"/>
>>>> 
>>>> I tried dumping nat and removing the autonat: above and using -nonat but
>>>> 
>>>> that did not work, registration proceeded but no calls inbound.
>>>> 
>>>> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>>>> 
>>>> Whoops, I am using an IP address for at least one gateway so that is not the
>>>> 
>>>> problem:
>>>> 
>>>> They look outbound to me and I am using dns for 2 and an IP for one so that
>>>> 
>>>> is not the issue. I was able to get FS to clear this up by doing "nat_map
>>>> 
>>>> reinit" which is why I think this is a nat problem. I will do the trace you
>>>> 
>>>> mentioned. I will plug an ip address into one of the gateways to see what
>>>> 
>>>> happens, they all fail at once. Thanks for responding!
>>>> 
>>>> Mario
>>>> 
>>>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>>>> 
>>>> I really need help on this as I have weeks into this problem. I thought I
>>>> 
>>>> had it nailed but I guess not. After 5.5 hours I get:
>>>> 
>>>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>>>> 
>>>> Registration, setting retry to 15 seconds.
>>>> 
>>>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>> 
>>>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>> 
>>>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
>>>> 
>>>> Registration, setting retry to 30 seconds.
>>>> 
>>>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>>>> 
>>>> Registration, setting retry to 15 seconds.
>>>> 
>>>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>>>> 
>>>> Registration, setting retry to 15 seconds.
>>>> 
>>>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>> 
>>>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>> 
>>>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>>>> 
>>>> and no way to make/get calls until I restart FS. I did this:
>>>> 
>>>> 1. log 7
>>>> 
>>>> 2. sofia profile xxxx siptrace on   for each profile/gateway
>>>> 
>>>> 3. restarted router
>>>> 
>>>> All three did not solve the problem. The trace and log produced no
>>>> 
>>>> additional lines which is why I am wondering if FS has a problem since the
>>>> 
>>>> trace shows no SIP activity.
>>>> 
>>>> 3 gateways with 2 ITSPs
>>>> 
>>>> 2 DSL/WAN lines, 1 static and 1 dynamic
>>>> 
>>>> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
>>>> 
>>>> external static ip.
>>>> 
>>>> sofia status profile ... has the right ext ip
>>>> 
>>>> nat_map status shows the dynamic (wrong) IP
>>>> 
>>>> I tried starting with -nonat but that was worse
>>>> 
>>>> the only way to fix is restart FS.
>>>> 
>>>> I read the wiki on external nat, auto_nat and everything else many times.
>>>> 
>>>> Thanks Mario
>>>> 
>>>> 
>>>> _______________________________________________
>>>> 
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>>>> 
>>>> 
>>>> --
>>>> 
>>>> Anthony Minessale II
>>>> 
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> 
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>>>> 
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>>>> 
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>>>> --
>>>> 
>>>> Anthony Minessale II
>>>> 
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> 
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>>>> 
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
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>>> 
>>> 
>>> 
>>> --
>>> 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
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>>> 
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>>> 
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>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
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>> 
> 
> 
> 
> -- 
> 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
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org



From anthony.minessale at gmail.com  Fri Oct 29 12:11:37 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 29 Oct 2010 14:11:37 -0500
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <14205291-75B3-470F-ABA1-EE7F08A695E9@mgtech.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
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	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
	<AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
	<6D9FD5E2-21FA-4630-8CF4-A87ADB30AC87@mgtech.com>
	<AANLkTi=BiDiXLgEoj_MfwmZbGjvXoOuNJgUYdetciNCj@mail.gmail.com>
	<9BAFDAF8-1E66-444C-A024-A88FF65D8BCA@mgtech.com>
	<3447E946-0BF8-4F7E-B5E9-E28F53DD0968@mgtech.com>
	<AANLkTi=LRbz6ADhLvBKkjGxOywAVLf--GTkzXEpUoDbi@mail.gmail.com>
	<BE754258-BE27-454F-B740-527A35DBDFFE@mgtech.com>
	<AANLkTi=3Kd-bCHJSED7VEh-cvtpdXMs_EkWZL=s8J4Ux@mail.gmail.com>
	<14205291-75B3-470F-ABA1-EE7F08A695E9@mgtech.com>
Message-ID: <AANLkTi=D6siEAquSq6gYh-RnZeJJWrk11mHWK8De2tLh@mail.gmail.com>

stun-enabled must be true in your profile XML to see what you pasted.

Get me a sip trace of this from when it works until when it fails
only enable the sip trace on the profile with the gateway to reduce traffic



On Fri, Oct 29, 2010 at 1:56 PM, Mario G <mario_fs at mgtech.com> wrote:
> Oh my... ?looks it was not NAT after all? Please help! I changed to all profiles to static per instructions below and still have the problem:
> 2010-10-29 11:15:18.536446 [NOTICE] sofia_reg.c:342 Registering uuid1
> 2010-10-29 11:15:34.313150 [WARNING] sofia_reg.c:387 uuid1 Failed Registration, setting retry to 15 seconds.
>
> sofia global siptrace on did not show any activity for this gateway in or out, others were fine but eventually fail. I setup static:
> 1. set the params ext-sip-ip and ext-rtp-ip to my external static IP
> 2. map the sip ports (5060-5080) and all of the rtp ports (16384-32767) ?to FS lan addr.
> 3. set sip-ip and rtp-ip to the lan addr of FS
> 4. start FS with -nonat
>
> I don't know what to try next. BTW, the sofia status for the profiles shows stun enabled but I did not set it up anywhere:
>
> Name ? ? ? ? ? ? ? ? ? ?uuid1
> Domain Name ? ? ? ? ? ? N/A
> Auto-NAT ? ? ? ? ? ? ? ?false
> DBName ? ? ? ? ? ? ? ? ?sofia_reg_mvvyl
> Pres Hosts
> Dialplan ? ? ? ? ? ? ? ?XML
> Context ? ? ? ? ? ? ? ? public
> Challenge Realm ? ? ? ? auto_to
> RTP-IP ? ? ? ? ? ? ? ? ?10.x.x.20
> Ext-RTP-IP ? ? ? ? ? ? ?210.x.x.100
> SIP-IP ? ? ? ? ? ? ? ? ?10.x.x.20
> Ext-SIP-IP ? ? ? ? ? ? ?210.x.x.100
> URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 210.x.x.100:5068
> BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 210.x.x.100:5068;maddr=10.x.x.20
> HOLD-MUSIC ? ? ? ? ? ? ?local_stream://moh
> OUTBOUND-PROXY ? ? ? ? ?N/A
> CODECS IN ? ? ? ? ? ? ? G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
> CODECS OUT ? ? ? ? ? ? ?PCMU,PCMA,GSM
> TEL-EVENT ? ? ? ? ? ? ? 101
> DTMF-MODE ? ? ? ? ? ? ? rfc2833
> CNG ? ? ? ? ? ? ? ? ? ? 13
> SESSION-TO ? ? ? ? ? ? ?0
> MAX-DIALOG ? ? ? ? ? ? ?0
> NOMEDIA ? ? ? ? ? ? ? ? false
> LATE-NEG ? ? ? ? ? ? ? ?false
> PROXY-MEDIA ? ? ? ? ? ? false
> AGGRESSIVENAT ? ? ? ? ? false
> STUN-ENABLED ? ? ? ? ? ?true
> STUN-AUTO-DISABLE ? ? ? false
> CALLS-IN ? ? ? ? ? ? ? ?0
> FAILED-CALLS-IN ? ? ? ? 0
> CALLS-OUT ? ? ? ? ? ? ? 0
> FAILED-CALLS-OUT ? ? ? ?0
>
>
> On Oct 27, 2010, at 10:04 AM, Anthony Minessale wrote:
>
>> if you map it or not, a scanner would penetrate it.
>> There are lot of sip scanners out there now, you just need to beware of them.
>>
>>
>> On Wed, Oct 27, 2010 at 11:50 AM, Mario G <mario_fs at mgtech.com> wrote:
>>> Thanks so much! I am sure many others will find this info invaluable. I will try the static route again but have one question: When I started with FS I found a "sip scanner" ?in FS and someone on this group said not to use port mapping since it was a security risk. Is that true?
>>>
>>> On Oct 27, 2010, at 9:10 AM, Anthony Minessale wrote:
>>>
>>>> you are completely guessing at things.
>>>> I want you to understand that the only reason you are having problems
>>>> with this is because you don't understand how it works enough to know
>>>> what you are doing 100%
>>>>
>>>> Its a given that the pnp stuff is only for your dynamic IP.
>>>> aggressive-nat-detection and sip-force-expires are all related to
>>>> inbound calls when the things who are registering to you may be behind
>>>> nat.
>>>>
>>>> You need to learn the difference between which nat tools are
>>>> *) designed for your FS to run behind nat
>>>> *) designed for FS to run public and accept connections from devices behind nat.
>>>>
>>>> If you have a static IP, you don't need the pnp stuff so -nonat is fine
>>>> What you need to do is set
>
>>>> 1) set the params ext-sip-ip and ext-rtp-ip to your external static IP
>>>> 2) map the sip ports and all of the rtp ports from your static IP to FS lan addr
>>>> 3) set sip-ip and rtp-ip to the lan addr you forwarded through.
>>>
>>>>
>>>> If you don't do this: your outbound registration will use NAT to your
>>>> provider and if there is no activity for the expire time on your NAT
>>>> mapping the reverse port mapping from your provider back to you is
>>>> lost. ?This is why you set your register expires to a very low number,
>>>> (you need to make sure the provider does not turn the expires back up
>>>> in the reply because it will beat your choice *see sip trace) if this
>>>> is the case then you need the "ping" option set to 30, to continuously
>>>> send an options to your provider.
>>>>
>>>> The static mapping is obviously the better, easier and more reliable solution.
>>>>
>>>> So I want you to understand that the only way to keep a nat mapped
>>>> port alive is to continuously send traffic, all the other methods that
>>>> you are mentioning are to detect that phones registered to your are
>>>> behind nat, I gave you that force-expires option before because your
>>>> trace was full of inbound reg so I thought that is what you wanted
>>>> help with.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Wed, Oct 27, 2010 at 10:43 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>>> I should mention that I did not have this problem with an SPA9000 PBX
>>>>> (asterisk based) for over two years so FS may be pickier about upnp and/or
>>>>> nat, or just better at it exposing a problem in the router.
>>>>> I made different changes to the gateways to test different things. One
>>>>> failed after 17 hours, the other two stayed up. ?What did not work:
>>>>> added <variable name="sip-force-expires" value="30"/> to the directory
>>>>> entries as suggested.
>>>>> set the gateway expire times to 30 seconds.
>>>>> What worked (could be coincidental) for the two gateways that stayed up:
>>>>> I Added <param name="aggressive-nat-detection" value="true"/>
>>>>> I originally setup FS to use the static ip by setting external sip/rtp to
>>>>> just the static ip (no autonat:) and ran with -nonat but I could not get
>>>>> incoming calls. The only way it worked was to use autonat:1.2.3.4. The
>>>>> router has 1 static public address and 1 dynamic external IP, this is the
>>>>> root of the problem, upnp only tells FS about the dynamic ip ?Will keep this
>>>>> thread up-to-date for anyone who may be in the same boat someday. Thanks
>>>>> again for looking at the trace.
>>>>> Mario
>>>>>
>>>>> You should be setting the req freq to a low number on the outbound gateways
>>>>>
>>>>> The examples you showed had a series of inbound reg
>>>>>
>>>>> also set expire-seconds to 30 in your gateway xml
>>>>>
>>>>>
>>>>> The problem is if you are not constantly sending traffic to the box
>>>>>
>>>>> the nat mapping will go away.
>>>>>
>>>>> If you are in production you should be using a static ip with a static
>>>>>
>>>>> mapping, any trouble you are having is your own fault for playing with
>>>>>
>>>>> fire. ?The best we can do is tell you how to keep it contained.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>
>>>>> I made the change. I had no idea the settings for the inside phones effected
>>>>> nat for the outside sip accounts. I was looking into aggressive-nat-
>>>>> detection since the internal profile status always shows the right external
>>>>> static IP but the nat_ap status always shows the dynamic ip. Crossing
>>>>> fingers/etc since this problem is 85% of time (weeks!) into FS changeover.
>>>>> Thanks!
>>>>>
>>>>> Mario
>>>>>
>>>>> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
>>>>>
>>>>> add
>>>>>
>>>>> <variable name="sip-force-expires" value="30"/>
>>>>>
>>>>> to the <variables> section of your <user>
>>>>>
>>>>> you have it at 600 and the nat mapping is timing out while the 600
>>>>>
>>>>> seconds is ticking away
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>
>>>>> From the TSP:
>>>>>
>>>>> "I have enabled the SIP trace on your account. We are not currently seeing
>>>>>
>>>>> any registration attempts to your account within the last 15 minutes. Please
>>>>>
>>>>> restart FreeSwitch so that registration attempts begin again. Thank you. ".
>>>>>
>>>>> So FS is not getting past router.
>>>>>
>>>>> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>>>>>
>>>>> I ran the global trace during the problem and it is
>>>>>
>>>>> at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
>>>>>
>>>>> "accttwo", "acct3". The trace includes phones since it was global. I am
>>>>>
>>>>> using:
>>>>>
>>>>> ? ?<param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>>>>>
>>>>> ? ?<param name="ext-sip-ip" value="autonat:my-static.ip"/>
>>>>>
>>>>> I tried dumping nat and removing the autonat: above and using -nonat but
>>>>>
>>>>> that did not work, registration proceeded but no calls inbound.
>>>>>
>>>>> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>>>>>
>>>>> Whoops, I am using an IP address for at least one gateway so that is not the
>>>>>
>>>>> problem:
>>>>>
>>>>> They look outbound to me and I am using dns for 2 and an IP for one so that
>>>>>
>>>>> is not the issue. I was able to get FS to clear this up by doing "nat_map
>>>>>
>>>>> reinit" which is why I think this is a nat problem. I will do the trace you
>>>>>
>>>>> mentioned. I will plug an ip address into one of the gateways to see what
>>>>>
>>>>> happens, they all fail at once. Thanks for responding!
>>>>>
>>>>> Mario
>>>>>
>>>>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>>>>>
>>>>> I really need help on this as I have weeks into this problem. I thought I
>>>>>
>>>>> had it nailed but I guess not. After 5.5 hours I get:
>>>>>
>>>>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>>>>>
>>>>> Registration, setting retry to 15 seconds.
>>>>>
>>>>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>>>
>>>>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>>>
>>>>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
>>>>>
>>>>> Registration, setting retry to 30 seconds.
>>>>>
>>>>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>>>>>
>>>>> Registration, setting retry to 15 seconds.
>>>>>
>>>>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>>>>>
>>>>> Registration, setting retry to 15 seconds.
>>>>>
>>>>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>>>
>>>>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>>>
>>>>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>>>>>
>>>>> and no way to make/get calls until I restart FS. I did this:
>>>>>
>>>>> 1. log 7
>>>>>
>>>>> 2. sofia profile xxxx siptrace on ? for each profile/gateway
>>>>>
>>>>> 3. restarted router
>>>>>
>>>>> All three did not solve the problem. The trace and log produced no
>>>>>
>>>>> additional lines which is why I am wondering if FS has a problem since the
>>>>>
>>>>> trace shows no SIP activity.
>>>>>
>>>>> 3 gateways with 2 ITSPs
>>>>>
>>>>> 2 DSL/WAN lines, 1 static and 1 dynamic
>>>>>
>>>>> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
>>>>>
>>>>> external static ip.
>>>>>
>>>>> sofia status profile ... has the right ext ip
>>>>>
>>>>> nat_map status shows the dynamic (wrong) IP
>>>>>
>>>>> I tried starting with -nonat but that was worse
>>>>>
>>>>> the only way to fix is restart FS.
>>>>>
>>>>> I read the wiki on external nat, auto_nat and everything else many times.
>>>>>
>>>>> Thanks Mario
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>
>>>>> http://www.freeswitch.org
>>>>>
>>>>> _______________________________________________
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>
>>>>> 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
>>>>>
>>>>> _______________________________________________
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>
>>>>> 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
>>>>>
>>>>> _______________________________________________
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> 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
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From steveayre at gmail.com  Fri Oct 29 12:51:17 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 29 Oct 2010 20:51:17 +0100
Subject: [Freeswitch-users] mod_h323 h323plus debian shlibdeps dependancy
	problem
Message-ID: <AANLkTimFWVgeDG-U1MkJvZ277jn9phddjSOOyOUyKR4A@mail.gmail.com>

I have install ptlib 2.8.2 and h323plus trunk as per the instructions
on the mod_h323 wiki page.

When I build the debian packages it finishes with this error:
dpkg-shlibdeps: failure: no dependency information found for
/usr/local/lib/libh323_linux_x86_64_.so.1.22.0 (used by
debian/freeswitch/opt/freeswitch/mod/mod_h323.so)

Does anyone know what causes this error, and how to fix it?

-Steve


From willbelair at yahoo.com  Fri Oct 29 13:01:51 2010
From: willbelair at yahoo.com (Will Smith)
Date: Fri, 29 Oct 2010 13:01:51 -0700 (PDT)
Subject: [Freeswitch-users] Mod Event Socket
In-Reply-To: <AANLkTi=6HgNczOBs110tX-ofAO50FbapWZ1P3ascMy0h@mail.gmail.com>
Message-ID: <900263.75973.qm@web55703.mail.re3.yahoo.com>

hi Michael,Thank you for your input. Really appreciate when having someone giving help when in need.I bought FS ebook, read chapter 9 and 10. And did what you said:1- put the event_socket_conf.xml back to original state, just modify the ip address, because I work through a vpn network. Here , could you confirm that mod_event_socket supports different browsers ? This is really important to me, because if client could not access from www, there is no point to work on this. Thank you.-------- the event_socket_conf.xml<configuration name="event_socket.conf" description="Socket Client">??<settings>?? ?<param name="nat-map" value="false"/>?? ?<param name="listen-ip" value="192.168.51.133"/>?? ?<param name="listen-port" value="8021"/>?? ?<param name="password" value="ClueCon"/>?? ?<!--param name="apply-inbound-acl" value="lan"/-->??</settings></configuration>-----------
2- I put this in both dialplan/default.xml ?and dialplan/public.xml?------------<extension name="outbound-socket">		<condition field="destination_number" expression="^444$">		<action application="socket" data="192.168.51.133:8021 async full"/>		</condition></extension>------------
3- I have a simple php script just to connect to the FS, this is a sample script from FS wiki page------<?php
//require_once('ESL.php');require_once('/usr/src/freeswitch/freeswitch-snapshot/libs/esl/php/ESL.php');set_time_limit(0); // Remove the PHP time limit of 30 seconds for completion due to loop watching eventsecho "HERE";// Connect to FreeSWITCH$sock = new ESLconnection('localhost', '8021', 'ClueCon');// We want all Events (probably will want to change this depending on your needs)$sock->sendRecv("event plain ALL");

// Grab Events until process is killed//while($sock->connected()){//echo "CONECTED";//$event = $sock->recvEvent();//print_r($event->serialize());//}
?>------
when running this from web browser, I got blank page. So, I run from console, this is the error messages:--------PHP Warning: ?dl(): Unable to load dynamic library '/usr/lib/php/modules/ESL.so' - /usr/lib/php/modules/ESL.so: cannot open shared object file: No such file or directory in /usr/src/freeswitch/freeswitch-snapshot/libs/esl/php/ESL.php on line 23PHP Stack trace:PHP ? 1. {main}() /var/www/html/eventsocket.php:0PHP ? 2. require_once() /var/www/html/eventsocket.php:4PHP ? 3. dl() /usr/src/freeswitch/freeswitch-snapshot/libs/esl/php/ESL.php:23HEREPHP Fatal error: ?Call to undefined function new_ESLconnection() in /usr/src/freeswitch/freeswitch-snapshot/libs/esl/php/ESL.php on line 119PHP Stack trace:PHP ? 1. {main}() /var/www/html/eventsocket.php:0PHP ? 2. ESLconnection->__construct() /var/www/html/eventsocket.php:8----------
So, I guess the ESL.so is missing. I tried to ?"locate ESL.so" , but nothing returned. I tried to?--------yum install libxml2-devel pcre-devel bzip2-devel curl-devel gmp-devel aspell-devel php-devel libtermcap-devel gdbm-devel db4-devel--------it ran fine. But I did not do this:-------NOTE:?For PHP you must edit the libs/esl/php/Makefile and add -lpthread to the LOCAL_LDFLAGS line.-------because I could not fine the LOCAL_LDFLAGS line.
Please give me some help, I am banging my head against the wall here. If I could get the file ESL.so , is the set up above working? I thought it would be simple just to connect.
Thank you.
--- On Thu, 10/28/10, Michael Collins <msc at freeswitch.org> wrote:

From: Michael Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] Mod Event Socket
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Received: Thursday, October 28, 2010, 5:57 PM

Be sure to learn the difference between inbound and outbound event socket. In your case you are doing outbound event socket. The dialplan calls the socket app which does an outbound socket connection to port 9090 on the localhost. Make sure that your app is listening on port 9090.


The event_socket.conf.xml file is for configuring inbound socket connections, e.g. when you have a script that attempts to connect to FS where FS is listening on port 8021. So change your event_socket.conf.xml file back to the default and retry. 


Report back here if you have trouble. Also, buy or borrow the FreeSWITCH book and check out chapter 9. Lots of good stuff there on how the even system works.

-MC

On Thu, Oct 28, 2010 at 2:14 PM, Will Smith <willbelair at yahoo.com> wrote:

Hi,I am new to FS, and trying to get the mod event socket installed and running. I have FS running, with SIP account, can dial in/out via gateway.
Now I want to dial in, FS will send some info to client browser, here is one question, does this work over the internet, or just local ? Info sent could be the uuid, so that client browser could decide to bridge the call, send to IVR or transfer ...
Please give me some guide line how to set this up.I added to dialplan/default.xml----- <extension name="socketTest">
      <condition field="destination_number" expression="^444$">
        <action application="socket" data="127.0.0.1:9090 async full"/>
      </condition>
 </extension>-----
got the php sample file: <?php

/**
 * Based loosely on the NET_Server code in PEAR.
 * This is only an example - considerable additional work is needed
 * Specifically, the code in the handleConnection method should be 
 * handled in a subclass
 *
 * 
 */

class Message
{
  var $properties = array();
  var $content = null;
}

class EventSocketListener
{

  var $host;
  var $port;
  var $sock;
  var $is_parent = true;
  var $clientInfo;
  var $clientFD;

  var $connectionContext = array();


  function &create($port, $host = "localhost")
      {
	$esl = new EventSocketListener;
	$esl->port = $port;
	$esl->host = $host;
        if (!function_exists('socket_create')) {
            return PEAR::raiseError('Sockets extension not available.');
        }
	return $esl;
      }

  function start()
  {
	if (($this->sock = socket_create(AF_INET, SOCK_STREAM, SOL_TCP)) === false) {
	  echo "socket_create() failed: reason: " . socket_strerror(socket_last_error()) . "\n";
	}

	if (!socket_set_option($this->sock, SOL_SOCKET, SO_REUSEADDR, 1)) {
	  echo 'Unable to set option on socket: '. socket_strerror(socket_last_error()) . PHP_EOL;
	}

	if (socket_bind($this->sock, $this->host, $this->port) === false) {
	  echo "socket_bind() failed: reason: " . socket_strerror(socket_last_error($this->sock)) . "\n";
	}

	if (socket_listen($this->sock, 5) === false) {
	  echo "socket_listen() failed: reason: " . socket_strerror(socket_last_error($this->sock)) . "\n";
	}

        // Dear children, please do not become zombies
        pcntl_signal(SIGCHLD, SIG_IGN);
        
        // wait for incmoning connections
        while (true)
        {
            // new connection
            if(($fd = socket_accept($this->sock)))
            {
	      $pid = pcntl_fork();
	      if($pid == -1) {
		  return  PEAR::raiseError('Could not fork child process.');
                }
                // This is the child => handle the request
                elseif($pid == 0) {
                    // this is not the parent
                    $this->_isParent = false;
                    // store the new file descriptor
                    $this->clientFD = $fd;

                    $peer_host    =    "";
                    $peer_port    =    "";
                    socket_getpeername($this->clientFD, $peer_host, $peer_port);
                    $this->clientInfo    =    array(
                                                      "host"        =>    $peer_host,
                                                      "port"        =>    $peer_port,
                                                      "connectOn"   =>    time()
                                                   );
		    $this->handleConnection();
		    socket_shutdown($this->clientFD, 2);
		    socket_close($this->clientFD);
		}
		else /* Parent does nothing */
		  {
		  }
	    }
	}
  }

  function handleConnection()
  {
    $fd = $this->clientFD;
    //first, read headers & setup a state for this connection
    $line = "";
    socket_write($fd, "CONNECT\n\n");
    do 
      {
	$line = socket_read($fd, 2048, PHP_NORMAL_READ);
	if (trim($line) == "")
	  break;
	//we got a header, we need to add it to the 
	list($key, $value) = explode(":", $line);
	$key = trim($key);
	$value = trim(urldecode($value));
	$this->connectionContext[$key] = $value;
      }
    while ($line != "");
    
    //    print_r($this->connectionContext);
    $this->callConnected();

    exit();
    
  }

  function processMessages($returnOnReply = false)
  {
    $fd = $this->clientFD;
    $result = new Message();
    $props = array();
    while (true)
      {
	do 
	  {
	    $line = @socket_read($fd, 2048, PHP_NORMAL_READ);
	    if (socket_last_error($fd) == 104)
	      return null;
	    if ($line == null || $line == FALSE || trim($line) == "")
	      break;
	    //we got a header, we need to add it to the message
	    list($key, $value) = explode(":", $line);
	    $key = trim($key);
	    $value = trim(urldecode($value));
	    $props[$key] = $value;
	  }
	while ($line != "");
	$result->properties = $props;

	if (isset($props['Content-Length']))
	  {
	    $length = $props['Content-Length'];
	    print("Reading content - $length\n");
	    $data = socket_read($fd, $length);
	    $result->content = $data;
	  }
	if (isset($props['Content-Type']))
	  {
	    $type = $props['Content-Type'];
	    if ($returnOnReply && 
		($type == "command/reply" || $type == "api/response"))
	      {
		return $result;
	      }
	    else if ($type == "text/event-plain") //only plain events for now
	      {
		$this->handleEvent($result);
	      }
	  }
	else
	  {
	    print("UNKNOWN MESSAGE: \n");
	    print_r($result);
	  }
      }    
  }
    


  function invokeCommand($command)
  {
    //Send the command
    print("Invoking: $command\n");
    $this->sendCommand($command);
    // Wait for the response
    $result = $this->processMessages(true);
    return $result;
  }

  function sendCommand($command)
  {
    $fd = $this->clientFD;
    socket_write($fd, trim($command) . "\n\n"); 
  }

/*-----------------------------------------------------*/
  /* Abstract Methods - should move to subclass*/

  function callConnected()
  {
    print_r($this->connectionContext);
    print("----------------\n");
    $result = $this->invokeCommand("log DEBUG");
    print_r($result);
    $result = $this->invokeCommand("event plain ALL");
    print_r($result);

    $this->processMessages(false);
    print("DONE PROCESSING MESSAGES");
    print_r($this->connectionContext);
  }

  function handleCommandResponse($response)
  {
    print("Recieved Unhandled Response:\n");
    print_r($response);
  }

  function handleEvent($event)
  {
    print("Recieved Unhandled Event:\n");
    print_r($event);
  }
}

  

  
// create a server that forks new processes
$server  = &EventSocketListener::create(9090);

// start the server
$server->start();
?>--------( this is the original file, not perfect sample) I tried to run this, and got error with auth.
Also, I modify the even_socket_conf.xml  in autoload_configs/  change listen-ip to  0.0.0.0 , port = 9090 , disable password

What did I miss? Thankyou for your help
Will


_______________________________________________


FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org





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From freeswitch at peely.com  Fri Oct 29 13:27:48 2010
From: freeswitch at peely.com (peely)
Date: Fri, 29 Oct 2010 13:27:48 -0700 (PDT)
Subject: [Freeswitch-users] Problem with T.38 fax receive using ESL
In-Reply-To: <1288358483496-5686430.post@n2.nabble.com>
References: <1288272464502-5682561.post@n2.nabble.com>
	<AANLkTinzujxqm7gT8k20+yeSKR+nYSRzjmdUSg_BGLZT@mail.gmail.com>
	<1288358483496-5686430.post@n2.nabble.com>
Message-ID: <1288384068806-5687937.post@n2.nabble.com>


Hi,

OK, I set those specifically using SendMsg with the set application, but
still nothing happened.

After some bashing around though I've figured that the behaviour is slightly
different in esl compared to the dialplan.

It seems that you MUST not issue an answer followed by the broadcast of
silence, instead just call rxfax and let it answer the incoming call, if you
don't do this then a T.38 session is never reinvited.

It's working perfectly now, thank-you!



Regards,



Neil.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-T-38-fax-receive-using-ESL-tp5682561p5687937.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Fri Oct 29 13:39:09 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 29 Oct 2010 15:39:09 -0500
Subject: [Freeswitch-users] Loss of first second of media
In-Reply-To: <AANLkTinrSNUr49b3LE6c6ZP7NZ9d13t1CK2bPW2x+anD@mail.gmail.com>
References: <AANLkTimECwtVmnbFsBJRiN3YSaGyXbvh-sKtKYPey8xx@mail.gmail.com>
	<AANLkTik5H74MupY8iyx6Ao4=Rv=wCm8jv-w=xCtSnm0Z@mail.gmail.com>
	<AANLkTikpVH-s6AvJ7fQoZu2PE4dduTg+d-ygyBWSfr4R@mail.gmail.com>
	<AANLkTi=FZDqev9R1aTMpdvTk8t9NfxiyeisQSRBxfTEb@mail.gmail.com>
	<AANLkTikY68ovuiSO9xkbZZPSskxCNsjrpTSdb-42bwJB@mail.gmail.com>
	<AANLkTikiG3xPKDVkwU4GDk5t6KDe67AQmM-dkheMCSu7@mail.gmail.com>
	<AANLkTinJRcpRx_BQyKifECcurYb+vG2P8Nct1xT_X3XO@mail.gmail.com>
	<AANLkTinpaRivNHteDp4i6WJjmgsPAE6otVx5LjgJcvWr@mail.gmail.com>
	<AANLkTikiVxFAOwftrmfqd3LtD4H__UUytsM117WBhZy0@mail.gmail.com>
	<AANLkTikLtKhBiCZeMY+GH6X7jb4wVdybVF=OiMePK8Rm@mail.gmail.com>
	<AANLkTimyRrtba8nD7jkWJY2DysebRaycpvRXY3b_A3r-@mail.gmail.com>
	<AANLkTinrSNUr49b3LE6c6ZP7NZ9d13t1CK2bPW2x+anD@mail.gmail.com>
Message-ID: <AANLkTi=HOiA38khKujHo4y=oP90OKP4Pcmd-L5m70P7K@mail.gmail.com>

OK so,
The phone sends a 180 ringing with NO SDP
then it starts sending RTP
That's is not right.  It's a bug in the phone.



On Fri, Oct 29, 2010 at 1:35 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> can you try another one with just udp and not "port 5060"
> so I can see the rtp too
>
>
> On Fri, Oct 29, 2010 at 12:28 PM, Fraser Redmond
> <fraserredmond at gmail.com> wrote:
>> Thanks Anthony, it's here:
>> http://pastebin.freeswitch.org/14350
>>
>> And pcap is attached.
>>
>> The call connects around (or just before) the 16:58:35 mark (line 558 is
>> what I see in the terminal while waiting for it to connect - both
>> early-media and the missing start of the media)
>>
>> Cheers,
>> Fraser
>>
>>
>>
>>
>> On Thu, Oct 28, 2010 at 3:59 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>>
>>> Can you do this trace with debug level logging in addition to the sip
>>> trace
>>> console loglevel debug
>>>
>>> you also may want to get a pcap of it
>>>
>>> tshark udp and port 5060 -w test.pcap
>>>
>>>
>>>
>>> On Sat, Oct 23, 2010 at 11:44 AM, Fraser Redmond
>>> <fraserredmond at gmail.com> wrote:
>>> > Thanks Anthony,
>>> >
>>> > Finally managed to get a sip trace - could you do me a favor and take a
>>> > look
>>> > and/or give me some ideas of what to look for?
>>> >
>>> > http://pastebin.freeswitch.org/14300
>>> >
>>> > I've highlighted lines 168 and 193. In between these lines is where the
>>> > number is dialed and rings once, then picks up, then theres silence for
>>> > a
>>> > second or two, and that second SIP message is when I start hearing
>>> > audio.
>>> >
>>> > Thanks,
>>> > Fraser
>>> >
>>> >
>>> >
>>> >
>>> > On Thu, Oct 21, 2010 at 6:44 PM, Anthony Minessale
>>> > <anthony.minessale at gmail.com> wrote:
>>> >>
>>> >> its a blue message on cli
>>> >>
>>> >> It could also be the other side expecting us to send media first or
>>> >> something silly.
>>> >> try getting a sip trace and figure out when the rtp starts arriving.
>>> >>
>>> >>
>>> >> On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond
>>> >> <fraserredmond at gmail.com> wrote:
>>> >> > Sorry, yes, I am setting ignore_early_media=true in the first area.
>>> >> > (Or
>>> >> > are
>>> >> > you saying that should be off? I forget now why I needed it on, but
>>> >> > there
>>> >> > was a reason I added it.)
>>> >> >
>>> >> > Yes, the bridge doesn't start until after the A-leg has answered.
>>> >> >
>>> >> > Thanks for the suggestion about nat/auto-changing port, I'll have a
>>> >> > look
>>> >> > into that - would that be in the cli output or in a sip trace? I've
>>> >> > already
>>> >> > looked and it's not appearing in the CLI output (with
>>> >> > loglevel=debug),
>>> >> > haven't looked in the sip trace yet.
>>> >> >
>>> >> > Cheers,
>>> >> > Fraser
>>> >> >
>>> >> >
>>> >> >
>>> >> >
>>> >> > On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale
>>> >> > <anthony.minessale at gmail.com> wrote:
>>> >> >>
>>> >> >> are you setting ignore_early_media=true in the first vars=values
>>> >> >> area?
>>> >> >>
>>> >> >> This looks like you could be calling one leg who is still not
>>> >> >> answered
>>> >> >> and then bridging it to another dest. ?The bridge app will wait for
>>> >> >> the first leg to answer before bridging.
>>> >> >>
>>> >> >> Also if you have any NAT anywhere, look for an "auto-changing port"
>>> >> >> type message which can also be attributed to this due to a detection
>>> >> >> period for incorrect ports.
>>> >> >>
>>> >> >>
>>> >> >>
>>> >> >> On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond
>>> >> >> <fraserredmond at gmail.com> wrote:
>>> >> >> > event_socket:
>>> >> >> > api originate {vars=values}user/$fromExtn at Domain
>>> >> >> > 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline
>>> >> >> >
>>> >> >> > then
>>> >> >> >
>>> >> >> > dialplan:
>>> >> >> > <action application="set"
>>> >> >> > data="effective_caller_id_number=+1800number"/>
>>> >> >> > <action application="set" data="effective_caller_id_name="/>
>>> >> >> > (set and/or export a bunch of other vars too)
>>> >> >> > <action application="set"
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> > data="dial_string=sofia/gateway/gatewayname.com/00${destination_number}"/>
>>> >> >> > <action application="bridge" data="${dial_string}"/>
>>> >> >> >
>>> >> >> > Cheers,
>>> >> >> > Fraser
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> > On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale
>>> >> >> > <anthony.minessale at gmail.com> wrote:
>>> >> >> >>
>>> >> >> >> how are you accomplishing that? by which technique?
>>> >> >> >>
>>> >> >> >> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond
>>> >> >> >> <fraserredmond at gmail.com> wrote:
>>> >> >> >> > The call is originated from Freeswitch (via CLI) to a
>>> >> >> >> > softphone,
>>> >> >> >> > then
>>> >> >> >> > when
>>> >> >> >> > that is connected it bridges out to the gateway.
>>> >> >> >> >
>>> >> >> >> > Cheers,
>>> >> >> >> > Fraser
>>> >> >> >> >
>>> >> >> >> >
>>> >> >> >> >
>>> >> >> >> >
>>> >> >> >> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale
>>> >> >> >> > <anthony.minessale at gmail.com> wrote:
>>> >> >> >> >>
>>> >> >> >> >> Where is the other side of this call coming from?
>>> >> >> >> >>
>>> >> >> >> >> [ ( ? ) ] -> FS -> (PSTN via SIP)
>>> >> >> >> >>
>>> >> >> >> >> What goes in the empty space above?
>>> >> >> >
>>> >> >> >
>>> >> >> > _______________________________________________
>>> >> >> > FreeSWITCH-users mailing list
>>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >
>>> >> >> >
>>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> > 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
>>> >> >>
>>> >> >> _______________________________________________
>>> >> >> FreeSWITCH-users mailing list
>>> >> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >>
>>> >> >>
>>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> http://www.freeswitch.org
>>> >> >
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >
>>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > 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
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > 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
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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
>



-- 
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


From willbelair at yahoo.com  Fri Oct 29 14:03:29 2010
From: willbelair at yahoo.com (Will Smith)
Date: Fri, 29 Oct 2010 14:03:29 -0700 (PDT)
Subject: [Freeswitch-users] Mod Event Socket
In-Reply-To: <900263.75973.qm@web55703.mail.re3.yahoo.com>
Message-ID: <748154.86998.qm@web55708.mail.re3.yahoo.com>

hi Michael,
I fixed the ESL.so prob, and run the ?test.php, I got response.Next step is running the php script from browser. Hope that I can do that. Thank you for your help. If you have any tip, or samples, and have free time, please give.
Thanks,Will

--- On Fri, 10/29/10, Will Smith <willbelair at yahoo.com> wrote:

From: Will Smith <willbelair at yahoo.com>
Subject: Re: [Freeswitch-users] Mod Event Socket
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Received: Friday, October 29, 2010, 1:01 PM

hi Michael,Thank you for your input. Really appreciate when having someone giving help when in need.I bought FS ebook, read chapter 9 and 10. And did what you said:1- put the event_socket_conf.xml back to original state, just modify the ip address, because I work through a vpn network. Here , could you confirm that mod_event_socket supports different browsers ? This is really important to me, because if client could not access from www, there is no point to work on this. Thank you.-------- the
 event_socket_conf.xml<configuration name="event_socket.conf" description="Socket Client">??<settings>?? ?<param name="nat-map" value="false"/>?? ?<param name="listen-ip" value="192.168.51.133"/>?? ?<param name="listen-port" value="8021"/>?? ?<param name="password" value="ClueCon"/>?? ?<!--param name="apply-inbound-acl" value="lan"/-->??</settings></configuration>-----------
2- I put this in both dialplan/default.xml ?and dialplan/public.xml?------------<extension name="outbound-socket">		<condition field="destination_number" expression="^444$">		<action application="socket" data="192.168.51.133:8021 async full"/>		</condition></extension>------------
3- I have a simple php script just to connect to the FS, this is a sample script from FS wiki page------<?php
//require_once('ESL.php');require_once('/usr/src/freeswitch/freeswitch-snapshot/libs/esl/php/ESL.php');set_time_limit(0); // Remove the PHP time limit of 30 seconds for completion due to loop watching eventsecho
 "HERE";// Connect to FreeSWITCH$sock = new ESLconnection('localhost', '8021', 'ClueCon');// We want all Events (probably will want to change this depending on your needs)$sock->sendRecv("event plain ALL");

// Grab Events until process is killed//while($sock->connected()){//echo "CONECTED";//$event = $sock->recvEvent();//print_r($event->serialize());//}
?>------
when running this from web browser, I got blank page. So, I run from console, this is the error messages:--------PHP Warning: ?dl(): Unable to load dynamic library '/usr/lib/php/modules/ESL.so' - /usr/lib/php/modules/ESL.so: cannot open shared
 object file: No such file or directory in /usr/src/freeswitch/freeswitch-snapshot/libs/esl/php/ESL.php on line 23PHP Stack trace:PHP ? 1. {main}() /var/www/html/eventsocket.php:0PHP ? 2. require_once() /var/www/html/eventsocket.php:4PHP ? 3. dl() /usr/src/freeswitch/freeswitch-snapshot/libs/esl/php/ESL.php:23HEREPHP Fatal error: ?Call to undefined function new_ESLconnection() in /usr/src/freeswitch/freeswitch-snapshot/libs/esl/php/ESL.php on line 119PHP Stack trace:PHP ? 1. {main}() /var/www/html/eventsocket.php:0PHP ? 2. ESLconnection->__construct() /var/www/html/eventsocket.php:8----------
So, I guess
 the ESL.so is missing. I tried to ?"locate ESL.so" , but nothing returned. I tried to?--------yum install libxml2-devel pcre-devel bzip2-devel curl-devel gmp-devel aspell-devel php-devel libtermcap-devel gdbm-devel
 db4-devel--------it ran fine. But I did not do this:-------NOTE:?For PHP you must edit the libs/esl/php/Makefile and add -lpthread to the LOCAL_LDFLAGS line.-------because I could not fine the LOCAL_LDFLAGS line.
Please give me some help, I am banging my head against the wall here. If I could get the file ESL.so , is the set up above working? I thought it would be simple just to connect.
Thank you.
--- On Thu, 10/28/10, Michael Collins <msc at freeswitch.org> wrote:

From: Michael Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] Mod Event Socket
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Received: Thursday, October 28, 2010, 5:57 PM

Be sure to learn the difference between inbound and outbound event socket. In your case you are doing outbound event socket. The dialplan calls the socket app which does an outbound socket connection to port 9090 on the localhost. Make sure that your app is listening on port 9090.


The event_socket.conf.xml file is for configuring inbound socket connections, e.g. when you have a script that attempts to connect to FS where FS is listening on port 8021. So change your event_socket.conf.xml file back to the default and retry. 


Report back here if you have trouble. Also, buy or borrow the FreeSWITCH book and check out chapter 9. Lots of good stuff there on how the even system works.

-MC

On Thu, Oct 28, 2010 at 2:14 PM, Will Smith <willbelair at yahoo.com> wrote:

Hi,I am new to FS, and trying to get the mod event socket installed and running. I have FS running, with SIP account, can dial in/out via gateway.
Now I want to dial in, FS will send some info to client browser, here is one question, does this work over the internet, or just local ? Info sent could be the uuid, so that client browser could decide to bridge the call, send to IVR or transfer ...
Please give me some guide line how to set this up.I added to dialplan/default.xml----- <extension name="socketTest">
      <condition field="destination_number" expression="^444$">
        <action application="socket" data="127.0.0.1:9090 async full"/>
      </condition>
 </extension>-----
got the php sample file: <?php

/**
 * Based loosely on the NET_Server code in PEAR.
 * This is only an example - considerable additional work is needed
 * Specifically, the code in the handleConnection method should be 
 * handled in a subclass
 *
 * 
 */

class Message
{
  var $properties = array();
  var $content = null;
}

class EventSocketListener
{

  var $host;
  var $port;
  var $sock;
  var $is_parent = true;
  var $clientInfo;
  var $clientFD;

  var $connectionContext = array();


  function &create($port, $host = "localhost")
      {
	$esl = new EventSocketListener;
	$esl->port = $port;
	$esl->host = $host;
        if (!function_exists('socket_create')) {
            return PEAR::raiseError('Sockets extension not available.');
        }
	return $esl;
      }

  function start()
  {
	if (($this->sock = socket_create(AF_INET, SOCK_STREAM, SOL_TCP)) === false) {
	  echo "socket_create() failed: reason: " . socket_strerror(socket_last_error()) . "\n";
	}

	if (!socket_set_option($this->sock, SOL_SOCKET, SO_REUSEADDR, 1)) {
	  echo 'Unable to set option on socket: '. socket_strerror(socket_last_error()) . PHP_EOL;
	}

	if (socket_bind($this->sock, $this->host, $this->port) === false) {
	  echo "socket_bind() failed: reason: " . socket_strerror(socket_last_error($this->sock)) . "\n";
	}

	if (socket_listen($this->sock, 5) === false) {
	  echo "socket_listen() failed: reason: " . socket_strerror(socket_last_error($this->sock)) . "\n";
	}

        // Dear children, please do not become zombies
        pcntl_signal(SIGCHLD, SIG_IGN);
        
        // wait for incmoning connections
        while (true)
        {
            // new connection
            if(($fd = socket_accept($this->sock)))
            {
	      $pid = pcntl_fork();
	      if($pid == -1) {
		  return  PEAR::raiseError('Could not fork child process.');
                }
                // This is the child => handle the request
                elseif($pid == 0) {
                    // this is not the parent
                    $this->_isParent = false;
                    // store the new file descriptor
                    $this->clientFD = $fd;

                    $peer_host    =    "";
                    $peer_port    =    "";
                    socket_getpeername($this->clientFD, $peer_host, $peer_port);
                    $this->clientInfo    =    array(
                                                      "host"        =>    $peer_host,
                                                      "port"        =>    $peer_port,
                                                      "connectOn"   =>    time()
                                                   );
		    $this->handleConnection();
		    socket_shutdown($this->clientFD, 2);
		    socket_close($this->clientFD);
		}
		else /* Parent does nothing */
		  {
		  }
	    }
	}
  }

  function handleConnection()
  {
    $fd = $this->clientFD;
    //first, read headers & setup a state for this connection
    $line = "";
    socket_write($fd, "CONNECT\n\n");
    do 
      {
	$line = socket_read($fd, 2048, PHP_NORMAL_READ);
	if (trim($line) == "")
	  break;
	//we got a header, we need to add it to the 
	list($key, $value) = explode(":", $line);
	$key = trim($key);
	$value = trim(urldecode($value));
	$this->connectionContext[$key] = $value;
      }
    while ($line != "");
    
    //    print_r($this->connectionContext);
    $this->callConnected();

    exit();
    
  }

  function processMessages($returnOnReply = false)
  {
    $fd = $this->clientFD;
    $result = new Message();
    $props = array();
    while (true)
      {
	do 
	  {
	    $line = @socket_read($fd, 2048, PHP_NORMAL_READ);
	    if (socket_last_error($fd) == 104)
	      return null;
	    if ($line == null || $line == FALSE || trim($line) == "")
	      break;
	    //we got a header, we need to add it to the message
	    list($key, $value) = explode(":", $line);
	    $key = trim($key);
	    $value = trim(urldecode($value));
	    $props[$key] = $value;
	  }
	while ($line != "");
	$result->properties = $props;

	if (isset($props['Content-Length']))
	  {
	    $length = $props['Content-Length'];
	    print("Reading content - $length\n");
	    $data = socket_read($fd, $length);
	    $result->content = $data;
	  }
	if (isset($props['Content-Type']))
	  {
	    $type = $props['Content-Type'];
	    if ($returnOnReply && 
		($type == "command/reply" || $type == "api/response"))
	      {
		return $result;
	      }
	    else if ($type == "text/event-plain") //only plain events for now
	      {
		$this->handleEvent($result);
	      }
	  }
	else
	  {
	    print("UNKNOWN MESSAGE: \n");
	    print_r($result);
	  }
      }    
  }
    


  function invokeCommand($command)
  {
    //Send the command
    print("Invoking: $command\n");
    $this->sendCommand($command);
    // Wait for the response
    $result = $this->processMessages(true);
    return $result;
  }

  function sendCommand($command)
  {
    $fd = $this->clientFD;
    socket_write($fd, trim($command) . "\n\n"); 
  }

/*-----------------------------------------------------*/
  /* Abstract Methods - should move to subclass*/

  function callConnected()
  {
    print_r($this->connectionContext);
    print("----------------\n");
    $result = $this->invokeCommand("log DEBUG");
    print_r($result);
    $result = $this->invokeCommand("event plain ALL");
    print_r($result);

    $this->processMessages(false);
    print("DONE PROCESSING MESSAGES");
    print_r($this->connectionContext);
  }

  function handleCommandResponse($response)
  {
    print("Recieved Unhandled Response:\n");
    print_r($response);
  }

  function handleEvent($event)
  {
    print("Recieved Unhandled Event:\n");
    print_r($event);
  }
}

  

  
// create a server that forks new processes
$server  = &EventSocketListener::create(9090);

// start the server
$server->start();
?>--------( this is the original file, not perfect sample) I tried to run this, and got error with auth.
Also, I modify the even_socket_conf.xml  in autoload_configs/  change listen-ip to  0.0.0.0 , port = 9090 , disable password

What did I miss? Thankyou for your help
Will


_______________________________________________


FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org





-----Inline Attachment Follows-----

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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-----Inline Attachment Follows-----

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From fraserredmond at gmail.com  Fri Oct 29 14:23:36 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 29 Oct 2010 22:23:36 +0100
Subject: [Freeswitch-users] Loss of first second of media
In-Reply-To: <AANLkTi=HOiA38khKujHo4y=oP90OKP4Pcmd-L5m70P7K@mail.gmail.com>
References: <AANLkTimECwtVmnbFsBJRiN3YSaGyXbvh-sKtKYPey8xx@mail.gmail.com>
	<AANLkTik5H74MupY8iyx6Ao4=Rv=wCm8jv-w=xCtSnm0Z@mail.gmail.com>
	<AANLkTikpVH-s6AvJ7fQoZu2PE4dduTg+d-ygyBWSfr4R@mail.gmail.com>
	<AANLkTi=FZDqev9R1aTMpdvTk8t9NfxiyeisQSRBxfTEb@mail.gmail.com>
	<AANLkTikY68ovuiSO9xkbZZPSskxCNsjrpTSdb-42bwJB@mail.gmail.com>
	<AANLkTikiG3xPKDVkwU4GDk5t6KDe67AQmM-dkheMCSu7@mail.gmail.com>
	<AANLkTinJRcpRx_BQyKifECcurYb+vG2P8Nct1xT_X3XO@mail.gmail.com>
	<AANLkTinpaRivNHteDp4i6WJjmgsPAE6otVx5LjgJcvWr@mail.gmail.com>
	<AANLkTikiVxFAOwftrmfqd3LtD4H__UUytsM117WBhZy0@mail.gmail.com>
	<AANLkTikLtKhBiCZeMY+GH6X7jb4wVdybVF=OiMePK8Rm@mail.gmail.com>
	<AANLkTimyRrtba8nD7jkWJY2DysebRaycpvRXY3b_A3r-@mail.gmail.com>
	<AANLkTinrSNUr49b3LE6c6ZP7NZ9d13t1CK2bPW2x+anD@mail.gmail.com>
	<AANLkTi=HOiA38khKujHo4y=oP90OKP4Pcmd-L5m70P7K@mail.gmail.com>
Message-ID: <AANLkTin7NowMjGG7Ze4XOgUr0qn9XMoKG4SK9M7_be0r@mail.gmail.com>

Thanks, but that doesn't fit - I've tried it with Bria, and with the recent
version of XLite 4, and its happened on both. They're both made by
Counterpath, so I've also tried 3CXphone, and it's happening on that too.

More suspicious though is that it only happens with some gateways, and not
others. Or did you mean that you think it's a bug with the gateway's
systems? The two I've tried that have this problem are both big outfits, not
some guy operating out of his garage :-) Also, both were already listed in
the Gateways on the wiki, so I'm guessing someone else is already using them
with Freeswitch.

My server is running on Amazon AWS - could it be a timing/virtualization
type problem, or something like that?

Any other ideas?

Cheers,
Fraser




On Fri, Oct 29, 2010 at 9:39 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> OK so,
> The phone sends a 180 ringing with NO SDP
> then it starts sending RTP
> That's is not right.  It's a bug in the phone.
>
>
>
> On Fri, Oct 29, 2010 at 1:35 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > can you try another one with just udp and not "port 5060"
> > so I can see the rtp too
> >
> >
> > On Fri, Oct 29, 2010 at 12:28 PM, Fraser Redmond
> > <fraserredmond at gmail.com> wrote:
> >> Thanks Anthony, it's here:
> >> http://pastebin.freeswitch.org/14350
> >>
> >> And pcap is attached.
> >>
> >> The call connects around (or just before) the 16:58:35 mark (line 558 is
> >> what I see in the terminal while waiting for it to connect - both
> >> early-media and the missing start of the media)
> >>
> >> Cheers,
> >> Fraser
> >>
> >>
> >>
> >>
> >> On Thu, Oct 28, 2010 at 3:59 PM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >>>
> >>> Can you do this trace with debug level logging in addition to the sip
> >>> trace
> >>> console loglevel debug
> >>>
> >>> you also may want to get a pcap of it
> >>>
> >>> tshark udp and port 5060 -w test.pcap
> >>>
> >>>
> >>>
> >>> On Sat, Oct 23, 2010 at 11:44 AM, Fraser Redmond
> >>> <fraserredmond at gmail.com> wrote:
> >>> > Thanks Anthony,
> >>> >
> >>> > Finally managed to get a sip trace - could you do me a favor and take
> a
> >>> > look
> >>> > and/or give me some ideas of what to look for?
> >>> >
> >>> > http://pastebin.freeswitch.org/14300
> >>> >
> >>> > I've highlighted lines 168 and 193. In between these lines is where
> the
> >>> > number is dialed and rings once, then picks up, then theres silence
> for
> >>> > a
> >>> > second or two, and that second SIP message is when I start hearing
> >>> > audio.
> >>> >
> >>> > Thanks,
> >>> > Fraser
> >>> >
> >>> >
> >>> >
> >>> >
> >>> > On Thu, Oct 21, 2010 at 6:44 PM, Anthony Minessale
> >>> > <anthony.minessale at gmail.com> wrote:
> >>> >>
> >>> >> its a blue message on cli
> >>> >>
> >>> >> It could also be the other side expecting us to send media first or
> >>> >> something silly.
> >>> >> try getting a sip trace and figure out when the rtp starts arriving.
> >>> >>
> >>> >>
> >>> >> On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond
> >>> >> <fraserredmond at gmail.com> wrote:
> >>> >> > Sorry, yes, I am setting ignore_early_media=true in the first
> area.
> >>> >> > (Or
> >>> >> > are
> >>> >> > you saying that should be off? I forget now why I needed it on,
> but
> >>> >> > there
> >>> >> > was a reason I added it.)
> >>> >> >
> >>> >> > Yes, the bridge doesn't start until after the A-leg has answered.
> >>> >> >
> >>> >> > Thanks for the suggestion about nat/auto-changing port, I'll have
> a
> >>> >> > look
> >>> >> > into that - would that be in the cli output or in a sip trace?
> I've
> >>> >> > already
> >>> >> > looked and it's not appearing in the CLI output (with
> >>> >> > loglevel=debug),
> >>> >> > haven't looked in the sip trace yet.
> >>> >> >
> >>> >> > Cheers,
> >>> >> > Fraser
> >>> >> >
> >>> >> >
> >>> >> >
> >>> >> >
> >>> >> > On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale
> >>> >> > <anthony.minessale at gmail.com> wrote:
> >>> >> >>
> >>> >> >> are you setting ignore_early_media=true in the first vars=values
> >>> >> >> area?
> >>> >> >>
> >>> >> >> This looks like you could be calling one leg who is still not
> >>> >> >> answered
> >>> >> >> and then bridging it to another dest.  The bridge app will wait
> for
> >>> >> >> the first leg to answer before bridging.
> >>> >> >>
> >>> >> >> Also if you have any NAT anywhere, look for an "auto-changing
> port"
> >>> >> >> type message which can also be attributed to this due to a
> detection
> >>> >> >> period for incorrect ports.
> >>> >> >>
> >>> >> >>
> >>> >> >>
> >>> >> >> On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond
> >>> >> >> <fraserredmond at gmail.com> wrote:
> >>> >> >> > event_socket:
> >>> >> >> > api originate {vars=values}user/$fromExtn at Domain
> >>> >> >> > 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline
> >>> >> >> >
> >>> >> >> > then
> >>> >> >> >
> >>> >> >> > dialplan:
> >>> >> >> > <action application="set"
> >>> >> >> > data="effective_caller_id_number=+1800number"/>
> >>> >> >> > <action application="set" data="effective_caller_id_name="/>
> >>> >> >> > (set and/or export a bunch of other vars too)
> >>> >> >> > <action application="set"
> >>> >> >> >
> >>> >> >> >
> >>> >> >> >
> >>> >> >> > data="dial_string=sofia/gateway/
> gatewayname.com/00${destination_number}<http://gatewayname.com/00$%7Bdestination_number%7D>
> "/>
> >>> >> >> > <action application="bridge" data="${dial_string}"/>
> >>> >> >> >
> >>> >> >> > Cheers,
> >>> >> >> > Fraser
> >>> >> >> >
> >>> >> >> >
> >>> >> >> >
> >>> >> >> >
> >>> >> >> > On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale
> >>> >> >> > <anthony.minessale at gmail.com> wrote:
> >>> >> >> >>
> >>> >> >> >> how are you accomplishing that? by which technique?
> >>> >> >> >>
> >>> >> >> >> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond
> >>> >> >> >> <fraserredmond at gmail.com> wrote:
> >>> >> >> >> > The call is originated from Freeswitch (via CLI) to a
> >>> >> >> >> > softphone,
> >>> >> >> >> > then
> >>> >> >> >> > when
> >>> >> >> >> > that is connected it bridges out to the gateway.
> >>> >> >> >> >
> >>> >> >> >> > Cheers,
> >>> >> >> >> > Fraser
> >>> >> >> >> >
> >>> >> >> >> >
> >>> >> >> >> >
> >>> >> >> >> >
> >>> >> >> >> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale
> >>> >> >> >> > <anthony.minessale at gmail.com> wrote:
> >>> >> >> >> >>
> >>> >> >> >> >> Where is the other side of this call coming from?
> >>> >> >> >> >>
> >>> >> >> >> >> [ (   ) ] -> FS -> (PSTN via SIP)
> >>> >> >> >> >>
> >>> >> >> >> >> What goes in the empty space above?
> >>> >> >> >
> >>> >> >> >
> >>> >> >> > _______________________________________________
> >>> >> >> > FreeSWITCH-users mailing list
> >>> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> >> >
> >>> >> >> >
> >>> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> >> > 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<MSN%3Aanthony_minessale at hotmail.com>
> >>> >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> >> >> IRC: irc.freenode.net #freeswitch
> >>> >> >>
> >>> >> >> FreeSWITCH Developer Conference
> >>> >> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> >> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> >> >> pstn:+19193869900
> >>> >> >>
> >>> >> >> _______________________________________________
> >>> >> >> FreeSWITCH-users mailing list
> >>> >> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> >>
> >>> >> >>
> >>> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> >> http://www.freeswitch.org
> >>> >> >
> >>> >> >
> >>> >> > _______________________________________________
> >>> >> > FreeSWITCH-users mailing list
> >>> >> > FreeSWITCH-users at lists.freeswitch.org
> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >> >
> >>> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> > 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<MSN%3Aanthony_minessale at hotmail.com>
> >>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> >> IRC: irc.freenode.net #freeswitch
> >>> >>
> >>> >> FreeSWITCH Developer Conference
> >>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> >> pstn:+19193869900
> >>> >>
> >>> >> _______________________________________________
> >>> >> FreeSWITCH-users mailing list
> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>
> >>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> http://www.freeswitch.org
> >>> >
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > 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<MSN%3Aanthony_minessale at hotmail.com>
> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> IRC: irc.freenode.net #freeswitch
> >>>
> >>> FreeSWITCH Developer Conference
> >>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> pstn:+19193869900
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> 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 <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
>
>
>
> --
> 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tisyfreeswitch at hotmail.fr  Fri Oct 29 12:52:02 2010
From: tisyfreeswitch at hotmail.fr (Tidiane Sy)
Date: Fri, 29 Oct 2010 21:52:02 +0200
Subject: [Freeswitch-users] no audio when originate to 2 PSTNs
Message-ID: <BLU145-W2762A83E99DB06B5F7C262AB450@phx.gbl>





Hi all,
When I do an originate between two PSTNS, the two telephones ring. But when both answer, there is no audio.


I have audio when the originate is between an sip client and a PSTN.


My freeswitch is not natted
my console logs here: http://pastebin.freeswitch.org/14351
my network capture here: https://rcpt.yousendit.com/978754413/751404552085a1a09ea574cb819a5f74

Your help will be really appreciated

Tid 		 	   		  
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From mario_fs at mgtech.com  Fri Oct 29 14:51:14 2010
From: mario_fs at mgtech.com (Mario G)
Date: Fri, 29 Oct 2010 14:51:14 -0700
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <AANLkTi=D6siEAquSq6gYh-RnZeJJWrk11mHWK8De2tLh@mail.gmail.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
	<407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>
	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
	<AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
	<6D9FD5E2-21FA-4630-8CF4-A87ADB30AC87@mgtech.com>
	<AANLkTi=BiDiXLgEoj_MfwmZbGjvXoOuNJgUYdetciNCj@mail.gmail.com>
	<9BAFDAF8-1E66-444C-A024-A88FF65D8BCA@mgtech.com>
	<3447E946-0BF8-4F7E-B5E9-E28F53DD0968@mgtech.com>
	<AANLkTi=LRbz6ADhLvBKkjGxOywAVLf--GTkzXEpUoDbi@mail.gmail.com>
	<BE754258-BE27-454F-B740-527A35DBDFFE@mgtech.com>
	<AANLkTi=3Kd-bCHJSED7VEh-cvtpdXMs_EkWZL=s8J4Ux@mail.gmail.com>
	<14205291-75B3-470F-ABA1-EE7F08A695E9@mgtech.com>
	<AANLkTi=D6siEAquSq6gYh-RnZeJJWrk11mHWK8De2tLh@mail.gmail.com>
Message-ID: <2C35DBE8-13B3-4264-BB43-7496A1054AD1@mgtech.com>

Here is a short one: http://pastebin.freeswitch.org/14359 - I caught one right after I started FS, failure occurred in minutes (lucky). Look at the bottom of the trace, you see SIP trace activity and then when it fails no SIP trace activity. Could this possibly be a FS bug? (I am a mainframe assembler systems programmer and I might think so if there were error retry messages but nothing showing in one of my traces...)
Notes:
idone is gateway 1
idtwo is gateway 2  I had to trace both because it was impossible to figure out which one would fail first. Ran several times but kept missing the right one.
I use a url for one gateway and ip for another but it makes no difference since both eventually fail.
10. is local lan
210. is external ip
216. is itsp

Here is a longer one from earlier   http://pastebin.freeswitch.org/14357
Notes:
A call was received and hung up for idtwo - beginning of trace
11 minutes later idtwo failed - see last line of trace

Thank you very much!
Mario

On Oct 29, 2010, at 12:11 PM, Anthony Minessale wrote:

> stun-enabled must be true in your profile XML to see what you pasted.
> 
> Get me a sip trace of this from when it works until when it fails
> only enable the sip trace on the profile with the gateway to reduce traffic
> 
> 
> 
> On Fri, Oct 29, 2010 at 1:56 PM, Mario G <mario_fs at mgtech.com> wrote:
>> Oh my...  looks it was not NAT after all? Please help! I changed to all profiles to static per instructions below and still have the problem:
>> 2010-10-29 11:15:18.536446 [NOTICE] sofia_reg.c:342 Registering uuid1
>> 2010-10-29 11:15:34.313150 [WARNING] sofia_reg.c:387 uuid1 Failed Registration, setting retry to 15 seconds.
>> 
>> sofia global siptrace on did not show any activity for this gateway in or out, others were fine but eventually fail. I setup static:
>> 1. set the params ext-sip-ip and ext-rtp-ip to my external static IP
>> 2. map the sip ports (5060-5080) and all of the rtp ports (16384-32767)  to FS lan addr.
>> 3. set sip-ip and rtp-ip to the lan addr of FS
>> 4. start FS with -nonat
>> 
>> I don't know what to try next. BTW, the sofia status for the profiles shows stun enabled but I did not set it up anywhere:
>> 
>> Name                    uuid1
>> Domain Name             N/A
>> Auto-NAT                false
>> DBName                  sofia_reg_mvvyl
>> Pres Hosts
>> Dialplan                XML
>> Context                 public
>> Challenge Realm         auto_to
>> RTP-IP                  10.x.x.20
>> Ext-RTP-IP              210.x.x.100
>> SIP-IP                  10.x.x.20
>> Ext-SIP-IP              210.x.x.100
>> URL                     sip:mod_sofia at 210.x.x.100:5068
>> BIND-URL                sip:mod_sofia at 210.x.x.100:5068;maddr=10.x.x.20
>> HOLD-MUSIC              local_stream://moh
>> OUTBOUND-PROXY          N/A
>> CODECS IN               G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
>> CODECS OUT              PCMU,PCMA,GSM
>> TEL-EVENT               101
>> DTMF-MODE               rfc2833
>> CNG                     13
>> SESSION-TO              0
>> MAX-DIALOG              0
>> NOMEDIA                 false
>> LATE-NEG                false
>> PROXY-MEDIA             false
>> AGGRESSIVENAT           false
>> STUN-ENABLED            true
>> STUN-AUTO-DISABLE       false
>> CALLS-IN                0
>> FAILED-CALLS-IN         0
>> CALLS-OUT               0
>> FAILED-CALLS-OUT        0
>> 
>> 
>> On Oct 27, 2010, at 10:04 AM, Anthony Minessale wrote:
>> 
>>> if you map it or not, a scanner would penetrate it.
>>> There are lot of sip scanners out there now, you just need to beware of them.
>>> 
>>> 
>>> On Wed, Oct 27, 2010 at 11:50 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>> Thanks so much! I am sure many others will find this info invaluable. I will try the static route again but have one question: When I started with FS I found a "sip scanner"  in FS and someone on this group said not to use port mapping since it was a security risk. Is that true?
>>>> 
>>>> On Oct 27, 2010, at 9:10 AM, Anthony Minessale wrote:
>>>> 
>>>>> you are completely guessing at things.
>>>>> I want you to understand that the only reason you are having problems
>>>>> with this is because you don't understand how it works enough to know
>>>>> what you are doing 100%
>>>>> 
>>>>> Its a given that the pnp stuff is only for your dynamic IP.
>>>>> aggressive-nat-detection and sip-force-expires are all related to
>>>>> inbound calls when the things who are registering to you may be behind
>>>>> nat.
>>>>> 
>>>>> You need to learn the difference between which nat tools are
>>>>> *) designed for your FS to run behind nat
>>>>> *) designed for FS to run public and accept connections from devices behind nat.
>>>>> 
>>>>> If you have a static IP, you don't need the pnp stuff so -nonat is fine
>>>>> What you need to do is set
>> 
>>>>> 1) set the params ext-sip-ip and ext-rtp-ip to your external static IP
>>>>> 2) map the sip ports and all of the rtp ports from your static IP to FS lan addr
>>>>> 3) set sip-ip and rtp-ip to the lan addr you forwarded through.
>>>> 
>>>>> 
>>>>> If you don't do this: your outbound registration will use NAT to your
>>>>> provider and if there is no activity for the expire time on your NAT
>>>>> mapping the reverse port mapping from your provider back to you is
>>>>> lost.  This is why you set your register expires to a very low number,
>>>>> (you need to make sure the provider does not turn the expires back up
>>>>> in the reply because it will beat your choice *see sip trace) if this
>>>>> is the case then you need the "ping" option set to 30, to continuously
>>>>> send an options to your provider.
>>>>> 
>>>>> The static mapping is obviously the better, easier and more reliable solution.
>>>>> 
>>>>> So I want you to understand that the only way to keep a nat mapped
>>>>> port alive is to continuously send traffic, all the other methods that
>>>>> you are mentioning are to detect that phones registered to your are
>>>>> behind nat, I gave you that force-expires option before because your
>>>>> trace was full of inbound reg so I thought that is what you wanted
>>>>> help with.
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> On Wed, Oct 27, 2010 at 10:43 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>> I should mention that I did not have this problem with an SPA9000 PBX
>>>>>> (asterisk based) for over two years so FS may be pickier about upnp and/or
>>>>>> nat, or just better at it exposing a problem in the router.
>>>>>> I made different changes to the gateways to test different things. One
>>>>>> failed after 17 hours, the other two stayed up.  What did not work:
>>>>>> added <variable name="sip-force-expires" value="30"/> to the directory
>>>>>> entries as suggested.
>>>>>> set the gateway expire times to 30 seconds.
>>>>>> What worked (could be coincidental) for the two gateways that stayed up:
>>>>>> I Added <param name="aggressive-nat-detection" value="true"/>
>>>>>> I originally setup FS to use the static ip by setting external sip/rtp to
>>>>>> just the static ip (no autonat:) and ran with -nonat but I could not get
>>>>>> incoming calls. The only way it worked was to use autonat:1.2.3.4. The
>>>>>> router has 1 static public address and 1 dynamic external IP, this is the
>>>>>> root of the problem, upnp only tells FS about the dynamic ip  Will keep this
>>>>>> thread up-to-date for anyone who may be in the same boat someday. Thanks
>>>>>> again for looking at the trace.
>>>>>> Mario
>>>>>> 
>>>>>> You should be setting the req freq to a low number on the outbound gateways
>>>>>> 
>>>>>> The examples you showed had a series of inbound reg
>>>>>> 
>>>>>> also set expire-seconds to 30 in your gateway xml
>>>>>> 
>>>>>> 
>>>>>> The problem is if you are not constantly sending traffic to the box
>>>>>> 
>>>>>> the nat mapping will go away.
>>>>>> 
>>>>>> If you are in production you should be using a static ip with a static
>>>>>> 
>>>>>> mapping, any trouble you are having is your own fault for playing with
>>>>>> 
>>>>>> fire.  The best we can do is tell you how to keep it contained.
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>> 
>>>>>> I made the change. I had no idea the settings for the inside phones effected
>>>>>> nat for the outside sip accounts. I was looking into aggressive-nat-
>>>>>> detection since the internal profile status always shows the right external
>>>>>> static IP but the nat_ap status always shows the dynamic ip. Crossing
>>>>>> fingers/etc since this problem is 85% of time (weeks!) into FS changeover.
>>>>>> Thanks!
>>>>>> 
>>>>>> Mario
>>>>>> 
>>>>>> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
>>>>>> 
>>>>>> add
>>>>>> 
>>>>>> <variable name="sip-force-expires" value="30"/>
>>>>>> 
>>>>>> to the <variables> section of your <user>
>>>>>> 
>>>>>> you have it at 600 and the nat mapping is timing out while the 600
>>>>>> 
>>>>>> seconds is ticking away
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>> 
>>>>>> From the TSP:
>>>>>> 
>>>>>> "I have enabled the SIP trace on your account. We are not currently seeing
>>>>>> 
>>>>>> any registration attempts to your account within the last 15 minutes. Please
>>>>>> 
>>>>>> restart FreeSwitch so that registration attempts begin again. Thank you. ".
>>>>>> 
>>>>>> So FS is not getting past router.
>>>>>> 
>>>>>> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>>>>>> 
>>>>>> I ran the global trace during the problem and it is
>>>>>> 
>>>>>> at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
>>>>>> 
>>>>>> "accttwo", "acct3". The trace includes phones since it was global. I am
>>>>>> 
>>>>>> using:
>>>>>> 
>>>>>>    <param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>>>>>> 
>>>>>>    <param name="ext-sip-ip" value="autonat:my-static.ip"/>
>>>>>> 
>>>>>> I tried dumping nat and removing the autonat: above and using -nonat but
>>>>>> 
>>>>>> that did not work, registration proceeded but no calls inbound.
>>>>>> 
>>>>>> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>>>>>> 
>>>>>> Whoops, I am using an IP address for at least one gateway so that is not the
>>>>>> 
>>>>>> problem:
>>>>>> 
>>>>>> They look outbound to me and I am using dns for 2 and an IP for one so that
>>>>>> 
>>>>>> is not the issue. I was able to get FS to clear this up by doing "nat_map
>>>>>> 
>>>>>> reinit" which is why I think this is a nat problem. I will do the trace you
>>>>>> 
>>>>>> mentioned. I will plug an ip address into one of the gateways to see what
>>>>>> 
>>>>>> happens, they all fail at once. Thanks for responding!
>>>>>> 
>>>>>> Mario
>>>>>> 
>>>>>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>>>>>> 
>>>>>> I really need help on this as I have weeks into this problem. I thought I
>>>>>> 
>>>>>> had it nailed but I guess not. After 5.5 hours I get:
>>>>>> 
>>>>>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>>>>>> 
>>>>>> Registration, setting retry to 15 seconds.
>>>>>> 
>>>>>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>>>> 
>>>>>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>>>> 
>>>>>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
>>>>>> 
>>>>>> Registration, setting retry to 30 seconds.
>>>>>> 
>>>>>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>>>>>> 
>>>>>> Registration, setting retry to 15 seconds.
>>>>>> 
>>>>>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>>>>>> 
>>>>>> Registration, setting retry to 15 seconds.
>>>>>> 
>>>>>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>>>> 
>>>>>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>>>> 
>>>>>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>>>>>> 
>>>>>> and no way to make/get calls until I restart FS. I did this:
>>>>>> 
>>>>>> 1. log 7
>>>>>> 
>>>>>> 2. sofia profile xxxx siptrace on   for each profile/gateway
>>>>>> 
>>>>>> 3. restarted router
>>>>>> 
>>>>>> All three did not solve the problem. The trace and log produced no
>>>>>> 
>>>>>> additional lines which is why I am wondering if FS has a problem since the
>>>>>> 
>>>>>> trace shows no SIP activity.
>>>>>> 
>>>>>> 3 gateways with 2 ITSPs
>>>>>> 
>>>>>> 2 DSL/WAN lines, 1 static and 1 dynamic
>>>>>> 
>>>>>> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
>>>>>> 
>>>>>> external static ip.
>>>>>> 
>>>>>> sofia status profile ... has the right ext ip
>>>>>> 
>>>>>> nat_map status shows the dynamic (wrong) IP
>>>>>> 
>>>>>> I tried starting with -nonat but that was worse
>>>>>> 
>>>>>> the only way to fix is restart FS.
>>>>>> 
>>>>>> I read the wiki on external nat, auto_nat and everything else many times.
>>>>>> 
>>>>>> Thanks Mario
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> 
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> 
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> 
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> 
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 
>>>>>> 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
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> 
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> 
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 
>>>>>> 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
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> 
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 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
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> 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
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Fri Oct 29 14:53:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 29 Oct 2010 16:53:38 -0500
Subject: [Freeswitch-users] Loss of first second of media
In-Reply-To: <AANLkTin7NowMjGG7Ze4XOgUr0qn9XMoKG4SK9M7_be0r@mail.gmail.com>
References: <AANLkTimECwtVmnbFsBJRiN3YSaGyXbvh-sKtKYPey8xx@mail.gmail.com>
	<AANLkTik5H74MupY8iyx6Ao4=Rv=wCm8jv-w=xCtSnm0Z@mail.gmail.com>
	<AANLkTikpVH-s6AvJ7fQoZu2PE4dduTg+d-ygyBWSfr4R@mail.gmail.com>
	<AANLkTi=FZDqev9R1aTMpdvTk8t9NfxiyeisQSRBxfTEb@mail.gmail.com>
	<AANLkTikY68ovuiSO9xkbZZPSskxCNsjrpTSdb-42bwJB@mail.gmail.com>
	<AANLkTikiG3xPKDVkwU4GDk5t6KDe67AQmM-dkheMCSu7@mail.gmail.com>
	<AANLkTinJRcpRx_BQyKifECcurYb+vG2P8Nct1xT_X3XO@mail.gmail.com>
	<AANLkTinpaRivNHteDp4i6WJjmgsPAE6otVx5LjgJcvWr@mail.gmail.com>
	<AANLkTikiVxFAOwftrmfqd3LtD4H__UUytsM117WBhZy0@mail.gmail.com>
	<AANLkTikLtKhBiCZeMY+GH6X7jb4wVdybVF=OiMePK8Rm@mail.gmail.com>
	<AANLkTimyRrtba8nD7jkWJY2DysebRaycpvRXY3b_A3r-@mail.gmail.com>
	<AANLkTinrSNUr49b3LE6c6ZP7NZ9d13t1CK2bPW2x+anD@mail.gmail.com>
	<AANLkTi=HOiA38khKujHo4y=oP90OKP4Pcmd-L5m70P7K@mail.gmail.com>
	<AANLkTin7NowMjGG7Ze4XOgUr0qn9XMoKG4SK9M7_be0r@mail.gmail.com>
Message-ID: <AANLkTi=g=9g8qAokDzROk=8-2SJy4TTUOpvPmp0uD_N6@mail.gmail.com>

I'm not sure what you mean it doesn't fit.

Did you look at your trace you sent me in wireshark ?  (see bad_call.png)
You can see the bria phone answer 180 ringing with no sdp in packet 7
Then it starts sending RTP .. ??? That is completely wrong.

It never told us it's sending RTP how can we receive it?  We don't have a SDP.


Then in bad_call2.png it sends a 200+sdp (yes that's what we wanted)
Now you have 2 way audio.




On Fri, Oct 29, 2010 at 4:23 PM, Fraser Redmond <fraserredmond at gmail.com> wrote:
> Thanks, but that doesn't fit - I've tried it with Bria, and with the recent
> version of XLite 4, and its happened on both. They're both made by
> Counterpath, so I've also tried 3CXphone, and it's happening on that too.
>
> More suspicious though is that it only happens with some gateways, and not
> others. Or did you mean that you think it's a bug with the gateway's
> systems? The two I've tried that have this problem are both big outfits, not
> some guy operating out of his garage :-) Also, both were already listed in
> the Gateways on the wiki, so I'm guessing someone else is already using them
> with Freeswitch.
>
> My server is running on Amazon AWS - could it be a timing/virtualization
> type problem, or something like that?
>
> Any other ideas?
>
> Cheers,
> Fraser
>
>
>
>
> On Fri, Oct 29, 2010 at 9:39 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> OK so,
>> The phone sends a 180 ringing with NO SDP
>> then it starts sending RTP
>> That's is not right. ?It's a bug in the phone.
>>
>>
>>
>> On Fri, Oct 29, 2010 at 1:35 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > can you try another one with just udp and not "port 5060"
>> > so I can see the rtp too
>> >
>> >
>> > On Fri, Oct 29, 2010 at 12:28 PM, Fraser Redmond
>> > <fraserredmond at gmail.com> wrote:
>> >> Thanks Anthony, it's here:
>> >> http://pastebin.freeswitch.org/14350
>> >>
>> >> And pcap is attached.
>> >>
>> >> The call connects around (or just before) the 16:58:35 mark (line 558
>> >> is
>> >> what I see in the terminal while waiting for it to connect - both
>> >> early-media and the missing start of the media)
>> >>
>> >> Cheers,
>> >> Fraser
>> >>
>> >>
>> >>
>> >>
>> >> On Thu, Oct 28, 2010 at 3:59 PM, Anthony Minessale
>> >> <anthony.minessale at gmail.com> wrote:
>> >>>
>> >>> Can you do this trace with debug level logging in addition to the sip
>> >>> trace
>> >>> console loglevel debug
>> >>>
>> >>> you also may want to get a pcap of it
>> >>>
>> >>> tshark udp and port 5060 -w test.pcap
>> >>>
>> >>>
>> >>>
>> >>> On Sat, Oct 23, 2010 at 11:44 AM, Fraser Redmond
>> >>> <fraserredmond at gmail.com> wrote:
>> >>> > Thanks Anthony,
>> >>> >
>> >>> > Finally managed to get a sip trace - could you do me a favor and
>> >>> > take a
>> >>> > look
>> >>> > and/or give me some ideas of what to look for?
>> >>> >
>> >>> > http://pastebin.freeswitch.org/14300
>> >>> >
>> >>> > I've highlighted lines 168 and 193. In between these lines is where
>> >>> > the
>> >>> > number is dialed and rings once, then picks up, then theres silence
>> >>> > for
>> >>> > a
>> >>> > second or two, and that second SIP message is when I start hearing
>> >>> > audio.
>> >>> >
>> >>> > Thanks,
>> >>> > Fraser
>> >>> >
>> >>> >
>> >>> >
>> >>> >
>> >>> > On Thu, Oct 21, 2010 at 6:44 PM, Anthony Minessale
>> >>> > <anthony.minessale at gmail.com> wrote:
>> >>> >>
>> >>> >> its a blue message on cli
>> >>> >>
>> >>> >> It could also be the other side expecting us to send media first or
>> >>> >> something silly.
>> >>> >> try getting a sip trace and figure out when the rtp starts
>> >>> >> arriving.
>> >>> >>
>> >>> >>
>> >>> >> On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond
>> >>> >> <fraserredmond at gmail.com> wrote:
>> >>> >> > Sorry, yes, I am setting ignore_early_media=true in the first
>> >>> >> > area.
>> >>> >> > (Or
>> >>> >> > are
>> >>> >> > you saying that should be off? I forget now why I needed it on,
>> >>> >> > but
>> >>> >> > there
>> >>> >> > was a reason I added it.)
>> >>> >> >
>> >>> >> > Yes, the bridge doesn't start until after the A-leg has answered.
>> >>> >> >
>> >>> >> > Thanks for the suggestion about nat/auto-changing port, I'll have
>> >>> >> > a
>> >>> >> > look
>> >>> >> > into that - would that be in the cli output or in a sip trace?
>> >>> >> > I've
>> >>> >> > already
>> >>> >> > looked and it's not appearing in the CLI output (with
>> >>> >> > loglevel=debug),
>> >>> >> > haven't looked in the sip trace yet.
>> >>> >> >
>> >>> >> > Cheers,
>> >>> >> > Fraser
>> >>> >> >
>> >>> >> >
>> >>> >> >
>> >>> >> >
>> >>> >> > On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale
>> >>> >> > <anthony.minessale at gmail.com> wrote:
>> >>> >> >>
>> >>> >> >> are you setting ignore_early_media=true in the first vars=values
>> >>> >> >> area?
>> >>> >> >>
>> >>> >> >> This looks like you could be calling one leg who is still not
>> >>> >> >> answered
>> >>> >> >> and then bridging it to another dest. ?The bridge app will wait
>> >>> >> >> for
>> >>> >> >> the first leg to answer before bridging.
>> >>> >> >>
>> >>> >> >> Also if you have any NAT anywhere, look for an "auto-changing
>> >>> >> >> port"
>> >>> >> >> type message which can also be attributed to this due to a
>> >>> >> >> detection
>> >>> >> >> period for incorrect ports.
>> >>> >> >>
>> >>> >> >>
>> >>> >> >>
>> >>> >> >> On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond
>> >>> >> >> <fraserredmond at gmail.com> wrote:
>> >>> >> >> > event_socket:
>> >>> >> >> > api originate {vars=values}user/$fromExtn at Domain
>> >>> >> >> > 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline
>> >>> >> >> >
>> >>> >> >> > then
>> >>> >> >> >
>> >>> >> >> > dialplan:
>> >>> >> >> > <action application="set"
>> >>> >> >> > data="effective_caller_id_number=+1800number"/>
>> >>> >> >> > <action application="set" data="effective_caller_id_name="/>
>> >>> >> >> > (set and/or export a bunch of other vars too)
>> >>> >> >> > <action application="set"
>> >>> >> >> >
>> >>> >> >> >
>> >>> >> >> >
>> >>> >> >> >
>> >>> >> >> > data="dial_string=sofia/gateway/gatewayname.com/00${destination_number}"/>
>> >>> >> >> > <action application="bridge" data="${dial_string}"/>
>> >>> >> >> >
>> >>> >> >> > Cheers,
>> >>> >> >> > Fraser
>> >>> >> >> >
>> >>> >> >> >
>> >>> >> >> >
>> >>> >> >> >
>> >>> >> >> > On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale
>> >>> >> >> > <anthony.minessale at gmail.com> wrote:
>> >>> >> >> >>
>> >>> >> >> >> how are you accomplishing that? by which technique?
>> >>> >> >> >>
>> >>> >> >> >> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond
>> >>> >> >> >> <fraserredmond at gmail.com> wrote:
>> >>> >> >> >> > The call is originated from Freeswitch (via CLI) to a
>> >>> >> >> >> > softphone,
>> >>> >> >> >> > then
>> >>> >> >> >> > when
>> >>> >> >> >> > that is connected it bridges out to the gateway.
>> >>> >> >> >> >
>> >>> >> >> >> > Cheers,
>> >>> >> >> >> > Fraser
>> >>> >> >> >> >
>> >>> >> >> >> >
>> >>> >> >> >> >
>> >>> >> >> >> >
>> >>> >> >> >> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale
>> >>> >> >> >> > <anthony.minessale at gmail.com> wrote:
>> >>> >> >> >> >>
>> >>> >> >> >> >> Where is the other side of this call coming from?
>> >>> >> >> >> >>
>> >>> >> >> >> >> [ ( ? ) ] -> FS -> (PSTN via SIP)
>> >>> >> >> >> >>
>> >>> >> >> >> >> What goes in the empty space above?
>> >>> >> >> >
>> >>> >> >> >
>> >>> >> >> > _______________________________________________
>> >>> >> >> > FreeSWITCH-users mailing list
>> >>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >> >> >
>> >>> >> >> >
>> >>> >> >> >
>> >>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >> >> > 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
>> >>> >> >>
>> >>> >> >> _______________________________________________
>> >>> >> >> FreeSWITCH-users mailing list
>> >>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >> >>
>> >>> >> >>
>> >>> >> >>
>> >>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >> >> http://www.freeswitch.org
>> >>> >> >
>> >>> >> >
>> >>> >> > _______________________________________________
>> >>> >> > FreeSWITCH-users mailing list
>> >>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >> >
>> >>> >> >
>> >>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >> > 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
>> >>> >>
>> >>> >> _______________________________________________
>> >>> >> FreeSWITCH-users mailing list
>> >>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>
>> >>> >>
>> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >> http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > 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
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> 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
>> >
>>
>>
>>
>> --
>> 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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 mario_fs at mgtech.com  Fri Oct 29 15:05:02 2010
From: mario_fs at mgtech.com (Mario G)
Date: Fri, 29 Oct 2010 15:05:02 -0700
Subject: [Freeswitch-users] wiki question - want to add Mac osX info
Message-ID: <AEFCAD97-9D78-498C-9B90-8712C39F6268@mgtech.com>

I am about to start work my wiki contribution which specifically target my installation on Mac osX and other related information to Mac osX. I saw there is no category for "installation". So then does this info go under "Examples"?
Mario

From fraserredmond at gmail.com  Fri Oct 29 15:38:58 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 29 Oct 2010 23:38:58 +0100
Subject: [Freeswitch-users] Loss of first second of media
In-Reply-To: <AANLkTi=g=9g8qAokDzROk=8-2SJy4TTUOpvPmp0uD_N6@mail.gmail.com>
References: <AANLkTimECwtVmnbFsBJRiN3YSaGyXbvh-sKtKYPey8xx@mail.gmail.com>
	<AANLkTik5H74MupY8iyx6Ao4=Rv=wCm8jv-w=xCtSnm0Z@mail.gmail.com>
	<AANLkTikpVH-s6AvJ7fQoZu2PE4dduTg+d-ygyBWSfr4R@mail.gmail.com>
	<AANLkTi=FZDqev9R1aTMpdvTk8t9NfxiyeisQSRBxfTEb@mail.gmail.com>
	<AANLkTikY68ovuiSO9xkbZZPSskxCNsjrpTSdb-42bwJB@mail.gmail.com>
	<AANLkTikiG3xPKDVkwU4GDk5t6KDe67AQmM-dkheMCSu7@mail.gmail.com>
	<AANLkTinJRcpRx_BQyKifECcurYb+vG2P8Nct1xT_X3XO@mail.gmail.com>
	<AANLkTinpaRivNHteDp4i6WJjmgsPAE6otVx5LjgJcvWr@mail.gmail.com>
	<AANLkTikiVxFAOwftrmfqd3LtD4H__UUytsM117WBhZy0@mail.gmail.com>
	<AANLkTikLtKhBiCZeMY+GH6X7jb4wVdybVF=OiMePK8Rm@mail.gmail.com>
	<AANLkTimyRrtba8nD7jkWJY2DysebRaycpvRXY3b_A3r-@mail.gmail.com>
	<AANLkTinrSNUr49b3LE6c6ZP7NZ9d13t1CK2bPW2x+anD@mail.gmail.com>
	<AANLkTi=HOiA38khKujHo4y=oP90OKP4Pcmd-L5m70P7K@mail.gmail.com>
	<AANLkTin7NowMjGG7Ze4XOgUr0qn9XMoKG4SK9M7_be0r@mail.gmail.com>
	<AANLkTi=g=9g8qAokDzROk=8-2SJy4TTUOpvPmp0uD_N6@mail.gmail.com>
Message-ID: <AANLkTikC052SeKz6=1yoxa9A3HD-CnhdjwfUmWLfqV0J@mail.gmail.com>

Ah, ok, thanks, I'll do some more tests with the trace running, and see
whats happening with different softphones and gateways.

Thanks,
Fraser




On Fri, Oct 29, 2010 at 10:53 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I'm not sure what you mean it doesn't fit.
>
> Did you look at your trace you sent me in wireshark ?  (see bad_call.png)
> You can see the bria phone answer 180 ringing with no sdp in packet 7
> Then it starts sending RTP .. ??? That is completely wrong.
>
> It never told us it's sending RTP how can we receive it?  We don't have a
> SDP.
>
>
> Then in bad_call2.png it sends a 200+sdp (yes that's what we wanted)
> Now you have 2 way audio.
>
>
>
>
> On Fri, Oct 29, 2010 at 4:23 PM, Fraser Redmond <fraserredmond at gmail.com>
> wrote:
> > Thanks, but that doesn't fit - I've tried it with Bria, and with the
> recent
> > version of XLite 4, and its happened on both. They're both made by
> > Counterpath, so I've also tried 3CXphone, and it's happening on that too.
> >
> > More suspicious though is that it only happens with some gateways, and
> not
> > others. Or did you mean that you think it's a bug with the gateway's
> > systems? The two I've tried that have this problem are both big outfits,
> not
> > some guy operating out of his garage :-) Also, both were already listed
> in
> > the Gateways on the wiki, so I'm guessing someone else is already using
> them
> > with Freeswitch.
> >
> > My server is running on Amazon AWS - could it be a timing/virtualization
> > type problem, or something like that?
> >
> > Any other ideas?
> >
> > Cheers,
> > Fraser
> >
> >
> >
> >
> > On Fri, Oct 29, 2010 at 9:39 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> OK so,
> >> The phone sends a 180 ringing with NO SDP
> >> then it starts sending RTP
> >> That's is not right.  It's a bug in the phone.
> >>
> >>
> >>
> >> On Fri, Oct 29, 2010 at 1:35 PM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >> > can you try another one with just udp and not "port 5060"
> >> > so I can see the rtp too
> >> >
> >> >
> >> > On Fri, Oct 29, 2010 at 12:28 PM, Fraser Redmond
> >> > <fraserredmond at gmail.com> wrote:
> >> >> Thanks Anthony, it's here:
> >> >> http://pastebin.freeswitch.org/14350
> >> >>
> >> >> And pcap is attached.
> >> >>
> >> >> The call connects around (or just before) the 16:58:35 mark (line 558
> >> >> is
> >> >> what I see in the terminal while waiting for it to connect - both
> >> >> early-media and the missing start of the media)
> >> >>
> >> >> Cheers,
> >> >> Fraser
> >> >>
> >> >>
> >> >>
> >> >>
> >> >> On Thu, Oct 28, 2010 at 3:59 PM, Anthony Minessale
> >> >> <anthony.minessale at gmail.com> wrote:
> >> >>>
> >> >>> Can you do this trace with debug level logging in addition to the
> sip
> >> >>> trace
> >> >>> console loglevel debug
> >> >>>
> >> >>> you also may want to get a pcap of it
> >> >>>
> >> >>> tshark udp and port 5060 -w test.pcap
> >> >>>
> >> >>>
> >> >>>
> >> >>> On Sat, Oct 23, 2010 at 11:44 AM, Fraser Redmond
> >> >>> <fraserredmond at gmail.com> wrote:
> >> >>> > Thanks Anthony,
> >> >>> >
> >> >>> > Finally managed to get a sip trace - could you do me a favor and
> >> >>> > take a
> >> >>> > look
> >> >>> > and/or give me some ideas of what to look for?
> >> >>> >
> >> >>> > http://pastebin.freeswitch.org/14300
> >> >>> >
> >> >>> > I've highlighted lines 168 and 193. In between these lines is
> where
> >> >>> > the
> >> >>> > number is dialed and rings once, then picks up, then theres
> silence
> >> >>> > for
> >> >>> > a
> >> >>> > second or two, and that second SIP message is when I start hearing
> >> >>> > audio.
> >> >>> >
> >> >>> > Thanks,
> >> >>> > Fraser
> >> >>> >
> >> >>> >
> >> >>> >
> >> >>> >
> >> >>> > On Thu, Oct 21, 2010 at 6:44 PM, Anthony Minessale
> >> >>> > <anthony.minessale at gmail.com> wrote:
> >> >>> >>
> >> >>> >> its a blue message on cli
> >> >>> >>
> >> >>> >> It could also be the other side expecting us to send media first
> or
> >> >>> >> something silly.
> >> >>> >> try getting a sip trace and figure out when the rtp starts
> >> >>> >> arriving.
> >> >>> >>
> >> >>> >>
> >> >>> >> On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond
> >> >>> >> <fraserredmond at gmail.com> wrote:
> >> >>> >> > Sorry, yes, I am setting ignore_early_media=true in the first
> >> >>> >> > area.
> >> >>> >> > (Or
> >> >>> >> > are
> >> >>> >> > you saying that should be off? I forget now why I needed it on,
> >> >>> >> > but
> >> >>> >> > there
> >> >>> >> > was a reason I added it.)
> >> >>> >> >
> >> >>> >> > Yes, the bridge doesn't start until after the A-leg has
> answered.
> >> >>> >> >
> >> >>> >> > Thanks for the suggestion about nat/auto-changing port, I'll
> have
> >> >>> >> > a
> >> >>> >> > look
> >> >>> >> > into that - would that be in the cli output or in a sip trace?
> >> >>> >> > I've
> >> >>> >> > already
> >> >>> >> > looked and it's not appearing in the CLI output (with
> >> >>> >> > loglevel=debug),
> >> >>> >> > haven't looked in the sip trace yet.
> >> >>> >> >
> >> >>> >> > Cheers,
> >> >>> >> > Fraser
> >> >>> >> >
> >> >>> >> >
> >> >>> >> >
> >> >>> >> >
> >> >>> >> > On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale
> >> >>> >> > <anthony.minessale at gmail.com> wrote:
> >> >>> >> >>
> >> >>> >> >> are you setting ignore_early_media=true in the first
> vars=values
> >> >>> >> >> area?
> >> >>> >> >>
> >> >>> >> >> This looks like you could be calling one leg who is still not
> >> >>> >> >> answered
> >> >>> >> >> and then bridging it to another dest.  The bridge app will
> wait
> >> >>> >> >> for
> >> >>> >> >> the first leg to answer before bridging.
> >> >>> >> >>
> >> >>> >> >> Also if you have any NAT anywhere, look for an "auto-changing
> >> >>> >> >> port"
> >> >>> >> >> type message which can also be attributed to this due to a
> >> >>> >> >> detection
> >> >>> >> >> period for incorrect ports.
> >> >>> >> >>
> >> >>> >> >>
> >> >>> >> >>
> >> >>> >> >> On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond
> >> >>> >> >> <fraserredmond at gmail.com> wrote:
> >> >>> >> >> > event_socket:
> >> >>> >> >> > api originate {vars=values}user/$fromExtn at Domain
> >> >>> >> >> > 'set:bLegVars=values,transfer:$toNum xml outbound_call'
> inline
> >> >>> >> >> >
> >> >>> >> >> > then
> >> >>> >> >> >
> >> >>> >> >> > dialplan:
> >> >>> >> >> > <action application="set"
> >> >>> >> >> > data="effective_caller_id_number=+1800number"/>
> >> >>> >> >> > <action application="set" data="effective_caller_id_name="/>
> >> >>> >> >> > (set and/or export a bunch of other vars too)
> >> >>> >> >> > <action application="set"
> >> >>> >> >> >
> >> >>> >> >> >
> >> >>> >> >> >
> >> >>> >> >> >
> >> >>> >> >> > data="dial_string=sofia/gateway/
> gatewayname.com/00${destination_number}<http://gatewayname.com/00$%7Bdestination_number%7D>
> "/>
> >> >>> >> >> > <action application="bridge" data="${dial_string}"/>
> >> >>> >> >> >
> >> >>> >> >> > Cheers,
> >> >>> >> >> > Fraser
> >> >>> >> >> >
> >> >>> >> >> >
> >> >>> >> >> >
> >> >>> >> >> >
> >> >>> >> >> > On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale
> >> >>> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >>> >> >> >>
> >> >>> >> >> >> how are you accomplishing that? by which technique?
> >> >>> >> >> >>
> >> >>> >> >> >> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond
> >> >>> >> >> >> <fraserredmond at gmail.com> wrote:
> >> >>> >> >> >> > The call is originated from Freeswitch (via CLI) to a
> >> >>> >> >> >> > softphone,
> >> >>> >> >> >> > then
> >> >>> >> >> >> > when
> >> >>> >> >> >> > that is connected it bridges out to the gateway.
> >> >>> >> >> >> >
> >> >>> >> >> >> > Cheers,
> >> >>> >> >> >> > Fraser
> >> >>> >> >> >> >
> >> >>> >> >> >> >
> >> >>> >> >> >> >
> >> >>> >> >> >> >
> >> >>> >> >> >> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale
> >> >>> >> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >>> >> >> >> >>
> >> >>> >> >> >> >> Where is the other side of this call coming from?
> >> >>> >> >> >> >>
> >> >>> >> >> >> >> [ (   ) ] -> FS -> (PSTN via SIP)
> >> >>> >> >> >> >>
> >> >>> >> >> >> >> What goes in the empty space above?
> >> >>> >> >> >
> >> >>> >> >> >
> >> >>> >> >> > _______________________________________________
> >> >>> >> >> > FreeSWITCH-users mailing list
> >> >>> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >>> >> >> >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >> >> >
> >> >>> >> >> >
> >> >>> >> >> >
> >> >>> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> >> >> > 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<MSN%3Aanthony_minessale at hotmail.com>
> >> >>> >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >>> >> >> IRC: irc.freenode.net #freeswitch
> >> >>> >> >>
> >> >>> >> >> FreeSWITCH Developer Conference
> >> >>> >> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >>> >> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >>> >> >> pstn:+19193869900
> >> >>> >> >>
> >> >>> >> >> _______________________________________________
> >> >>> >> >> FreeSWITCH-users mailing list
> >> >>> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >> >>
> >> >>> >> >>
> >> >>> >> >>
> >> >>> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> >> >> http://www.freeswitch.org
> >> >>> >> >
> >> >>> >> >
> >> >>> >> > _______________________________________________
> >> >>> >> > FreeSWITCH-users mailing list
> >> >>> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >> >
> >> >>> >> >
> >> >>> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> >> > 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<MSN%3Aanthony_minessale at hotmail.com>
> >> >>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >>> >> IRC: irc.freenode.net #freeswitch
> >> >>> >>
> >> >>> >> FreeSWITCH Developer Conference
> >> >>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >>> >> pstn:+19193869900
> >> >>> >>
> >> >>> >> _______________________________________________
> >> >>> >> FreeSWITCH-users mailing list
> >> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >>
> >> >>> >>
> >> >>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> >> http://www.freeswitch.org
> >> >>> >
> >> >>> >
> >> >>> > _______________________________________________
> >> >>> > FreeSWITCH-users mailing list
> >> >>> > FreeSWITCH-users at lists.freeswitch.org
> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >
> >> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> > 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<MSN%3Aanthony_minessale at hotmail.com>
> >> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >>> IRC: irc.freenode.net #freeswitch
> >> >>>
> >> >>> FreeSWITCH Developer Conference
> >> >>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >>> pstn:+19193869900
> >> >>>
> >> >>> _______________________________________________
> >> >>> FreeSWITCH-users mailing list
> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>
> >> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> http://www.freeswitch.org
> >> >>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> 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<MSN%3Aanthony_minessale at hotmail.com>
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> > pstn:+19193869900
> >> >
> >>
> >>
> >>
> >> --
> >> 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 <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mario_fs at mgtech.com  Fri Oct 29 16:30:29 2010
From: mario_fs at mgtech.com (Mario G)
Date: Fri, 29 Oct 2010 16:30:29 -0700
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <2C35DBE8-13B3-4264-BB43-7496A1054AD1@mgtech.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
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	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
	<AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
	<6D9FD5E2-21FA-4630-8CF4-A87ADB30AC87@mgtech.com>
	<AANLkTi=BiDiXLgEoj_MfwmZbGjvXoOuNJgUYdetciNCj@mail.gmail.com>
	<9BAFDAF8-1E66-444C-A024-A88FF65D8BCA@mgtech.com>
	<3447E946-0BF8-4F7E-B5E9-E28F53DD0968@mgtech.com>
	<AANLkTi=LRbz6ADhLvBKkjGxOywAVLf--GTkzXEpUoDbi@mail.gmail.com>
	<BE754258-BE27-454F-B740-527A35DBDFFE@mgtech.com>
	<AANLkTi=3Kd-bCHJSED7VEh-cvtpdXMs_EkWZL=s8J4Ux@mail.gmail.com>
	<14205291-75B3-470F-ABA1-EE7F08A695E9@mgtech.com>
	<AANLkTi=D6siEAquSq6gYh-RnZeJJWrk11mHWK8De2tLh@mail.gmail.com>
	<2C35DBE8-13B3-4264-BB43-7496A1054AD1@mgtech.com>
Message-ID: <86E8AE3F-E945-4952-A438-C674F23881D6@mgtech.com>

Oh Boy: Not only is everything set to the static route but I turned off/disconnected the dynamic DSL line so I only had 1 static line to the router. The second wan is set in the router off. Even turned off router uPnP even though I am using -nonat. Guess what.... I still have the problem. Look like when FS says is going to retry it actually does not.

Here is a short one: http://pastebin.freeswitch.org/14359 - I caught one right after I started FS, failure occurred in minutes (lucky). Look at the bottom of the trace, you see SIP trace activity and then when it fails no SIP trace activity. Could this possibly be a FS bug? (I am a mainframe assembler systems programmer and I might think so if there were error retry messages but nothing showing in one of my traces...)
Notes:
idone is gateway 1
idtwo is gateway 2  I had to trace both because it was impossible to figure out which one would fail first. Ran several times but kept missing the right one.
I use a url for one gateway and ip for another but it makes no difference since both eventually fail.
10. is local lan
210. is external ip
216. is itsp

Here is a longer one from earlier   http://pastebin.freeswitch.org/14357
Notes:
A call was received and hung up for idtwo - beginning of trace
11 minutes later idtwo failed - see last line of trace

Thank you very much!
Mario

> 
> On Oct 29, 2010, at 12:11 PM, Anthony Minessale wrote:
> 
>> stun-enabled must be true in your profile XML to see what you pasted.
>> 
>> Get me a sip trace of this from when it works until when it fails
>> only enable the sip trace on the profile with the gateway to reduce traffic
>> 
>> 
>> 
>> On Fri, Oct 29, 2010 at 1:56 PM, Mario G <mario_fs at mgtech.com> wrote:
>>> Oh my...  looks it was not NAT after all? Please help! I changed to all profiles to static per instructions below and still have the problem:
>>> 2010-10-29 11:15:18.536446 [NOTICE] sofia_reg.c:342 Registering uuid1
>>> 2010-10-29 11:15:34.313150 [WARNING] sofia_reg.c:387 uuid1 Failed Registration, setting retry to 15 seconds.
>>> 
>>> sofia global siptrace on did not show any activity for this gateway in or out, others were fine but eventually fail. I setup static:
>>> 1. set the params ext-sip-ip and ext-rtp-ip to my external static IP
>>> 2. map the sip ports (5060-5080) and all of the rtp ports (16384-32767)  to FS lan addr.
>>> 3. set sip-ip and rtp-ip to the lan addr of FS
>>> 4. start FS with -nonat
>>> 
>>> I don't know what to try next. BTW, the sofia status for the profiles shows stun enabled but I did not set it up anywhere:
>>> 
>>> Name                    uuid1
>>> Domain Name             N/A
>>> Auto-NAT                false
>>> DBName                  sofia_reg_mvvyl
>>> Pres Hosts
>>> Dialplan                XML
>>> Context                 public
>>> Challenge Realm         auto_to
>>> RTP-IP                  10.x.x.20
>>> Ext-RTP-IP              210.x.x.100
>>> SIP-IP                  10.x.x.20
>>> Ext-SIP-IP              210.x.x.100
>>> URL                     sip:mod_sofia at 210.x.x.100:5068
>>> BIND-URL                sip:mod_sofia at 210.x.x.100:5068;maddr=10.x.x.20
>>> HOLD-MUSIC              local_stream://moh
>>> OUTBOUND-PROXY          N/A
>>> CODECS IN               G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
>>> CODECS OUT              PCMU,PCMA,GSM
>>> TEL-EVENT               101
>>> DTMF-MODE               rfc2833
>>> CNG                     13
>>> SESSION-TO              0
>>> MAX-DIALOG              0
>>> NOMEDIA                 false
>>> LATE-NEG                false
>>> PROXY-MEDIA             false
>>> AGGRESSIVENAT           false
>>> STUN-ENABLED            true
>>> STUN-AUTO-DISABLE       false
>>> CALLS-IN                0
>>> FAILED-CALLS-IN         0
>>> CALLS-OUT               0
>>> FAILED-CALLS-OUT        0
>>> 
>>> 
>>> On Oct 27, 2010, at 10:04 AM, Anthony Minessale wrote:
>>> 
>>>> if you map it or not, a scanner would penetrate it.
>>>> There are lot of sip scanners out there now, you just need to beware of them.
>>>> 
>>>> 
>>>> On Wed, Oct 27, 2010 at 11:50 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>>> Thanks so much! I am sure many others will find this info invaluable. I will try the static route again but have one question: When I started with FS I found a "sip scanner"  in FS and someone on this group said not to use port mapping since it was a security risk. Is that true?
>>>>> 
>>>>> On Oct 27, 2010, at 9:10 AM, Anthony Minessale wrote:
>>>>> 
>>>>>> you are completely guessing at things.
>>>>>> I want you to understand that the only reason you are having problems
>>>>>> with this is because you don't understand how it works enough to know
>>>>>> what you are doing 100%
>>>>>> 
>>>>>> Its a given that the pnp stuff is only for your dynamic IP.
>>>>>> aggressive-nat-detection and sip-force-expires are all related to
>>>>>> inbound calls when the things who are registering to you may be behind
>>>>>> nat.
>>>>>> 
>>>>>> You need to learn the difference between which nat tools are
>>>>>> *) designed for your FS to run behind nat
>>>>>> *) designed for FS to run public and accept connections from devices behind nat.
>>>>>> 
>>>>>> If you have a static IP, you don't need the pnp stuff so -nonat is fine
>>>>>> What you need to do is set
>>> 
>>>>>> 1) set the params ext-sip-ip and ext-rtp-ip to your external static IP
>>>>>> 2) map the sip ports and all of the rtp ports from your static IP to FS lan addr
>>>>>> 3) set sip-ip and rtp-ip to the lan addr you forwarded through.
>>>>> 
>>>>>> 
>>>>>> If you don't do this: your outbound registration will use NAT to your
>>>>>> provider and if there is no activity for the expire time on your NAT
>>>>>> mapping the reverse port mapping from your provider back to you is
>>>>>> lost.  This is why you set your register expires to a very low number,
>>>>>> (you need to make sure the provider does not turn the expires back up
>>>>>> in the reply because it will beat your choice *see sip trace) if this
>>>>>> is the case then you need the "ping" option set to 30, to continuously
>>>>>> send an options to your provider.
>>>>>> 
>>>>>> The static mapping is obviously the better, easier and more reliable solution.
>>>>>> 
>>>>>> So I want you to understand that the only way to keep a nat mapped
>>>>>> port alive is to continuously send traffic, all the other methods that
>>>>>> you are mentioning are to detect that phones registered to your are
>>>>>> behind nat, I gave you that force-expires option before because your
>>>>>> trace was full of inbound reg so I thought that is what you wanted
>>>>>> help with.
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> On Wed, Oct 27, 2010 at 10:43 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>>> I should mention that I did not have this problem with an SPA9000 PBX
>>>>>>> (asterisk based) for over two years so FS may be pickier about upnp and/or
>>>>>>> nat, or just better at it exposing a problem in the router.
>>>>>>> I made different changes to the gateways to test different things. One
>>>>>>> failed after 17 hours, the other two stayed up.  What did not work:
>>>>>>> added <variable name="sip-force-expires" value="30"/> to the directory
>>>>>>> entries as suggested.
>>>>>>> set the gateway expire times to 30 seconds.
>>>>>>> What worked (could be coincidental) for the two gateways that stayed up:
>>>>>>> I Added <param name="aggressive-nat-detection" value="true"/>
>>>>>>> I originally setup FS to use the static ip by setting external sip/rtp to
>>>>>>> just the static ip (no autonat:) and ran with -nonat but I could not get
>>>>>>> incoming calls. The only way it worked was to use autonat:1.2.3.4. The
>>>>>>> router has 1 static public address and 1 dynamic external IP, this is the
>>>>>>> root of the problem, upnp only tells FS about the dynamic ip  Will keep this
>>>>>>> thread up-to-date for anyone who may be in the same boat someday. Thanks
>>>>>>> again for looking at the trace.
>>>>>>> Mario
>>>>>>> 
>>>>>>> You should be setting the req freq to a low number on the outbound gateways
>>>>>>> 
>>>>>>> The examples you showed had a series of inbound reg
>>>>>>> 
>>>>>>> also set expire-seconds to 30 in your gateway xml
>>>>>>> 
>>>>>>> 
>>>>>>> The problem is if you are not constantly sending traffic to the box
>>>>>>> 
>>>>>>> the nat mapping will go away.
>>>>>>> 
>>>>>>> If you are in production you should be using a static ip with a static
>>>>>>> 
>>>>>>> mapping, any trouble you are having is your own fault for playing with
>>>>>>> 
>>>>>>> fire.  The best we can do is tell you how to keep it contained.
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>>> 
>>>>>>> I made the change. I had no idea the settings for the inside phones effected
>>>>>>> nat for the outside sip accounts. I was looking into aggressive-nat-
>>>>>>> detection since the internal profile status always shows the right external
>>>>>>> static IP but the nat_ap status always shows the dynamic ip. Crossing
>>>>>>> fingers/etc since this problem is 85% of time (weeks!) into FS changeover.
>>>>>>> Thanks!
>>>>>>> 
>>>>>>> Mario
>>>>>>> 
>>>>>>> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
>>>>>>> 
>>>>>>> add
>>>>>>> 
>>>>>>> <variable name="sip-force-expires" value="30"/>
>>>>>>> 
>>>>>>> to the <variables> section of your <user>
>>>>>>> 
>>>>>>> you have it at 600 and the nat mapping is timing out while the 600
>>>>>>> 
>>>>>>> seconds is ticking away
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>>> 
>>>>>>> From the TSP:
>>>>>>> 
>>>>>>> "I have enabled the SIP trace on your account. We are not currently seeing
>>>>>>> 
>>>>>>> any registration attempts to your account within the last 15 minutes. Please
>>>>>>> 
>>>>>>> restart FreeSwitch so that registration attempts begin again. Thank you. ".
>>>>>>> 
>>>>>>> So FS is not getting past router.
>>>>>>> 
>>>>>>> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>>>>>>> 
>>>>>>> I ran the global trace during the problem and it is
>>>>>>> 
>>>>>>> at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
>>>>>>> 
>>>>>>> "accttwo", "acct3". The trace includes phones since it was global. I am
>>>>>>> 
>>>>>>> using:
>>>>>>> 
>>>>>>>   <param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>>>>>>> 
>>>>>>>   <param name="ext-sip-ip" value="autonat:my-static.ip"/>
>>>>>>> 
>>>>>>> I tried dumping nat and removing the autonat: above and using -nonat but
>>>>>>> 
>>>>>>> that did not work, registration proceeded but no calls inbound.
>>>>>>> 
>>>>>>> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>>>>>>> 
>>>>>>> Whoops, I am using an IP address for at least one gateway so that is not the
>>>>>>> 
>>>>>>> problem:
>>>>>>> 
>>>>>>> They look outbound to me and I am using dns for 2 and an IP for one so that
>>>>>>> 
>>>>>>> is not the issue. I was able to get FS to clear this up by doing "nat_map
>>>>>>> 
>>>>>>> reinit" which is why I think this is a nat problem. I will do the trace you
>>>>>>> 
>>>>>>> mentioned. I will plug an ip address into one of the gateways to see what
>>>>>>> 
>>>>>>> happens, they all fail at once. Thanks for responding!
>>>>>>> 
>>>>>>> Mario
>>>>>>> 
>>>>>>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>>>>>>> 
>>>>>>> I really need help on this as I have weeks into this problem. I thought I
>>>>>>> 
>>>>>>> had it nailed but I guess not. After 5.5 hours I get:
>>>>>>> 
>>>>>>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>>>>>>> 
>>>>>>> Registration, setting retry to 15 seconds.
>>>>>>> 
>>>>>>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>>>>> 
>>>>>>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>>>>> 
>>>>>>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
>>>>>>> 
>>>>>>> Registration, setting retry to 30 seconds.
>>>>>>> 
>>>>>>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>>>>>>> 
>>>>>>> Registration, setting retry to 15 seconds.
>>>>>>> 
>>>>>>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>>>>>>> 
>>>>>>> Registration, setting retry to 15 seconds.
>>>>>>> 
>>>>>>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>>>>> 
>>>>>>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>>>>> 
>>>>>>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>>>>>>> 
>>>>>>> and no way to make/get calls until I restart FS. I did this:
>>>>>>> 
>>>>>>> 1. log 7
>>>>>>> 
>>>>>>> 2. sofia profile xxxx siptrace on   for each profile/gateway
>>>>>>> 
>>>>>>> 3. restarted router
>>>>>>> 
>>>>>>> All three did not solve the problem. The trace and log produced no
>>>>>>> 
>>>>>>> additional lines which is why I am wondering if FS has a problem since the
>>>>>>> 
>>>>>>> trace shows no SIP activity.
>>>>>>> 
>>>>>>> 3 gateways with 2 ITSPs
>>>>>>> 
>>>>>>> 2 DSL/WAN lines, 1 static and 1 dynamic
>>>>>>> 
>>>>>>> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
>>>>>>> 
>>>>>>> external static ip.
>>>>>>> 
>>>>>>> sofia status profile ... has the right ext ip
>>>>>>> 
>>>>>>> nat_map status shows the dynamic (wrong) IP
>>>>>>> 
>>>>>>> I tried starting with -nonat but that was worse
>>>>>>> 
>>>>>>> the only way to fix is restart FS.
>>>>>>> 
>>>>>>> I read the wiki on external nat, auto_nat and everything else many times.
>>>>>>> 
>>>>>>> Thanks Mario
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> 
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> 
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> 
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> 
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 
>>>>>>> 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
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> 
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> 
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 
>>>>>>> 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
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> 
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 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
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> 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
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Fri Oct 29 16:46:06 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Oct 2010 16:46:06 -0700
Subject: [Freeswitch-users] Mod Event Socket
In-Reply-To: <900263.75973.qm@web55703.mail.re3.yahoo.com>
References: <AANLkTi=6HgNczOBs110tX-ofAO50FbapWZ1P3ascMy0h@mail.gmail.com>
	<900263.75973.qm@web55703.mail.re3.yahoo.com>
Message-ID: <AANLkTinUxgdd9YTK3tDvOhN8AmAqyCeCCs4k08qtCP0h@mail.gmail.com>

Did you properly build the php ESL module?
http://wiki.freeswitch.org/wiki/Esl#Installation
In libs/esl you need to "make" and "make phpmod-install"

Note: If your php script is running on a system that does not have
FreeSWITCH then you will need to download and build ESL on the machine where
the script is running, or you need to move the necessary *.so files over to
that target machine.

Also, I see that you mention that you are running this in a browser. I'm not
sure that is what you need to be doing. I'll defer to more experienced Web
programmers. However this much I can tell you: if your php script isn't
actively running and waiting for a socket connection on port 9090 then this
whole thing will fail completely. Using the socket app from the dialplan
implies that the program on the other end will be sitting around waiting for
a phone call to come in while listening on the port in question and will
actively control the call. The test script you pasted here is the exact
opposite of that: it is a script that connects *to* FreeSWITCH, not one that
sits around waiting for a connection *from* FreeSWITCH.

I think you may need to break down your project into some smaller pieces.
The first thing is to do is define the scope of your project: what all does
it need to do? Does it need to answer a call and control it? If so you need
a daemon of some sort that sits around waiting for calls from FS. Do you
need to update a Web page? If so then you'll need to have some way for the
Web server to communicate either with FreeSWITCH directly or with your
daemon, or with a 3rd party data source, or whatever.

You might want to join #freeswitch at irc.freenode.net so that you can
connect with other FS users and developers. You also should consider
#fusionpbx and #2600hz because both of those channels are frequented by
programmers who are familiar with PHP Web development and FreeSWITCH.

Hope this helps.
-MC

On Fri, Oct 29, 2010 at 1:01 PM, Will Smith <willbelair at yahoo.com> wrote:

> hi Michael,
> Thank you for your input. Really appreciate when having someone giving help
> when in need.
> I bought FS ebook, read chapter 9 and 10. And did what you said:
> 1- put the event_socket_conf.xml back to original state, just modify the ip
> address, because I work through a vpn network. Here , could you confirm that
> mod_event_socket supports different browsers ? This is really important to
> me, because if client could not access from www, there is no point to work
> on this. Thank you.
> -------- the event_socket_conf.xml
> <configuration name="event_socket.conf" description="Socket Client">
>   <settings>
>     <param name="nat-map" value="false"/>
>     <param name="listen-ip" value="192.168.51.133"/>
>     <param name="listen-port" value="8021"/>
>     <param name="password" value="ClueCon"/>
>     <!--param name="apply-inbound-acl" value="lan"/-->
>   </settings>
> </configuration>
> -----------
>
> 2- I put this in both dialplan/default.xml  and dialplan/public.xml
> ------------
> <extension name="outbound-socket">
> <condition field="destination_number" expression="^444$">
> <action application="socket" data="192.168.51.133:8021 async full"/>
> </condition>
> </extension>
> ------------
>
> 3- I have a simple php script just to connect to the FS, this is a sample
> script from FS wiki page
> ------
> <?php
>
> //require_once('ESL.php');
>
> require_once('/usr/src/freeswitch/freeswitch-snapshot/libs/esl/php/ESL.php');
> set_time_limit(0); // Remove the PHP time limit of 30 seconds for
> completion due to loop watching events
> echo "HERE";
> // Connect to FreeSWITCH
> $sock = new ESLconnection('localhost', '8021', 'ClueCon');
> // We want all Events (probably will want to change this depending on your
> needs)
> $sock->sendRecv("event plain ALL");
>
>
> // Grab Events until process is killed
> //while($sock->connected()){
> //echo "CONECTED";
> //$event = $sock->recvEvent();
> //print_r($event->serialize());
> //}
>
> ?>
> ------
>
> when running this from web browser, I got blank page. So, I run from
> console, this is the error messages:
> --------
> PHP Warning:  dl(): Unable to load dynamic library
> '/usr/lib/php/modules/ESL.so' - /usr/lib/php/modules/ESL.so: cannot open
> shared object file: No such file or directory in
> /usr/src/freeswitch/freeswitch-snapshot/libs/esl/php/ESL.php on line 23
> PHP Stack trace:
> PHP   1. {main}() /var/www/html/eventsocket.php:0
> PHP   2. require_once() /var/www/html/eventsocket.php:4
> PHP   3. dl()
> /usr/src/freeswitch/freeswitch-snapshot/libs/esl/php/ESL.php:23
> HEREPHP Fatal error:  Call to undefined function new_ESLconnection() in
> /usr/src/freeswitch/freeswitch-snapshot/libs/esl/php/ESL.php on line 119
> PHP Stack trace:
> PHP   1. {main}() /var/www/html/eventsocket.php:0
> PHP   2. ESLconnection->__construct() /var/www/html/eventsocket.php:8
> ----------
>
> So, I guess the ESL.so is missing. I tried to  "locate ESL.so" , but
> nothing returned. I tried to
> --------
>
> yum install libxml2-devel pcre-devel bzip2-devel curl-devel gmp-devel aspell-devel php-devel libtermcap-devel gdbm-devel
>  db4-devel
>
> --------
> it ran fine. But I did not do this:
> -------
> *NOTE:* For PHP you must edit the libs/esl/php/Makefile and add -lpthread
> to the LOCAL_LDFLAGS line.
> -------
> because I could not fine the LOCAL_LDFLAGS line.
>
> Please give me some help, I am banging my head against the wall here. If I
> could get the file ESL.so , is the set up above working? I thought it would
> be simple just to connect.
>
> Thank you.
>
> --- On *Thu, 10/28/10, Michael Collins <msc at freeswitch.org>* wrote:
>
>
> From: Michael Collins <msc at freeswitch.org>
> Subject: Re: [Freeswitch-users] Mod Event Socket
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Received: Thursday, October 28, 2010, 5:57 PM
>
>
> Be sure to learn the difference between inbound and outbound event socket.
> In your case you are doing outbound event socket. The dialplan calls the
> socket app which does an outbound socket connection to port 9090 on the
> localhost. Make sure that your app is listening on port 9090.
>
> The event_socket.conf.xml file is for configuring inbound socket
> connections, e.g. when you have a script that attempts to connect to FS
> where FS is listening on port 8021. So change your event_socket.conf.xml
> file back to the default and retry.
>
> Report back here if you have trouble. Also, buy or borrow the FreeSWITCH
> book and check out chapter 9. Lots of good stuff there on how the even
> system works.
>
> -MC
>
> On Thu, Oct 28, 2010 at 2:14 PM, Will Smith <willbelair at yahoo.com<http://mc/compose?to=willbelair at yahoo.com>
> > wrote:
>
> Hi,
> I am new to FS, and trying to get the mod event socket installed and
> running. I have FS running, with SIP account, can dial in/out via gateway.
> Now I want to dial in, FS will send some info to client browser, here is
> one question, does this work over the internet, or just local ? Info sent
> could be the uuid, so that client browser could decide to bridge the call,
> send to IVR or transfer ...
> Please give me some guide line how to set this up.
> I added to dialplan/default.xml
> -----
>
>  <extension name="socketTest">
>       <condition field="destination_number" expression="^444$">
>         <action application="socket" data="127.0.0.1:9090 async full"/>
>       </condition>
>  </extension>
>
> -----
>
>
> got the php sample file:
>
> <?php
>
> /**
>  * Based loosely on the NET_Server code in PEAR.
>  * This is only an example - considerable additional work is needed
>  * Specifically, the code in the handleConnection method should be
>  * handled in a subclass
>  *
>  *
>  */
>
> class Message
> {
>   var $properties = array();
>   var $content = null;
> }
>
> class EventSocketListener
> {
>
>   var $host;
>   var $port;
>   var $sock;
>   var $is_parent = true;
>   var $clientInfo;
>   var $clientFD;
>
>   var $connectionContext = array();
>
>
>   function &create($port, $host = "localhost")
>       {
> 	$esl = new EventSocketListener;
> 	$esl->port = $port;
> 	$esl->host = $host;
>         if (!function_exists('socket_create')) {
>             return PEAR::raiseError('Sockets extension not available.');
>         }
> 	return $esl;
>       }
>
>   function start()
>   {
> 	if (($this->sock = socket_create(AF_INET, SOCK_STREAM, SOL_TCP)) === false) {
> 	  echo "socket_create() failed: reason: " . socket_strerror(socket_last_error()) . "\n";
> 	}
>
> 	if (!socket_set_option($this->sock, SOL_SOCKET, SO_REUSEADDR, 1)) {
> 	  echo 'Unable to set option on socket: '. socket_strerror(socket_last_error()) . PHP_EOL;
> 	}
>
> 	if (socket_bind($this->sock, $this->host, $this->port) === false) {
> 	  echo "socket_bind() failed: reason: " . socket_strerror(socket_last_error($this->sock)) . "\n";
> 	}
>
> 	if (socket_listen($this->sock, 5) === false) {
> 	  echo "socket_listen() failed: reason: " . socket_strerror(socket_last_error($this->sock)) . "\n";
> 	}
>
>         // Dear children, please do not become zombies
>         pcntl_signal(SIGCHLD, SIG_IGN);
>
>         // wait for incmoning connections
>         while (true)
>         {
>             // new connection
>             if(($fd = socket_accept($this->sock)))
>             {
> 	      $pid = pcntl_fork();
> 	      if($pid == -1) {
> 		  return  PEAR::raiseError('Could not fork child process.');
>                 }
>                 // This is the child => handle the request
>                 elseif($pid == 0) {
>                     // this is not the parent
>                     $this->_isParent = false;
>                     // store the new file descriptor
>                     $this->clientFD = $fd;
>
>                     $peer_host    =    "";
>                     $peer_port    =    "";
>                     socket_getpeername($this->clientFD, $peer_host, $peer_port);
>                     $this->clientInfo    =    array(
>                                                       "host"        =>    $peer_host,
>                                                       "port"        =>    $peer_port,
>                                                       "connectOn"   =>    time()
>                                                    );
> 		    $this->handleConnection();
> 		    socket_shutdown($this->clientFD, 2);
> 		    socket_close($this->clientFD);
> 		}
> 		else /* Parent does nothing */
> 		  {
> 		  }
> 	    }
> 	}
>   }
>
>   function handleConnection()
>   {
>     $fd = $this->clientFD;
>     //first, read headers & setup a state for this connection
>     $line = "";
>     socket_write($fd, "CONNECT\n\n");
>     do
>       {
> 	$line = socket_read($fd, 2048, PHP_NORMAL_READ);
> 	if (trim($line) == "")
> 	  break;
> 	//we got a header, we need to add it to the
> 	list($key, $value) = explode(":", $line);
> 	$key = trim($key);
> 	$value = trim(urldecode($value));
> 	$this->connectionContext[$key] = $value;
>       }
>     while ($line != "");
>
>     //    print_r($this->connectionContext);
>     $this->callConnected();
>
>     exit();
>
>   }
>
>   function processMessages($returnOnReply = false)
>   {
>     $fd = $this->clientFD;
>     $result = new Message();
>     $props = array();
>     while (true)
>       {
> 	do
> 	  {
> 	    $line = @socket_read($fd, 2048, PHP_NORMAL_READ);
> 	    if (socket_last_error($fd) == 104)
> 	      return null;
> 	    if ($line == null || $line == FALSE || trim($line) == "")
> 	      break;
> 	    //we got a header, we need to add it to the message
> 	    list($key, $value) = explode(":", $line);
> 	    $key = trim($key);
> 	    $value = trim(urldecode($value));
> 	    $props[$key] = $value;
> 	  }
> 	while ($line != "");
> 	$result->properties = $props;
>
> 	if (isset($props['Content-Length']))
> 	  {
> 	    $length = $props['Content-Length'];
> 	    print("Reading content - $length\n");
> 	    $data = socket_read($fd, $length);
> 	    $result->content = $data;
> 	  }
> 	if (isset($props['Content-Type']))
> 	  {
> 	    $type = $props['Content-Type'];
> 	    if ($returnOnReply &&
> 		($type == "command/reply" || $type == "api/response"))
> 	      {
> 		return $result;
> 	      }
> 	    else if ($type == "text/event-plain") //only plain events for now
> 	      {
> 		$this->handleEvent($result);
> 	      }
> 	  }
> 	else
> 	  {
> 	    print("UNKNOWN MESSAGE: \n");
> 	    print_r($result);
> 	  }
>       }
>   }
>
>
>
>   function invokeCommand($command)
>   {
>     //Send the command
>     print("Invoking: $command\n");
>     $this->sendCommand($command);
>     // Wait for the response
>     $result = $this->processMessages(true);
>     return $result;
>   }
>
>   function sendCommand($command)
>   {
>     $fd = $this->clientFD;
>     socket_write($fd, trim($command) . "\n\n");
>   }
>
> /*-----------------------------------------------------*/
>   /* Abstract Methods - should move to subclass*/
>
>   function callConnected()
>   {
>     print_r($this->connectionContext);
>     print("----------------\n");
>     $result = $this->invokeCommand("log DEBUG");
>     print_r($result);
>     $result = $this->invokeCommand("event plain ALL");
>     print_r($result);
>
>     $this->processMessages(false);
>     print("DONE PROCESSING MESSAGES");
>     print_r($this->connectionContext);
>   }
>
>   function handleCommandResponse($response)
>   {
>     print("Recieved Unhandled Response:\n");
>     print_r($response);
>   }
>
>   function handleEvent($event)
>   {
>     print("Recieved Unhandled Event:\n");
>     print_r($event);
>   }
> }
>
>
>
>
> // create a server that forks new processes
> $server  = &EventSocketListener::create(9090);
>
> // start the server
> $server->start();
> ?>
>
> --------
>
> ( this is the original file, not perfect sample) I tried to run this, and got error with auth.
>
> Also, I modify the even_socket_conf.xml  in autoload_configs/
>
> change listen-ip to  0.0.0.0 , port = 9090 , disable password
>
>
> What did I miss?
>
> Thankyou for your help
>
> Will
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> -----Inline Attachment Follows-----
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Fri Oct 29 17:17:04 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 29 Oct 2010 19:17:04 -0500
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <86E8AE3F-E945-4952-A438-C674F23881D6@mgtech.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
	<407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>
	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
	<AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
	<6D9FD5E2-21FA-4630-8CF4-A87ADB30AC87@mgtech.com>
	<AANLkTi=BiDiXLgEoj_MfwmZbGjvXoOuNJgUYdetciNCj@mail.gmail.com>
	<9BAFDAF8-1E66-444C-A024-A88FF65D8BCA@mgtech.com>
	<3447E946-0BF8-4F7E-B5E9-E28F53DD0968@mgtech.com>
	<AANLkTi=LRbz6ADhLvBKkjGxOywAVLf--GTkzXEpUoDbi@mail.gmail.com>
	<BE754258-BE27-454F-B740-527A35DBDFFE@mgtech.com>
	<AANLkTi=3Kd-bCHJSED7VEh-cvtpdXMs_EkWZL=s8J4Ux@mail.gmail.com>
	<14205291-75B3-470F-ABA1-EE7F08A695E9@mgtech.com>
	<AANLkTi=D6siEAquSq6gYh-RnZeJJWrk11mHWK8De2tLh@mail.gmail.com>
	<2C35DBE8-13B3-4264-BB43-7496A1054AD1@mgtech.com>
	<86E8AE3F-E945-4952-A438-C674F23881D6@mgtech.com>
Message-ID: <AANLkTin5ni_P-D5+XnCKugoRLgoExACV=eBWo3496AXq@mail.gmail.com>

can you repeat that trace with sofia debug on
sofia loglevel all 9

Are you doing DNS by any chance in the gateway "proxy" param?
you could try filling in the register-proxy param in your gateway to
sip:<ip> <-- not dns but ip that dns resolves to
<param name="register-proxy" value="sip:1.2.3.4"/>

I'm just guessing but Its possible some bad dns query could be throwing FS off.
so this test would force all the packets to the exact IP of your host
instead of looking it up.



On Fri, Oct 29, 2010 at 6:30 PM, Mario G <mario_fs at mgtech.com> wrote:
> Oh Boy: Not only is everything set to the static route but I turned
> off/disconnected the dynamic DSL line so I only had 1 static line to the
> router. The second wan is set in the router off. Even turned off router uPnP
> even though I am using -nonat. Guess what.... I still have the problem. Look
> like when FS says is going to retry it actually does not.
> Here is a short one:?http://pastebin.freeswitch.org/14359?- I caught one
> right after I started FS, failure occurred in minutes (lucky). Look at the
> bottom of the trace, you see SIP trace activity and then when it fails no
> SIP trace activity. Could this possibly be a FS bug? (I am a mainframe
> assembler systems programmer and I might think so if there were error retry
> messages but nothing showing in one of my traces...)
> Notes:
> idone is gateway 1
> idtwo is gateway 2 ?I had to trace both because it was impossible to figure
> out which one would fail first. Ran several times but kept missing the right
> one.
> I use a url for one gateway and ip for another but it makes no difference
> since both eventually fail.
> 10. is local lan
> 210. is external ip
> 216. is itsp
>
> Here is a longer one from earlier ??http://pastebin.freeswitch.org/14357
> Notes:
> A call was received and hung up for idtwo - beginning of trace
> 11 minutes later idtwo failed - see last line of trace
>
> Thank you very much!
> Mario
>
> On Oct 29, 2010, at 12:11 PM, Anthony Minessale wrote:
>
> stun-enabled must be true in your profile XML to see what you pasted.
>
> Get me a sip trace of this from when it works until when it fails
>
> only enable the sip trace on the profile with the gateway to reduce traffic
>
>
>
> On Fri, Oct 29, 2010 at 1:56 PM, Mario G <mario_fs at mgtech.com> wrote:
>
> Oh my... ?looks it was not NAT after all? Please help! I changed to all
> profiles to static per instructions below and still have the problem:
>
> 2010-10-29 11:15:18.536446 [NOTICE] sofia_reg.c:342 Registering uuid1
>
> 2010-10-29 11:15:34.313150 [WARNING] sofia_reg.c:387 uuid1 Failed
> Registration, setting retry to 15 seconds.
>
> sofia global siptrace on did not show any activity for this gateway in or
> out, others were fine but eventually fail. I setup static:
>
> 1. set the params ext-sip-ip and ext-rtp-ip to my external static IP
>
> 2. map the sip ports (5060-5080) and all of the rtp ports (16384-32767) ?to
> FS lan addr.
>
> 3. set sip-ip and rtp-ip to the lan addr of FS
>
> 4. start FS with -nonat
>
> I don't know what to try next. BTW, the sofia status for the profiles shows
> stun enabled but I did not set it up anywhere:
>
> Name ???????????????????uuid1
>
> Domain Name ????????????N/A
>
> Auto-NAT ???????????????false
>
> DBName ?????????????????sofia_reg_mvvyl
>
> Pres Hosts
>
> Dialplan ???????????????XML
>
> Context ????????????????public
>
> Challenge Realm ????????auto_to
>
> RTP-IP ?????????????????10.x.x.20
>
> Ext-RTP-IP ?????????????210.x.x.100
>
> SIP-IP ?????????????????10.x.x.20
>
> Ext-SIP-IP ?????????????210.x.x.100
>
> URL ????????????????????sip:mod_sofia at 210.x.x.100:5068
>
> BIND-URL ???????????????sip:mod_sofia at 210.x.x.100:5068;maddr=10.x.x.20
>
> HOLD-MUSIC ?????????????local_stream://moh
>
> OUTBOUND-PROXY ?????????N/A
>
> CODECS IN ??????????????G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
>
> CODECS OUT ?????????????PCMU,PCMA,GSM
>
> TEL-EVENT ??????????????101
>
> DTMF-MODE ??????????????rfc2833
>
> CNG ????????????????????13
>
> SESSION-TO ?????????????0
>
> MAX-DIALOG ?????????????0
>
> NOMEDIA ????????????????false
>
> LATE-NEG ???????????????false
>
> PROXY-MEDIA ????????????false
>
> AGGRESSIVENAT ??????????false
>
> STUN-ENABLED ???????????true
>
> STUN-AUTO-DISABLE ??????false
>
> CALLS-IN ???????????????0
>
> FAILED-CALLS-IN ????????0
>
> CALLS-OUT ??????????????0
>
> FAILED-CALLS-OUT ???????0
>
>
> On Oct 27, 2010, at 10:04 AM, Anthony Minessale wrote:
>
> if you map it or not, a scanner would penetrate it.
>
> There are lot of sip scanners out there now, you just need to beware of
> them.
>
>
> On Wed, Oct 27, 2010 at 11:50 AM, Mario G <mario_fs at mgtech.com> wrote:
>
> Thanks so much! I am sure many others will find this info invaluable. I will
> try the static route again but have one question: When I started with FS I
> found a "sip scanner" ?in FS and someone on this group said not to use port
> mapping since it was a security risk. Is that true?
>
> On Oct 27, 2010, at 9:10 AM, Anthony Minessale wrote:
>
> you are completely guessing at things.
>
> I want you to understand that the only reason you are having problems
>
> with this is because you don't understand how it works enough to know
>
> what you are doing 100%
>
> Its a given that the pnp stuff is only for your dynamic IP.
>
> aggressive-nat-detection and sip-force-expires are all related to
>
> inbound calls when the things who are registering to you may be behind
>
> nat.
>
> You need to learn the difference between which nat tools are
>
> *) designed for your FS to run behind nat
>
> *) designed for FS to run public and accept connections from devices behind
> nat.
>
> If you have a static IP, you don't need the pnp stuff so -nonat is fine
>
> What you need to do is set
>
> 1) set the params ext-sip-ip and ext-rtp-ip to your external static IP
>
> 2) map the sip ports and all of the rtp ports from your static IP to FS lan
> addr
>
> 3) set sip-ip and rtp-ip to the lan addr you forwarded through.
>
>
> If you don't do this: your outbound registration will use NAT to your
>
> provider and if there is no activity for the expire time on your NAT
>
> mapping the reverse port mapping from your provider back to you is
>
> lost. ?This is why you set your register expires to a very low number,
>
> (you need to make sure the provider does not turn the expires back up
>
> in the reply because it will beat your choice *see sip trace) if this
>
> is the case then you need the "ping" option set to 30, to continuously
>
> send an options to your provider.
>
> The static mapping is obviously the better, easier and more reliable
> solution.
>
> So I want you to understand that the only way to keep a nat mapped
>
> port alive is to continuously send traffic, all the other methods that
>
> you are mentioning are to detect that phones registered to your are
>
> behind nat, I gave you that force-expires option before because your
>
> trace was full of inbound reg so I thought that is what you wanted
>
> help with.
>
>
>
>
>
>
>
>
> On Wed, Oct 27, 2010 at 10:43 AM, Mario G <mario_fs at mgtech.com> wrote:
>
> I should mention that I did not have this problem with an SPA9000 PBX
>
> (asterisk based) for over two years so FS may be pickier about upnp and/or
>
> nat, or just better at it exposing a problem in the router.
>
> I made different changes to the gateways to test different things. One
>
> failed after 17 hours, the other two stayed up. ?What did not work:
>
> added <variable name="sip-force-expires" value="30"/> to the directory
>
> entries as suggested.
>
> set the gateway expire times to 30 seconds.
>
> What worked (could be coincidental) for the two gateways that stayed up:
>
> I Added <param name="aggressive-nat-detection" value="true"/>
>
> I originally setup FS to use the static ip by setting external sip/rtp to
>
> just the static ip (no autonat:) and ran with -nonat but I could not get
>
> incoming calls. The only way it worked was to use autonat:1.2.3.4. The
>
> router has 1 static public address and 1 dynamic external IP, this is the
>
> root of the problem, upnp only tells FS about the dynamic ip ?Will keep this
>
> thread up-to-date for anyone who may be in the same boat someday. Thanks
>
> again for looking at the trace.
>
> Mario
>
> You should be setting the req freq to a low number on the outbound gateways
>
> The examples you showed had a series of inbound reg
>
> also set expire-seconds to 30 in your gateway xml
>
>
> The problem is if you are not constantly sending traffic to the box
>
> the nat mapping will go away.
>
> If you are in production you should be using a static ip with a static
>
> mapping, any trouble you are having is your own fault for playing with
>
> fire. ?The best we can do is tell you how to keep it contained.
>
>
>
>
> On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com> wrote:
>
> I made the change. I had no idea the settings for the inside phones effected
>
> nat for the outside sip accounts. I was looking into aggressive-nat-
>
> detection since the internal profile status always shows the right external
>
> static IP but the nat_ap status always shows the dynamic ip. Crossing
>
> fingers/etc since this problem is 85% of time (weeks!) into FS changeover.
>
> Thanks!
>
> Mario
>
> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
>
> add
>
> <variable name="sip-force-expires" value="30"/>
>
> to the <variables> section of your <user>
>
> you have it at 600 and the nat mapping is timing out while the 600
>
> seconds is ticking away
>
>
>
> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
>
> From the TSP:
>
> "I have enabled the SIP trace on your account. We are not currently seeing
>
> any registration attempts to your account within the last 15 minutes. Please
>
> restart FreeSwitch so that registration attempts begin again. Thank you. ".
>
> So FS is not getting past router.
>
> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>
> I ran the global trace during the problem and it is
>
> at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
>
> "accttwo", "acct3". The trace includes phones since it was global. I am
>
> using:
>
> ??<param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>
> ??<param name="ext-sip-ip" value="autonat:my-static.ip"/>
>
> I tried dumping nat and removing the autonat: above and using -nonat but
>
> that did not work, registration proceeded but no calls inbound.
>
> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>
> Whoops, I am using an IP address for at least one gateway so that is not the
>
> problem:
>
> They look outbound to me and I am using dns for 2 and an IP for one so that
>
> is not the issue. I was able to get FS to clear this up by doing "nat_map
>
> reinit" which is why I think this is a nat problem. I will do the trace you
>
> mentioned. I will plug an ip address into one of the gateways to see what
>
> happens, they all fail at once. Thanks for responding!
>
> Mario
>
> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>
> I really need help on this as I have weeks into this problem. I thought I
>
> had it nailed but I guess not. After 5.5 hours I get:
>
> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>
> Registration, setting retry to 15 seconds.
>
> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>
> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>
> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
>
> Registration, setting retry to 30 seconds.
>
> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>
> Registration, setting retry to 15 seconds.
>
> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>
> Registration, setting retry to 15 seconds.
>
> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>
> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>
> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>
> and no way to make/get calls until I restart FS. I did this:
>
> 1. log 7
>
> 2. sofia profile xxxx siptrace on ??for each profile/gateway
>
> 3. restarted router
>
> All three did not solve the problem. The trace and log produced no
>
> additional lines which is why I am wondering if FS has a problem since the
>
> trace shows no SIP activity.
>
> 3 gateways with 2 ITSPs
>
> 2 DSL/WAN lines, 1 static and 1 dynamic
>
> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
>
> external static ip.
>
> sofia status profile ... has the right ext ip
>
> nat_map status shows the dynamic (wrong) IP
>
> I tried starting with -nonat but that was worse
>
> the only way to fix is restart FS.
>
> I read the wiki on external nat, auto_nat and everything else many times.
>
> Thanks Mario
>
>
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>
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From mario_fs at mgtech.com  Fri Oct 29 17:29:16 2010
From: mario_fs at mgtech.com (Mario G)
Date: Fri, 29 Oct 2010 17:29:16 -0700
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <AANLkTin5ni_P-D5+XnCKugoRLgoExACV=eBWo3496AXq@mail.gmail.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
	<407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>
	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
	<AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
	<6D9FD5E2-21FA-4630-8CF4-A87ADB30AC87@mgtech.com>
	<AANLkTi=BiDiXLgEoj_MfwmZbGjvXoOuNJgUYdetciNCj@mail.gmail.com>
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Will so, BTW, I mentioned below one is dns and another uses IP, I tested that theory using an IP  does not help. I also updated to todays git version and no help there. Will post when the trace is done.

On Oct 29, 2010, at 5:17 PM, Anthony Minessale wrote:

> can you repeat that trace with sofia debug on
> sofia loglevel all 9
> 
> Are you doing DNS by any chance in the gateway "proxy" param?
> you could try filling in the register-proxy param in your gateway to
> sip:<ip> <-- not dns but ip that dns resolves to
> <param name="register-proxy" value="sip:1.2.3.4"/>
> 
> I'm just guessing but Its possible some bad dns query could be throwing FS off.
> so this test would force all the packets to the exact IP of your host
> instead of looking it up.
> 
> 
> 
> On Fri, Oct 29, 2010 at 6:30 PM, Mario G <mario_fs at mgtech.com> wrote:
>> Oh Boy: Not only is everything set to the static route but I turned
>> off/disconnected the dynamic DSL line so I only had 1 static line to the
>> router. The second wan is set in the router off. Even turned off router uPnP
>> even though I am using -nonat. Guess what.... I still have the problem. Look
>> like when FS says is going to retry it actually does not.
>> Here is a short one: http://pastebin.freeswitch.org/14359 - I caught one
>> right after I started FS, failure occurred in minutes (lucky). Look at the
>> bottom of the trace, you see SIP trace activity and then when it fails no
>> SIP trace activity. Could this possibly be a FS bug? (I am a mainframe
>> assembler systems programmer and I might think so if there were error retry
>> messages but nothing showing in one of my traces...)
>> Notes:
>> idone is gateway 1
>> idtwo is gateway 2  I had to trace both because it was impossible to figure
>> out which one would fail first. Ran several times but kept missing the right
>> one.
>> I use a url for one gateway and ip for another but it makes no difference
>> since both eventually fail.
>> 10. is local lan
>> 210. is external ip
>> 216. is itsp
>> 
>> Here is a longer one from earlier   http://pastebin.freeswitch.org/14357
>> Notes:
>> A call was received and hung up for idtwo - beginning of trace
>> 11 minutes later idtwo failed - see last line of trace
>> 
>> Thank you very much!
>> Mario
>> 
>> On Oct 29, 2010, at 12:11 PM, Anthony Minessale wrote:
>> 
>> stun-enabled must be true in your profile XML to see what you pasted.
>> 
>> Get me a sip trace of this from when it works until when it fails
>> 
>> only enable the sip trace on the profile with the gateway to reduce traffic
>> 
>> 
>> 
>> On Fri, Oct 29, 2010 at 1:56 PM, Mario G <mario_fs at mgtech.com> wrote:
>> 
>> Oh my...  looks it was not NAT after all? Please help! I changed to all
>> profiles to static per instructions below and still have the problem:
>> 
>> 2010-10-29 11:15:18.536446 [NOTICE] sofia_reg.c:342 Registering uuid1
>> 
>> 2010-10-29 11:15:34.313150 [WARNING] sofia_reg.c:387 uuid1 Failed
>> Registration, setting retry to 15 seconds.
>> 
>> sofia global siptrace on did not show any activity for this gateway in or
>> out, others were fine but eventually fail. I setup static:
>> 
>> 1. set the params ext-sip-ip and ext-rtp-ip to my external static IP
>> 
>> 2. map the sip ports (5060-5080) and all of the rtp ports (16384-32767)  to
>> FS lan addr.
>> 
>> 3. set sip-ip and rtp-ip to the lan addr of FS
>> 
>> 4. start FS with -nonat
>> 
>> I don't know what to try next. BTW, the sofia status for the profiles shows
>> stun enabled but I did not set it up anywhere:
>> 
>> Name                    uuid1
>> 
>> Domain Name             N/A
>> 
>> Auto-NAT                false
>> 
>> DBName                  sofia_reg_mvvyl
>> 
>> Pres Hosts
>> 
>> Dialplan                XML
>> 
>> Context                 public
>> 
>> Challenge Realm         auto_to
>> 
>> RTP-IP                  10.x.x.20
>> 
>> Ext-RTP-IP              210.x.x.100
>> 
>> SIP-IP                  10.x.x.20
>> 
>> Ext-SIP-IP              210.x.x.100
>> 
>> URL                     sip:mod_sofia at 210.x.x.100:5068
>> 
>> BIND-URL                sip:mod_sofia at 210.x.x.100:5068;maddr=10.x.x.20
>> 
>> HOLD-MUSIC              local_stream://moh
>> 
>> OUTBOUND-PROXY          N/A
>> 
>> CODECS IN               G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
>> 
>> CODECS OUT              PCMU,PCMA,GSM
>> 
>> TEL-EVENT               101
>> 
>> DTMF-MODE               rfc2833
>> 
>> CNG                     13
>> 
>> SESSION-TO              0
>> 
>> MAX-DIALOG              0
>> 
>> NOMEDIA                 false
>> 
>> LATE-NEG                false
>> 
>> PROXY-MEDIA             false
>> 
>> AGGRESSIVENAT           false
>> 
>> STUN-ENABLED            true
>> 
>> STUN-AUTO-DISABLE       false
>> 
>> CALLS-IN                0
>> 
>> FAILED-CALLS-IN         0
>> 
>> CALLS-OUT               0
>> 
>> FAILED-CALLS-OUT        0
>> 
>> 
>> On Oct 27, 2010, at 10:04 AM, Anthony Minessale wrote:
>> 
>> if you map it or not, a scanner would penetrate it.
>> 
>> There are lot of sip scanners out there now, you just need to beware of
>> them.
>> 
>> 
>> On Wed, Oct 27, 2010 at 11:50 AM, Mario G <mario_fs at mgtech.com> wrote:
>> 
>> Thanks so much! I am sure many others will find this info invaluable. I will
>> try the static route again but have one question: When I started with FS I
>> found a "sip scanner"  in FS and someone on this group said not to use port
>> mapping since it was a security risk. Is that true?
>> 
>> On Oct 27, 2010, at 9:10 AM, Anthony Minessale wrote:
>> 
>> you are completely guessing at things.
>> 
>> I want you to understand that the only reason you are having problems
>> 
>> with this is because you don't understand how it works enough to know
>> 
>> what you are doing 100%
>> 
>> Its a given that the pnp stuff is only for your dynamic IP.
>> 
>> aggressive-nat-detection and sip-force-expires are all related to
>> 
>> inbound calls when the things who are registering to you may be behind
>> 
>> nat.
>> 
>> You need to learn the difference between which nat tools are
>> 
>> *) designed for your FS to run behind nat
>> 
>> *) designed for FS to run public and accept connections from devices behind
>> nat.
>> 
>> If you have a static IP, you don't need the pnp stuff so -nonat is fine
>> 
>> What you need to do is set
>> 
>> 1) set the params ext-sip-ip and ext-rtp-ip to your external static IP
>> 
>> 2) map the sip ports and all of the rtp ports from your static IP to FS lan
>> addr
>> 
>> 3) set sip-ip and rtp-ip to the lan addr you forwarded through.
>> 
>> 
>> If you don't do this: your outbound registration will use NAT to your
>> 
>> provider and if there is no activity for the expire time on your NAT
>> 
>> mapping the reverse port mapping from your provider back to you is
>> 
>> lost.  This is why you set your register expires to a very low number,
>> 
>> (you need to make sure the provider does not turn the expires back up
>> 
>> in the reply because it will beat your choice *see sip trace) if this
>> 
>> is the case then you need the "ping" option set to 30, to continuously
>> 
>> send an options to your provider.
>> 
>> The static mapping is obviously the better, easier and more reliable
>> solution.
>> 
>> So I want you to understand that the only way to keep a nat mapped
>> 
>> port alive is to continuously send traffic, all the other methods that
>> 
>> you are mentioning are to detect that phones registered to your are
>> 
>> behind nat, I gave you that force-expires option before because your
>> 
>> trace was full of inbound reg so I thought that is what you wanted
>> 
>> help with.
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> On Wed, Oct 27, 2010 at 10:43 AM, Mario G <mario_fs at mgtech.com> wrote:
>> 
>> I should mention that I did not have this problem with an SPA9000 PBX
>> 
>> (asterisk based) for over two years so FS may be pickier about upnp and/or
>> 
>> nat, or just better at it exposing a problem in the router.
>> 
>> I made different changes to the gateways to test different things. One
>> 
>> failed after 17 hours, the other two stayed up.  What did not work:
>> 
>> added <variable name="sip-force-expires" value="30"/> to the directory
>> 
>> entries as suggested.
>> 
>> set the gateway expire times to 30 seconds.
>> 
>> What worked (could be coincidental) for the two gateways that stayed up:
>> 
>> I Added <param name="aggressive-nat-detection" value="true"/>
>> 
>> I originally setup FS to use the static ip by setting external sip/rtp to
>> 
>> just the static ip (no autonat:) and ran with -nonat but I could not get
>> 
>> incoming calls. The only way it worked was to use autonat:1.2.3.4. The
>> 
>> router has 1 static public address and 1 dynamic external IP, this is the
>> 
>> root of the problem, upnp only tells FS about the dynamic ip  Will keep this
>> 
>> thread up-to-date for anyone who may be in the same boat someday. Thanks
>> 
>> again for looking at the trace.
>> 
>> Mario
>> 
>> You should be setting the req freq to a low number on the outbound gateways
>> 
>> The examples you showed had a series of inbound reg
>> 
>> also set expire-seconds to 30 in your gateway xml
>> 
>> 
>> The problem is if you are not constantly sending traffic to the box
>> 
>> the nat mapping will go away.
>> 
>> If you are in production you should be using a static ip with a static
>> 
>> mapping, any trouble you are having is your own fault for playing with
>> 
>> fire.  The best we can do is tell you how to keep it contained.
>> 
>> 
>> 
>> 
>> On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com> wrote:
>> 
>> I made the change. I had no idea the settings for the inside phones effected
>> 
>> nat for the outside sip accounts. I was looking into aggressive-nat-
>> 
>> detection since the internal profile status always shows the right external
>> 
>> static IP but the nat_ap status always shows the dynamic ip. Crossing
>> 
>> fingers/etc since this problem is 85% of time (weeks!) into FS changeover.
>> 
>> Thanks!
>> 
>> Mario
>> 
>> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
>> 
>> add
>> 
>> <variable name="sip-force-expires" value="30"/>
>> 
>> to the <variables> section of your <user>
>> 
>> you have it at 600 and the nat mapping is timing out while the 600
>> 
>> seconds is ticking away
>> 
>> 
>> 
>> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
>> 
>> From the TSP:
>> 
>> "I have enabled the SIP trace on your account. We are not currently seeing
>> 
>> any registration attempts to your account within the last 15 minutes. Please
>> 
>> restart FreeSwitch so that registration attempts begin again. Thank you. ".
>> 
>> So FS is not getting past router.
>> 
>> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>> 
>> I ran the global trace during the problem and it is
>> 
>> at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
>> 
>> "accttwo", "acct3". The trace includes phones since it was global. I am
>> 
>> using:
>> 
>>   <param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>> 
>>   <param name="ext-sip-ip" value="autonat:my-static.ip"/>
>> 
>> I tried dumping nat and removing the autonat: above and using -nonat but
>> 
>> that did not work, registration proceeded but no calls inbound.
>> 
>> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>> 
>> Whoops, I am using an IP address for at least one gateway so that is not the
>> 
>> problem:
>> 
>> They look outbound to me and I am using dns for 2 and an IP for one so that
>> 
>> is not the issue. I was able to get FS to clear this up by doing "nat_map
>> 
>> reinit" which is why I think this is a nat problem. I will do the trace you
>> 
>> mentioned. I will plug an ip address into one of the gateways to see what
>> 
>> happens, they all fail at once. Thanks for responding!
>> 
>> Mario
>> 
>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>> 
>> I really need help on this as I have weeks into this problem. I thought I
>> 
>> had it nailed but I guess not. After 5.5 hours I get:
>> 
>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>> 
>> Registration, setting retry to 15 seconds.
>> 
>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>> 
>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>> 
>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
>> 
>> Registration, setting retry to 30 seconds.
>> 
>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>> 
>> Registration, setting retry to 15 seconds.
>> 
>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>> 
>> Registration, setting retry to 15 seconds.
>> 
>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>> 
>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>> 
>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>> 
>> and no way to make/get calls until I restart FS. I did this:
>> 
>> 1. log 7
>> 
>> 2. sofia profile xxxx siptrace on   for each profile/gateway
>> 
>> 3. restarted router
>> 
>> All three did not solve the problem. The trace and log produced no
>> 
>> additional lines which is why I am wondering if FS has a problem since the
>> 
>> trace shows no SIP activity.
>> 
>> 3 gateways with 2 ITSPs
>> 
>> 2 DSL/WAN lines, 1 static and 1 dynamic
>> 
>> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
>> 
>> external static ip.
>> 
>> sofia status profile ... has the right ext ip
>> 
>> nat_map status shows the dynamic (wrong) IP
>> 
>> I tried starting with -nonat but that was worse
>> 
>> the only way to fix is restart FS.
>> 
>> I read the wiki on external nat, auto_nat and everything else many times.
>> 
>> Thanks Mario
>> 
>> 
>> _______________________________________________
>> 
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>> 
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>> 
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>> 
>> 
>> --
>> 
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> 
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>> 
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
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>> 
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>> 
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>> 
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>> 
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>> 
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>> 
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>> 
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>> --
>> 
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> 
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>> Twitter: http://twitter.com/FreeSWITCH_wire
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>> 
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>> 
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>> 
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>> 
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>> 
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>> 
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>> 
>> FreeSWITCH http://www.freeswitch.org/
>> 
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>> Twitter: http://twitter.com/FreeSWITCH_wire
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>> 
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>> 
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>> 
>> FreeSWITCH http://www.freeswitch.org/
>> 
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>> 
>> Twitter: http://twitter.com/FreeSWITCH_wire
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>> 
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>> 
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>> 
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>> 
>> FreeSWITCH http://www.freeswitch.org/
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>> Twitter: http://twitter.com/FreeSWITCH_wire
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>> 
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>> 
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>> 
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>> 
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>> 
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>> 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
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> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
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> pstn:+19193869900
> 
> _______________________________________________
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From anthony.minessale at gmail.com  Fri Oct 29 17:43:27 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 29 Oct 2010 19:43:27 -0500
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <317A03A2-FE67-4D0D-BBA9-0661079DC61E@mgtech.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
	<407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>
	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
	<AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
	<6D9FD5E2-21FA-4630-8CF4-A87ADB30AC87@mgtech.com>
	<AANLkTi=BiDiXLgEoj_MfwmZbGjvXoOuNJgUYdetciNCj@mail.gmail.com>
	<9BAFDAF8-1E66-444C-A024-A88FF65D8BCA@mgtech.com>
	<3447E946-0BF8-4F7E-B5E9-E28F53DD0968@mgtech.com>
	<AANLkTi=LRbz6ADhLvBKkjGxOywAVLf--GTkzXEpUoDbi@mail.gmail.com>
	<BE754258-BE27-454F-B740-527A35DBDFFE@mgtech.com>
	<AANLkTi=3Kd-bCHJSED7VEh-cvtpdXMs_EkWZL=s8J4Ux@mail.gmail.com>
	<14205291-75B3-470F-ABA1-EE7F08A695E9@mgtech.com>
	<AANLkTi=D6siEAquSq6gYh-RnZeJJWrk11mHWK8De2tLh@mail.gmail.com>
	<2C35DBE8-13B3-4264-BB43-7496A1054AD1@mgtech.com>
	<86E8AE3F-E945-4952-A438-C674F23881D6@mgtech.com>
	<AANLkTin5ni_P-D5+XnCKugoRLgoExACV=eBWo3496AXq@mail.gmail.com>
	<317A03A2-FE67-4D0D-BBA9-0661079DC61E@mgtech.com>
Message-ID: <AANLkTik62y_uSx42tTsb9W6wGOFZ=bd7AV2pBcb=cAjg@mail.gmail.com>

also raise loglevel to debug
console loglevel debug


On Fri, Oct 29, 2010 at 7:29 PM, Mario G <mario_fs at mgtech.com> wrote:
> Will so, BTW, I mentioned below one is dns and another uses IP, I tested that theory using an IP ?does not help. I also updated to todays git version and no help there. Will post when the trace is done.
>
> On Oct 29, 2010, at 5:17 PM, Anthony Minessale wrote:
>
>> can you repeat that trace with sofia debug on
>> sofia loglevel all 9
>>
>> Are you doing DNS by any chance in the gateway "proxy" param?
>> you could try filling in the register-proxy param in your gateway to
>> sip:<ip> <-- not dns but ip that dns resolves to
>> <param name="register-proxy" value="sip:1.2.3.4"/>
>>
>> I'm just guessing but Its possible some bad dns query could be throwing FS off.
>> so this test would force all the packets to the exact IP of your host
>> instead of looking it up.
>>
>>
>>
>> On Fri, Oct 29, 2010 at 6:30 PM, Mario G <mario_fs at mgtech.com> wrote:
>>> Oh Boy: Not only is everything set to the static route but I turned
>>> off/disconnected the dynamic DSL line so I only had 1 static line to the
>>> router. The second wan is set in the router off. Even turned off router uPnP
>>> even though I am using -nonat. Guess what.... I still have the problem. Look
>>> like when FS says is going to retry it actually does not.
>>> Here is a short one: http://pastebin.freeswitch.org/14359 - I caught one
>>> right after I started FS, failure occurred in minutes (lucky). Look at the
>>> bottom of the trace, you see SIP trace activity and then when it fails no
>>> SIP trace activity. Could this possibly be a FS bug? (I am a mainframe
>>> assembler systems programmer and I might think so if there were error retry
>>> messages but nothing showing in one of my traces...)
>>> Notes:
>>> idone is gateway 1
>>> idtwo is gateway 2 ?I had to trace both because it was impossible to figure
>>> out which one would fail first. Ran several times but kept missing the right
>>> one.
>>> I use a url for one gateway and ip for another but it makes no difference
>>> since both eventually fail.
>>> 10. is local lan
>>> 210. is external ip
>>> 216. is itsp
>>>
>>> Here is a longer one from earlier ? http://pastebin.freeswitch.org/14357
>>> Notes:
>>> A call was received and hung up for idtwo - beginning of trace
>>> 11 minutes later idtwo failed - see last line of trace
>>>
>>> Thank you very much!
>>> Mario
>>>
>>> On Oct 29, 2010, at 12:11 PM, Anthony Minessale wrote:
>>>
>>> stun-enabled must be true in your profile XML to see what you pasted.
>>>
>>> Get me a sip trace of this from when it works until when it fails
>>>
>>> only enable the sip trace on the profile with the gateway to reduce traffic
>>>
>>>
>>>
>>> On Fri, Oct 29, 2010 at 1:56 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>
>>> Oh my... ?looks it was not NAT after all? Please help! I changed to all
>>> profiles to static per instructions below and still have the problem:
>>>
>>> 2010-10-29 11:15:18.536446 [NOTICE] sofia_reg.c:342 Registering uuid1
>>>
>>> 2010-10-29 11:15:34.313150 [WARNING] sofia_reg.c:387 uuid1 Failed
>>> Registration, setting retry to 15 seconds.
>>>
>>> sofia global siptrace on did not show any activity for this gateway in or
>>> out, others were fine but eventually fail. I setup static:
>>>
>>> 1. set the params ext-sip-ip and ext-rtp-ip to my external static IP
>>>
>>> 2. map the sip ports (5060-5080) and all of the rtp ports (16384-32767) ?to
>>> FS lan addr.
>>>
>>> 3. set sip-ip and rtp-ip to the lan addr of FS
>>>
>>> 4. start FS with -nonat
>>>
>>> I don't know what to try next. BTW, the sofia status for the profiles shows
>>> stun enabled but I did not set it up anywhere:
>>>
>>> Name ? ? ? ? ? ? ? ? ? ?uuid1
>>>
>>> Domain Name ? ? ? ? ? ? N/A
>>>
>>> Auto-NAT ? ? ? ? ? ? ? ?false
>>>
>>> DBName ? ? ? ? ? ? ? ? ?sofia_reg_mvvyl
>>>
>>> Pres Hosts
>>>
>>> Dialplan ? ? ? ? ? ? ? ?XML
>>>
>>> Context ? ? ? ? ? ? ? ? public
>>>
>>> Challenge Realm ? ? ? ? auto_to
>>>
>>> RTP-IP ? ? ? ? ? ? ? ? ?10.x.x.20
>>>
>>> Ext-RTP-IP ? ? ? ? ? ? ?210.x.x.100
>>>
>>> SIP-IP ? ? ? ? ? ? ? ? ?10.x.x.20
>>>
>>> Ext-SIP-IP ? ? ? ? ? ? ?210.x.x.100
>>>
>>> URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 210.x.x.100:5068
>>>
>>> BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 210.x.x.100:5068;maddr=10.x.x.20
>>>
>>> HOLD-MUSIC ? ? ? ? ? ? ?local_stream://moh
>>>
>>> OUTBOUND-PROXY ? ? ? ? ?N/A
>>>
>>> CODECS IN ? ? ? ? ? ? ? G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
>>>
>>> CODECS OUT ? ? ? ? ? ? ?PCMU,PCMA,GSM
>>>
>>> TEL-EVENT ? ? ? ? ? ? ? 101
>>>
>>> DTMF-MODE ? ? ? ? ? ? ? rfc2833
>>>
>>> CNG ? ? ? ? ? ? ? ? ? ? 13
>>>
>>> SESSION-TO ? ? ? ? ? ? ?0
>>>
>>> MAX-DIALOG ? ? ? ? ? ? ?0
>>>
>>> NOMEDIA ? ? ? ? ? ? ? ? false
>>>
>>> LATE-NEG ? ? ? ? ? ? ? ?false
>>>
>>> PROXY-MEDIA ? ? ? ? ? ? false
>>>
>>> AGGRESSIVENAT ? ? ? ? ? false
>>>
>>> STUN-ENABLED ? ? ? ? ? ?true
>>>
>>> STUN-AUTO-DISABLE ? ? ? false
>>>
>>> CALLS-IN ? ? ? ? ? ? ? ?0
>>>
>>> FAILED-CALLS-IN ? ? ? ? 0
>>>
>>> CALLS-OUT ? ? ? ? ? ? ? 0
>>>
>>> FAILED-CALLS-OUT ? ? ? ?0
>>>
>>>
>>> On Oct 27, 2010, at 10:04 AM, Anthony Minessale wrote:
>>>
>>> if you map it or not, a scanner would penetrate it.
>>>
>>> There are lot of sip scanners out there now, you just need to beware of
>>> them.
>>>
>>>
>>> On Wed, Oct 27, 2010 at 11:50 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>
>>> Thanks so much! I am sure many others will find this info invaluable. I will
>>> try the static route again but have one question: When I started with FS I
>>> found a "sip scanner" ?in FS and someone on this group said not to use port
>>> mapping since it was a security risk. Is that true?
>>>
>>> On Oct 27, 2010, at 9:10 AM, Anthony Minessale wrote:
>>>
>>> you are completely guessing at things.
>>>
>>> I want you to understand that the only reason you are having problems
>>>
>>> with this is because you don't understand how it works enough to know
>>>
>>> what you are doing 100%
>>>
>>> Its a given that the pnp stuff is only for your dynamic IP.
>>>
>>> aggressive-nat-detection and sip-force-expires are all related to
>>>
>>> inbound calls when the things who are registering to you may be behind
>>>
>>> nat.
>>>
>>> You need to learn the difference between which nat tools are
>>>
>>> *) designed for your FS to run behind nat
>>>
>>> *) designed for FS to run public and accept connections from devices behind
>>> nat.
>>>
>>> If you have a static IP, you don't need the pnp stuff so -nonat is fine
>>>
>>> What you need to do is set
>>>
>>> 1) set the params ext-sip-ip and ext-rtp-ip to your external static IP
>>>
>>> 2) map the sip ports and all of the rtp ports from your static IP to FS lan
>>> addr
>>>
>>> 3) set sip-ip and rtp-ip to the lan addr you forwarded through.
>>>
>>>
>>> If you don't do this: your outbound registration will use NAT to your
>>>
>>> provider and if there is no activity for the expire time on your NAT
>>>
>>> mapping the reverse port mapping from your provider back to you is
>>>
>>> lost. ?This is why you set your register expires to a very low number,
>>>
>>> (you need to make sure the provider does not turn the expires back up
>>>
>>> in the reply because it will beat your choice *see sip trace) if this
>>>
>>> is the case then you need the "ping" option set to 30, to continuously
>>>
>>> send an options to your provider.
>>>
>>> The static mapping is obviously the better, easier and more reliable
>>> solution.
>>>
>>> So I want you to understand that the only way to keep a nat mapped
>>>
>>> port alive is to continuously send traffic, all the other methods that
>>>
>>> you are mentioning are to detect that phones registered to your are
>>>
>>> behind nat, I gave you that force-expires option before because your
>>>
>>> trace was full of inbound reg so I thought that is what you wanted
>>>
>>> help with.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Oct 27, 2010 at 10:43 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>
>>> I should mention that I did not have this problem with an SPA9000 PBX
>>>
>>> (asterisk based) for over two years so FS may be pickier about upnp and/or
>>>
>>> nat, or just better at it exposing a problem in the router.
>>>
>>> I made different changes to the gateways to test different things. One
>>>
>>> failed after 17 hours, the other two stayed up. ?What did not work:
>>>
>>> added <variable name="sip-force-expires" value="30"/> to the directory
>>>
>>> entries as suggested.
>>>
>>> set the gateway expire times to 30 seconds.
>>>
>>> What worked (could be coincidental) for the two gateways that stayed up:
>>>
>>> I Added <param name="aggressive-nat-detection" value="true"/>
>>>
>>> I originally setup FS to use the static ip by setting external sip/rtp to
>>>
>>> just the static ip (no autonat:) and ran with -nonat but I could not get
>>>
>>> incoming calls. The only way it worked was to use autonat:1.2.3.4. The
>>>
>>> router has 1 static public address and 1 dynamic external IP, this is the
>>>
>>> root of the problem, upnp only tells FS about the dynamic ip ?Will keep this
>>>
>>> thread up-to-date for anyone who may be in the same boat someday. Thanks
>>>
>>> again for looking at the trace.
>>>
>>> Mario
>>>
>>> You should be setting the req freq to a low number on the outbound gateways
>>>
>>> The examples you showed had a series of inbound reg
>>>
>>> also set expire-seconds to 30 in your gateway xml
>>>
>>>
>>> The problem is if you are not constantly sending traffic to the box
>>>
>>> the nat mapping will go away.
>>>
>>> If you are in production you should be using a static ip with a static
>>>
>>> mapping, any trouble you are having is your own fault for playing with
>>>
>>> fire. ?The best we can do is tell you how to keep it contained.
>>>
>>>
>>>
>>>
>>> On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>
>>> I made the change. I had no idea the settings for the inside phones effected
>>>
>>> nat for the outside sip accounts. I was looking into aggressive-nat-
>>>
>>> detection since the internal profile status always shows the right external
>>>
>>> static IP but the nat_ap status always shows the dynamic ip. Crossing
>>>
>>> fingers/etc since this problem is 85% of time (weeks!) into FS changeover.
>>>
>>> Thanks!
>>>
>>> Mario
>>>
>>> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
>>>
>>> add
>>>
>>> <variable name="sip-force-expires" value="30"/>
>>>
>>> to the <variables> section of your <user>
>>>
>>> you have it at 600 and the nat mapping is timing out while the 600
>>>
>>> seconds is ticking away
>>>
>>>
>>>
>>> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>
>>> From the TSP:
>>>
>>> "I have enabled the SIP trace on your account. We are not currently seeing
>>>
>>> any registration attempts to your account within the last 15 minutes. Please
>>>
>>> restart FreeSwitch so that registration attempts begin again. Thank you. ".
>>>
>>> So FS is not getting past router.
>>>
>>> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>>>
>>> I ran the global trace during the problem and it is
>>>
>>> at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
>>>
>>> "accttwo", "acct3". The trace includes phones since it was global. I am
>>>
>>> using:
>>>
>>> ? <param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>>>
>>> ? <param name="ext-sip-ip" value="autonat:my-static.ip"/>
>>>
>>> I tried dumping nat and removing the autonat: above and using -nonat but
>>>
>>> that did not work, registration proceeded but no calls inbound.
>>>
>>> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>>>
>>> Whoops, I am using an IP address for at least one gateway so that is not the
>>>
>>> problem:
>>>
>>> They look outbound to me and I am using dns for 2 and an IP for one so that
>>>
>>> is not the issue. I was able to get FS to clear this up by doing "nat_map
>>>
>>> reinit" which is why I think this is a nat problem. I will do the trace you
>>>
>>> mentioned. I will plug an ip address into one of the gateways to see what
>>>
>>> happens, they all fail at once. Thanks for responding!
>>>
>>> Mario
>>>
>>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>>>
>>> I really need help on this as I have weeks into this problem. I thought I
>>>
>>> had it nailed but I guess not. After 5.5 hours I get:
>>>
>>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>>>
>>> Registration, setting retry to 15 seconds.
>>>
>>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>
>>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>
>>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
>>>
>>> Registration, setting retry to 30 seconds.
>>>
>>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>>>
>>> Registration, setting retry to 15 seconds.
>>>
>>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>>>
>>> Registration, setting retry to 15 seconds.
>>>
>>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>
>>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>
>>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>>>
>>> and no way to make/get calls until I restart FS. I did this:
>>>
>>> 1. log 7
>>>
>>> 2. sofia profile xxxx siptrace on ? for each profile/gateway
>>>
>>> 3. restarted router
>>>
>>> All three did not solve the problem. The trace and log produced no
>>>
>>> additional lines which is why I am wondering if FS has a problem since the
>>>
>>> trace shows no SIP activity.
>>>
>>> 3 gateways with 2 ITSPs
>>>
>>> 2 DSL/WAN lines, 1 static and 1 dynamic
>>>
>>> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
>>>
>>> external static ip.
>>>
>>> sofia status profile ... has the right ext ip
>>>
>>> nat_map status shows the dynamic (wrong) IP
>>>
>>> I tried starting with -nonat but that was worse
>>>
>>> the only way to fix is restart FS.
>>>
>>> I read the wiki on external nat, auto_nat and everything else many times.
>>>
>>> Thanks Mario
>>>
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> 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
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> 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
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> 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
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
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>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> 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
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> 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
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From dujinfang at gmail.com  Fri Oct 29 18:12:59 2010
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 30 Oct 2010 09:12:59 +0800
Subject: [Freeswitch-users] wiki question - want to add Mac osX info
In-Reply-To: <AEFCAD97-9D78-498C-9B90-8712C39F6268@mgtech.com>
References: <AEFCAD97-9D78-498C-9B90-8712C39F6268@mgtech.com>
Message-ID: <AANLkTimRtjiTc2Sp9wBZ-tue8KXtxO=KnuibzxgXSrrv@mail.gmail.com>

There's some entries about OSX,  you could just add below that or
start  a new page(if it is long) and drop a link there.


http://wiki.freeswitch.org/wiki/Installation_Guide#Unix-like_systems_.28Linux.2C_BSD.2C_OS_X.29

http://wiki.freeswitch.org/wiki/Installation_Guide#Mac_OS_X



On Sat, Oct 30, 2010 at 6:05 AM, Mario G <mario_fs at mgtech.com> wrote:
> I am about to start work my wiki contribution which specifically target my installation on Mac osX and other related information to Mac osX. I saw there is no category for "installation". So then does this info go under "Examples"?
> Mario
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From msc at freeswitch.org  Fri Oct 29 18:36:28 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Oct 2010 18:36:28 -0700
Subject: [Freeswitch-users] Problem with T.38 fax receive using ESL
In-Reply-To: <1288384068806-5687937.post@n2.nabble.com>
References: <1288272464502-5682561.post@n2.nabble.com>
	<AANLkTinzujxqm7gT8k20+yeSKR+nYSRzjmdUSg_BGLZT@mail.gmail.com>
	<1288358483496-5686430.post@n2.nabble.com>
	<1288384068806-5687937.post@n2.nabble.com>
Message-ID: <AANLkTimxg33wLG2xi_1q+cg-HXyzrUhy1csthefMr_J0@mail.gmail.com>

Interesting behavior. In any case, would you be willing to add this
knowledge to the wiki? You can put it on the the mod_spandsp page. Create a
section just like this one:
http://wiki.freeswitch.org/wiki/Mod_spandsp#Invoking_the_app_from_the_XML_dialplan

...and call it "Invoking_the_app_from_ESL"

Let me know if you have any questions and I will be happy to assist.

-MC


On Fri, Oct 29, 2010 at 1:27 PM, peely <freeswitch at peely.com> wrote:

>
> Hi,
>
> OK, I set those specifically using SendMsg with the set application, but
> still nothing happened.
>
> After some bashing around though I've figured that the behaviour is
> slightly
> different in esl compared to the dialplan.
>
> It seems that you MUST not issue an answer followed by the broadcast of
> silence, instead just call rxfax and let it answer the incoming call, if
> you
> don't do this then a T.38 session is never reinvited.
>
> It's working perfectly now, thank-you!
>
>
>
> Regards,
>
>
>
> Neil.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Problem-with-T-38-fax-receive-using-ESL-tp5682561p5687937.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Prometheus001 at gmx.net  Fri Oct 29 19:21:10 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Sat, 30 Oct 2010 04:21:10 +0200
Subject: [Freeswitch-users] no audio when originate to 2 PSTNs
In-Reply-To: <BLU145-W2762A83E99DB06B5F7C262AB450@phx.gbl>
References: <BLU145-W2762A83E99DB06B5F7C262AB450@phx.gbl>
Message-ID: <4CCB8116.9040703@gmx.net>

Looking at the SDP I can see that a number of different IPs are
involved. Seems that Audio goes to the wrong IP. But it's hard to
determine if no details are there about the callflow between the
involved components (IPs).
Is NAT involved?
You may use STUN?

Best regards
Peter

Tidiane Sy schrieb:
> Hi all,
> When I do an originate between two PSTNS, the two telephones ring. But
> when both answer, there is no audio.
> I have audio when the originate is between an sip client and a PSTN.
> My freeswitch is not natted
> my console logs here: http://pastebin.freeswitch.org/14351
> my network capture here:
> https://rcpt.yousendit.com/978754413/751404552085a1a09ea574cb819a5f74
>
> Your help will be really appreciated
>
> Tid
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From bwibowo at gmail.com  Fri Oct 29 22:53:56 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Sat, 30 Oct 2010 12:53:56 +0700
Subject: [Freeswitch-users] Problems making a receiveing calls with
	mod_jingling
In-Reply-To: <AANLkTin_0+04eA98dXTNZztLZfuCa8SFubaJr921j8JH@mail.gmail.com>
References: <201010131244.41086.justlikeef@gmail.com>
	<AANLkTim60-09hvL1=SBrb8R86_1qqPXE2-+42ZSxZ9Ja@mail.gmail.com>
	<201010131807.46035.justlikeef@gmail.com>
	<AANLkTi=K65YxFQAXudWiPTSAUSH2Xzjd4ks5_coSJ5K_@mail.gmail.com>
	<AANLkTi=Exfm08ZMfm7Eur8GQBViX6KuPt_ntRYJ48qW_@mail.gmail.com>
	<AANLkTi=ZmQo6QHfEFoHPaNd_Tx2HrM+aZZAgfh569scz@mail.gmail.com>
	<AANLkTin_0+04eA98dXTNZztLZfuCa8SFubaJr921j8JH@mail.gmail.com>
Message-ID: <AANLkTin7PG-u2yUdTd-hvLqKk7yZ7PJ+R_ntw-2yZTgq@mail.gmail.com>

thx and works now for somebody that need detail steps please visit
http://budi.blogsome.com

thx

<http://wiki.freeswitch.org/wiki/Google_Voice>

On Fri, Oct 29, 2010 at 11:19 AM, budi wibowo <bwibowo at gmail.com> wrote:

> thx fixed after upgrade.
> client.xml already configured as wiki said.
> for dialplan i put
> in usr/local/freeswitch/conf/dialplan/default/02_gtalk.xml containing
> <include>
>     <extension name="gtalk-out">
>       <condition field="destination_number" expression="^1(1\d{10})$">
>         <action application="set" data="hangup_after_bridge=true"/>
>         <action application="bridge" data="dingaling/gtalk/+$
> 1 at voice.google.com"/>
>        </condition>
>    </extension>
> </include>
>
> and got this error
> 2010-10-29 12:17:27.332766 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/1000 at 202.122.99.99 Standard REPORTING, cause:
> RECOVERY_ON_TIMER_EXPIRE
> 2010-10-29 12:17:27.332766 [DEBUG] switch_core_state_machine.c:599
> (sofia/internal/1000 at 202.122.99.99) State REPORTING going to sleep
> 2010-10-29 12:17:27.332766 [DEBUG] switch_core_state_machine.c:331
> (sofia/internal/1000 at 202.122.99.99) State Change CS_REPORTING ->
> CS_DESTROY
> 2010-10-29 12:17:27.332766 [DEBUG] switch_core_session.c:1057 Send signal
> sofia/internal/1000 at 202.122.99.99 [BREAK]
> 2010-10-29 12:17:27.332766 [DEBUG] switch_core_session.c:1224 Session 5
> (sofia/internal/1000 at 202.122.99.99) Locked, Waiting on external entities
>
> what is the trigger for recovery on timer expiry?
> what codec should i choose for gtalk, i use acrobits on iphone with GSM
> codec.
>
>
> regards
>
> budi
>
>
>
>
> On Thu, Oct 28, 2010 at 6:32 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> First, update to the latest git head. Second, turn on dingaling debugging
>> and capture the output:
>> dl_debug on
>>
>> Hopefully there will be clues.
>> -MC
>>
>>
>> On Wed, Oct 27, 2010 at 3:44 PM, budi wibowo <bwibowo at gmail.com> wrote:
>>
>>> hi
>>> i use FreeSWITCH version: 1.0.head (git-cf5c1d2 2010-10-20 16-40-26
>>> -0400)
>>> i have followed the suggestion but still found this error
>>>
>>>
>>>
>>> 2010-10-28 06:41:45.720905 [ERR] libdingaling.c:1205 NODE ERROR!
>>>
>>> 2010-10-28 06:41:45.934903 [ERR] libdingaling.c:1205 NODE ERROR!
>>>
>>> 2010-10-28 06:41:46.158907 [ERR] libdingaling.c:1205 NODE ERROR!
>>>
>>> 2010-10-28 06:41:46.379903 [ERR] libdingaling.c:1205 NODE ERROR!
>>>
>>> 2010-10-28 06:41:46.589904 [ERR] libdingaling.c:1209 DISCONNECTED!
>>>
>>> any help is highly appreciate
>>>
>>>
>>> On Thu, Oct 14, 2010 at 12:28 PM, Jeffrey Leung <
>>> curriegrad2004 at gmail.com> wrote:
>>>
>>>> This is my config inside the jingle_profiles:
>>>>  <profile type="client">
>>>> *    <param name="name" value="gtalk[Example]"/>*
>>>>    <param name="login" value="[EXAMPLE]@gmail.com/talk"/>
>>>>    <param name="password" value="[REDACTED]"/>
>>>>
>>>>    <param name="dialplan" value="XML"/>
>>>>    <param name="context" value="public"/>
>>>>    <param name="message" value="Online"/>
>>>>
>>>>    <param name="rtp-ip" value="$${bind_server_ip}"/>
>>>>    <!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
>>>>
>>>>    <param name="auto-login" value="true"/>
>>>>     <!-- SASL "plain" or "md5" -->
>>>>
>>>>    <param name="sasl" value="plain"/>
>>>>    <!-- if the server where the jabber is hosted is not the same as the
>>>> one in the jid -->
>>>>
>>>>    <param name="server" value="talk.google.com"/>
>>>>    <!-- Enable TLS or not -->
>>>>
>>>>    <param name="tls" value="true"/>
>>>>     <!-- disable to trade async for more calls -->
>>>>
>>>>    <param name="use-rtp-timer" value="true"/>
>>>>    <!-- default extension (if one cannot be determined) -->
>>>>    <param name="exten" value="[EXTENSION]"/>
>>>>    <!-- VAD choose one -->
>>>>    <!-- <param name="vad" value="in"/> -->
>>>>    <!-- <param name="vad" value="out"/> -->
>>>>
>>>>    <param name="vad" value="both"/>
>>>>     <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
>>>>
>>>>    <param name="candidate-acl" value="wan.auto"/>
>>>>    <param name="local-network-acl" value="localnet.auto"/>
>>>>   </profile>
>>>>
>>>> And dialplan to follow:
>>>>
>>>>  <extension name="gvoice_out">
>>>>    <condition field="destination_number" expression="^4664(\d{10})$">
>>>> *      <action application="bridge" data="dingaling/gtalk[EXAMPLE]/+1$
>>>> 1 at voice.google.com"/>*
>>>>    </condition>
>>>>  </extension>
>>>>
>>>> Pay special attention to the bolded areas of the configuration example
>>>> I've provided. Configuring Google talk to work with FreeSwitch is quite
>>>> straightforward with my configuration example I've provided above. I've ran
>>>> to this same configuration ambiguity when they first completed this feature
>>>> anyways, so I hope my configuration example does help you into resolving
>>>> that problem
>>>>
>>>>
>>>> On Wed, Oct 13, 2010 at 3:07 PM, Rob Hutton <justlikeef at gmail.com>
>>>> wrote:
>>>> > Caught and changed that, but it does the same thing.  I also changed
>>>> the profile name to gtalk in case there was a case sensitivity issue or
>>>> something, but same result.
>>>> >
>>>> > I am told that the wiki is not quite correct either, but with everyone
>>>> away from their development PCs, it will be the weekend before the
>>>> corrections are available.
>>>> >
>>>> > --
>>>> > Thanks,
>>>> > Rob
>>>> > On Wednesday 13 October 2010 17:51:07 Michael Collins wrote:
>>>> >> > Here is the profile:
>>>> >> >
>>>> >> >  <profile name="dingaling_1">
>>>> >> >
>>>> >> I believe that this should be:
>>>> >> <profile type="client">
>>>> >>
>>>> >> -MC
>>>> >>
>>>> > I
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From tisyfreeswitch at hotmail.fr  Sat Oct 30 05:38:07 2010
From: tisyfreeswitch at hotmail.fr (Tidiane Sy)
Date: Sat, 30 Oct 2010 14:38:07 +0200
Subject: [Freeswitch-users] no audio when originate to 2 PSTNs
In-Reply-To: <4CCB8116.9040703@gmx.net>
References: <BLU145-W2762A83E99DB06B5F7C262AB450@phx.gbl>,
	<4CCB8116.9040703@gmx.net>
Message-ID: <BLU145-W319B2C79B2BF50CE63209AAB460@phx.gbl>


You are right, different IPs are involved on PSTN side. So you think freeswitch is ending audio to the wrong IP ? But, if I look at network packets leaving my freeswitch server, I can't see any RTP. You can see my network capture here https://rcpt.yousendit.com/978754413/751404552085a1a09ea574cb819a5f74 
NAT is not involved on my freeswitch server. I'm not using STUN. 

Regards,
Tidiane
> Date: Sat, 30 Oct 2010 04:21:10 +0200
> From: Prometheus001 at gmx.net
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] no audio when originate to 2 PSTNs
> 
> Looking at the SDP I can see that a number of different IPs are
> involved. Seems that Audio goes to the wrong IP. But it's hard to
> determine if no details are there about the callflow between the
> involved components (IPs).
> Is NAT involved?
> You may use STUN?
> 
> Best regards
> Peter
> 
> Tidiane Sy schrieb:
> > Hi all,
> > When I do an originate between two PSTNS, the two telephones ring. But
> > when both answer, there is no audio.
> > I have audio when the originate is between an sip client and a PSTN.
> > My freeswitch is not natted
> > my console logs here: http://pastebin.freeswitch.org/14351
> > my network capture here:
> > https://rcpt.yousendit.com/978754413/751404552085a1a09ea574cb819a5f74
> >
> > Your help will be really appreciated
> >
> > Tid
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >   
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
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From christian.knoblauch at astylos.de  Sat Oct 30 06:06:15 2010
From: christian.knoblauch at astylos.de (Christian Knoblauch)
Date: Sat, 30 Oct 2010 16:06:15 +0300
Subject: [Freeswitch-users] mod_cepstral for Windows build -> Change language
Message-ID: <000001cb7833$3f5e4900$be1adb00$@astylos.de>

Hi,

 

Is there a method to "dynamically" change the language for cepstral TTS at
runtime ?

 

Thanks & Rewgards

Christian

 

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From mario_fs at mgtech.com  Sat Oct 30 08:24:09 2010
From: mario_fs at mgtech.com (Mario G)
Date: Sat, 30 Oct 2010 08:24:09 -0700
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <AANLkTik62y_uSx42tTsb9W6wGOFZ=bd7AV2pBcb=cAjg@mail.gmail.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
	<407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>
	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
	<AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
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You should be aware of this: Setting sofia loglevel to 9 made FS memory go from 20meg to eventually using all memory and locking the machine up. I spent all night trying to get a good trace. I need to get one when the failure occurs early before the lockup. Will be working on this all weekend. Thanks for you help.
Mario

On Oct 29, 2010, at 5:43 PM, Anthony Minessale wrote:

> also raise loglevel to debug
> console loglevel debug
> 
> 
> On Fri, Oct 29, 2010 at 7:29 PM, Mario G <mario_fs at mgtech.com> wrote:
>> Will so, BTW, I mentioned below one is dns and another uses IP, I tested that theory using an IP  does not help. I also updated to todays git version and no help there. Will post when the trace is done.
>> 
>> On Oct 29, 2010, at 5:17 PM, Anthony Minessale wrote:
>> 
>>> can you repeat that trace with sofia debug on
>>> sofia loglevel all 9
>>> 
>>> Are you doing DNS by any chance in the gateway "proxy" param?
>>> you could try filling in the register-proxy param in your gateway to
>>> sip:<ip> <-- not dns but ip that dns resolves to
>>> <param name="register-proxy" value="sip:1.2.3.4"/>
>>> 
>>> I'm just guessing but Its possible some bad dns query could be throwing FS off.
>>> so this test would force all the packets to the exact IP of your host
>>> instead of looking it up.
>>> 
>>> 
>>> 
>>> On Fri, Oct 29, 2010 at 6:30 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>> Oh Boy: Not only is everything set to the static route but I turned
>>>> off/disconnected the dynamic DSL line so I only had 1 static line to the
>>>> router. The second wan is set in the router off. Even turned off router uPnP
>>>> even though I am using -nonat. Guess what.... I still have the problem. Look
>>>> like when FS says is going to retry it actually does not.
>>>> Here is a short one: http://pastebin.freeswitch.org/14359 - I caught one
>>>> right after I started FS, failure occurred in minutes (lucky). Look at the
>>>> bottom of the trace, you see SIP trace activity and then when it fails no
>>>> SIP trace activity. Could this possibly be a FS bug? (I am a mainframe
>>>> assembler systems programmer and I might think so if there were error retry
>>>> messages but nothing showing in one of my traces...)
>>>> Notes:
>>>> idone is gateway 1
>>>> idtwo is gateway 2  I had to trace both because it was impossible to figure
>>>> out which one would fail first. Ran several times but kept missing the right
>>>> one.
>>>> I use a url for one gateway and ip for another but it makes no difference
>>>> since both eventually fail.
>>>> 10. is local lan
>>>> 210. is external ip
>>>> 216. is itsp
>>>> 
>>>> Here is a longer one from earlier   http://pastebin.freeswitch.org/14357
>>>> Notes:
>>>> A call was received and hung up for idtwo - beginning of trace
>>>> 11 minutes later idtwo failed - see last line of trace
>>>> 
>>>> Thank you very much!
>>>> Mario
>>>> 
>>>> On Oct 29, 2010, at 12:11 PM, Anthony Minessale wrote:
>>>> 
>>>> stun-enabled must be true in your profile XML to see what you pasted.
>>>> 
>>>> Get me a sip trace of this from when it works until when it fails
>>>> 
>>>> only enable the sip trace on the profile with the gateway to reduce traffic
>>>> 
>>>> 
>>>> 
>>>> On Fri, Oct 29, 2010 at 1:56 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>> 
>>>> Oh my...  looks it was not NAT after all? Please help! I changed to all
>>>> profiles to static per instructions below and still have the problem:
>>>> 
>>>> 2010-10-29 11:15:18.536446 [NOTICE] sofia_reg.c:342 Registering uuid1
>>>> 
>>>> 2010-10-29 11:15:34.313150 [WARNING] sofia_reg.c:387 uuid1 Failed
>>>> Registration, setting retry to 15 seconds.
>>>> 
>>>> sofia global siptrace on did not show any activity for this gateway in or
>>>> out, others were fine but eventually fail. I setup static:
>>>> 
>>>> 1. set the params ext-sip-ip and ext-rtp-ip to my external static IP
>>>> 
>>>> 2. map the sip ports (5060-5080) and all of the rtp ports (16384-32767)  to
>>>> FS lan addr.
>>>> 
>>>> 3. set sip-ip and rtp-ip to the lan addr of FS
>>>> 
>>>> 4. start FS with -nonat
>>>> 
>>>> I don't know what to try next. BTW, the sofia status for the profiles shows
>>>> stun enabled but I did not set it up anywhere:
>>>> 
>>>> Name                    uuid1
>>>> 
>>>> Domain Name             N/A
>>>> 
>>>> Auto-NAT                false
>>>> 
>>>> DBName                  sofia_reg_mvvyl
>>>> 
>>>> Pres Hosts
>>>> 
>>>> Dialplan                XML
>>>> 
>>>> Context                 public
>>>> 
>>>> Challenge Realm         auto_to
>>>> 
>>>> RTP-IP                  10.x.x.20
>>>> 
>>>> Ext-RTP-IP              210.x.x.100
>>>> 
>>>> SIP-IP                  10.x.x.20
>>>> 
>>>> Ext-SIP-IP              210.x.x.100
>>>> 
>>>> URL                     sip:mod_sofia at 210.x.x.100:5068
>>>> 
>>>> BIND-URL                sip:mod_sofia at 210.x.x.100:5068;maddr=10.x.x.20
>>>> 
>>>> HOLD-MUSIC              local_stream://moh
>>>> 
>>>> OUTBOUND-PROXY          N/A
>>>> 
>>>> CODECS IN               G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
>>>> 
>>>> CODECS OUT              PCMU,PCMA,GSM
>>>> 
>>>> TEL-EVENT               101
>>>> 
>>>> DTMF-MODE               rfc2833
>>>> 
>>>> CNG                     13
>>>> 
>>>> SESSION-TO              0
>>>> 
>>>> MAX-DIALOG              0
>>>> 
>>>> NOMEDIA                 false
>>>> 
>>>> LATE-NEG                false
>>>> 
>>>> PROXY-MEDIA             false
>>>> 
>>>> AGGRESSIVENAT           false
>>>> 
>>>> STUN-ENABLED            true
>>>> 
>>>> STUN-AUTO-DISABLE       false
>>>> 
>>>> CALLS-IN                0
>>>> 
>>>> FAILED-CALLS-IN         0
>>>> 
>>>> CALLS-OUT               0
>>>> 
>>>> FAILED-CALLS-OUT        0
>>>> 
>>>> 
>>>> On Oct 27, 2010, at 10:04 AM, Anthony Minessale wrote:
>>>> 
>>>> if you map it or not, a scanner would penetrate it.
>>>> 
>>>> There are lot of sip scanners out there now, you just need to beware of
>>>> them.
>>>> 
>>>> 
>>>> On Wed, Oct 27, 2010 at 11:50 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>> 
>>>> Thanks so much! I am sure many others will find this info invaluable. I will
>>>> try the static route again but have one question: When I started with FS I
>>>> found a "sip scanner"  in FS and someone on this group said not to use port
>>>> mapping since it was a security risk. Is that true?
>>>> 
>>>> On Oct 27, 2010, at 9:10 AM, Anthony Minessale wrote:
>>>> 
>>>> you are completely guessing at things.
>>>> 
>>>> I want you to understand that the only reason you are having problems
>>>> 
>>>> with this is because you don't understand how it works enough to know
>>>> 
>>>> what you are doing 100%
>>>> 
>>>> Its a given that the pnp stuff is only for your dynamic IP.
>>>> 
>>>> aggressive-nat-detection and sip-force-expires are all related to
>>>> 
>>>> inbound calls when the things who are registering to you may be behind
>>>> 
>>>> nat.
>>>> 
>>>> You need to learn the difference between which nat tools are
>>>> 
>>>> *) designed for your FS to run behind nat
>>>> 
>>>> *) designed for FS to run public and accept connections from devices behind
>>>> nat.
>>>> 
>>>> If you have a static IP, you don't need the pnp stuff so -nonat is fine
>>>> 
>>>> What you need to do is set
>>>> 
>>>> 1) set the params ext-sip-ip and ext-rtp-ip to your external static IP
>>>> 
>>>> 2) map the sip ports and all of the rtp ports from your static IP to FS lan
>>>> addr
>>>> 
>>>> 3) set sip-ip and rtp-ip to the lan addr you forwarded through.
>>>> 
>>>> 
>>>> If you don't do this: your outbound registration will use NAT to your
>>>> 
>>>> provider and if there is no activity for the expire time on your NAT
>>>> 
>>>> mapping the reverse port mapping from your provider back to you is
>>>> 
>>>> lost.  This is why you set your register expires to a very low number,
>>>> 
>>>> (you need to make sure the provider does not turn the expires back up
>>>> 
>>>> in the reply because it will beat your choice *see sip trace) if this
>>>> 
>>>> is the case then you need the "ping" option set to 30, to continuously
>>>> 
>>>> send an options to your provider.
>>>> 
>>>> The static mapping is obviously the better, easier and more reliable
>>>> solution.
>>>> 
>>>> So I want you to understand that the only way to keep a nat mapped
>>>> 
>>>> port alive is to continuously send traffic, all the other methods that
>>>> 
>>>> you are mentioning are to detect that phones registered to your are
>>>> 
>>>> behind nat, I gave you that force-expires option before because your
>>>> 
>>>> trace was full of inbound reg so I thought that is what you wanted
>>>> 
>>>> help with.
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> On Wed, Oct 27, 2010 at 10:43 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>> 
>>>> I should mention that I did not have this problem with an SPA9000 PBX
>>>> 
>>>> (asterisk based) for over two years so FS may be pickier about upnp and/or
>>>> 
>>>> nat, or just better at it exposing a problem in the router.
>>>> 
>>>> I made different changes to the gateways to test different things. One
>>>> 
>>>> failed after 17 hours, the other two stayed up.  What did not work:
>>>> 
>>>> added <variable name="sip-force-expires" value="30"/> to the directory
>>>> 
>>>> entries as suggested.
>>>> 
>>>> set the gateway expire times to 30 seconds.
>>>> 
>>>> What worked (could be coincidental) for the two gateways that stayed up:
>>>> 
>>>> I Added <param name="aggressive-nat-detection" value="true"/>
>>>> 
>>>> I originally setup FS to use the static ip by setting external sip/rtp to
>>>> 
>>>> just the static ip (no autonat:) and ran with -nonat but I could not get
>>>> 
>>>> incoming calls. The only way it worked was to use autonat:1.2.3.4. The
>>>> 
>>>> router has 1 static public address and 1 dynamic external IP, this is the
>>>> 
>>>> root of the problem, upnp only tells FS about the dynamic ip  Will keep this
>>>> 
>>>> thread up-to-date for anyone who may be in the same boat someday. Thanks
>>>> 
>>>> again for looking at the trace.
>>>> 
>>>> Mario
>>>> 
>>>> You should be setting the req freq to a low number on the outbound gateways
>>>> 
>>>> The examples you showed had a series of inbound reg
>>>> 
>>>> also set expire-seconds to 30 in your gateway xml
>>>> 
>>>> 
>>>> The problem is if you are not constantly sending traffic to the box
>>>> 
>>>> the nat mapping will go away.
>>>> 
>>>> If you are in production you should be using a static ip with a static
>>>> 
>>>> mapping, any trouble you are having is your own fault for playing with
>>>> 
>>>> fire.  The best we can do is tell you how to keep it contained.
>>>> 
>>>> 
>>>> 
>>>> 
>>>> On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>> 
>>>> I made the change. I had no idea the settings for the inside phones effected
>>>> 
>>>> nat for the outside sip accounts. I was looking into aggressive-nat-
>>>> 
>>>> detection since the internal profile status always shows the right external
>>>> 
>>>> static IP but the nat_ap status always shows the dynamic ip. Crossing
>>>> 
>>>> fingers/etc since this problem is 85% of time (weeks!) into FS changeover.
>>>> 
>>>> Thanks!
>>>> 
>>>> Mario
>>>> 
>>>> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
>>>> 
>>>> add
>>>> 
>>>> <variable name="sip-force-expires" value="30"/>
>>>> 
>>>> to the <variables> section of your <user>
>>>> 
>>>> you have it at 600 and the nat mapping is timing out while the 600
>>>> 
>>>> seconds is ticking away
>>>> 
>>>> 
>>>> 
>>>> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>> 
>>>> From the TSP:
>>>> 
>>>> "I have enabled the SIP trace on your account. We are not currently seeing
>>>> 
>>>> any registration attempts to your account within the last 15 minutes. Please
>>>> 
>>>> restart FreeSwitch so that registration attempts begin again. Thank you. ".
>>>> 
>>>> So FS is not getting past router.
>>>> 
>>>> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>>>> 
>>>> I ran the global trace during the problem and it is
>>>> 
>>>> at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
>>>> 
>>>> "accttwo", "acct3". The trace includes phones since it was global. I am
>>>> 
>>>> using:
>>>> 
>>>>   <param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>>>> 
>>>>   <param name="ext-sip-ip" value="autonat:my-static.ip"/>
>>>> 
>>>> I tried dumping nat and removing the autonat: above and using -nonat but
>>>> 
>>>> that did not work, registration proceeded but no calls inbound.
>>>> 
>>>> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>>>> 
>>>> Whoops, I am using an IP address for at least one gateway so that is not the
>>>> 
>>>> problem:
>>>> 
>>>> They look outbound to me and I am using dns for 2 and an IP for one so that
>>>> 
>>>> is not the issue. I was able to get FS to clear this up by doing "nat_map
>>>> 
>>>> reinit" which is why I think this is a nat problem. I will do the trace you
>>>> 
>>>> mentioned. I will plug an ip address into one of the gateways to see what
>>>> 
>>>> happens, they all fail at once. Thanks for responding!
>>>> 
>>>> Mario
>>>> 
>>>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>>>> 
>>>> I really need help on this as I have weeks into this problem. I thought I
>>>> 
>>>> had it nailed but I guess not. After 5.5 hours I get:
>>>> 
>>>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>>>> 
>>>> Registration, setting retry to 15 seconds.
>>>> 
>>>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>> 
>>>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>> 
>>>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
>>>> 
>>>> Registration, setting retry to 30 seconds.
>>>> 
>>>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>>>> 
>>>> Registration, setting retry to 15 seconds.
>>>> 
>>>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>>>> 
>>>> Registration, setting retry to 15 seconds.
>>>> 
>>>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>> 
>>>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>> 
>>>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>>>> 
>>>> and no way to make/get calls until I restart FS. I did this:
>>>> 
>>>> 1. log 7
>>>> 
>>>> 2. sofia profile xxxx siptrace on   for each profile/gateway
>>>> 
>>>> 3. restarted router
>>>> 
>>>> All three did not solve the problem. The trace and log produced no
>>>> 
>>>> additional lines which is why I am wondering if FS has a problem since the
>>>> 
>>>> trace shows no SIP activity.
>>>> 
>>>> 3 gateways with 2 ITSPs
>>>> 
>>>> 2 DSL/WAN lines, 1 static and 1 dynamic
>>>> 
>>>> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
>>>> 
>>>> external static ip.
>>>> 
>>>> sofia status profile ... has the right ext ip
>>>> 
>>>> nat_map status shows the dynamic (wrong) IP
>>>> 
>>>> I tried starting with -nonat but that was worse
>>>> 
>>>> the only way to fix is restart FS.
>>>> 
>>>> I read the wiki on external nat, auto_nat and everything else many times.
>>>> 
>>>> Thanks Mario
>>>> 
>>>> 
>>>> _______________________________________________
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> 
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> 
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>>>> 
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>>>> 
>>>> 
>>>> --
>>>> 
>>>> Anthony Minessale II
>>>> 
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> 
>>>> ClueCon http://www.cluecon.com/
>>>> 
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>> 
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>>>> 
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>>>> 
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>>>> 
>>>> 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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>>>> 
>>>> 
>>>> --
>>>> 
>>>> Anthony Minessale II
>>>> 
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> 
>>>> ClueCon http://www.cluecon.com/
>>>> 
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>> 
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>>>> 
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>>>> 
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>>>> 
>>>> 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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>>>> 
>>>> --
>>>> 
>>>> Anthony Minessale II
>>>> 
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> 
>>>> ClueCon http://www.cluecon.com/
>>>> 
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>> 
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>>>> 
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>>>> 
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>>>> 
>>>> 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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>>>> 
>>>> --
>>>> 
>>>> Anthony Minessale II
>>>> 
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> 
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>>>> 
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>> 
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>>>> 
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>>>> 
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>>>> 
>>>> IRC: irc.freenode.net #freeswitch
>>>> 
>>>> FreeSWITCH Developer Conference
>>>> 
>>>> sip:888 at conference.freeswitch.org
>>>> 
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>>>> 
>>>> pstn:+19193869900
>>>> 
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>>>> 
>>>> 
>>>> --
>>>> 
>>>> Anthony Minessale II
>>>> 
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> 
>>>> ClueCon http://www.cluecon.com/
>>>> 
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>> 
>>>> AIM: anthm
>>>> 
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>>>> 
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>>>> 
>>>> IRC: irc.freenode.net #freeswitch
>>>> 
>>>> FreeSWITCH Developer Conference
>>>> 
>>>> sip:888 at conference.freeswitch.org
>>>> 
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> 
>>>> pstn:+19193869900
>>>> 
>>>> _______________________________________________
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> 
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> 
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> 
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> 
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> 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
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From fraserredmond at gmail.com  Sat Oct 30 08:57:23 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Sat, 30 Oct 2010 16:57:23 +0100
Subject: [Freeswitch-users] Loss of first second of media
In-Reply-To: <AANLkTikC052SeKz6=1yoxa9A3HD-CnhdjwfUmWLfqV0J@mail.gmail.com>
References: <AANLkTimECwtVmnbFsBJRiN3YSaGyXbvh-sKtKYPey8xx@mail.gmail.com>
	<AANLkTik5H74MupY8iyx6Ao4=Rv=wCm8jv-w=xCtSnm0Z@mail.gmail.com>
	<AANLkTikpVH-s6AvJ7fQoZu2PE4dduTg+d-ygyBWSfr4R@mail.gmail.com>
	<AANLkTi=FZDqev9R1aTMpdvTk8t9NfxiyeisQSRBxfTEb@mail.gmail.com>
	<AANLkTikY68ovuiSO9xkbZZPSskxCNsjrpTSdb-42bwJB@mail.gmail.com>
	<AANLkTikiG3xPKDVkwU4GDk5t6KDe67AQmM-dkheMCSu7@mail.gmail.com>
	<AANLkTinJRcpRx_BQyKifECcurYb+vG2P8Nct1xT_X3XO@mail.gmail.com>
	<AANLkTinpaRivNHteDp4i6WJjmgsPAE6otVx5LjgJcvWr@mail.gmail.com>
	<AANLkTikiVxFAOwftrmfqd3LtD4H__UUytsM117WBhZy0@mail.gmail.com>
	<AANLkTikLtKhBiCZeMY+GH6X7jb4wVdybVF=OiMePK8Rm@mail.gmail.com>
	<AANLkTimyRrtba8nD7jkWJY2DysebRaycpvRXY3b_A3r-@mail.gmail.com>
	<AANLkTinrSNUr49b3LE6c6ZP7NZ9d13t1CK2bPW2x+anD@mail.gmail.com>
	<AANLkTi=HOiA38khKujHo4y=oP90OKP4Pcmd-L5m70P7K@mail.gmail.com>
	<AANLkTin7NowMjGG7Ze4XOgUr0qn9XMoKG4SK9M7_be0r@mail.gmail.com>
	<AANLkTi=g=9g8qAokDzROk=8-2SJy4TTUOpvPmp0uD_N6@mail.gmail.com>
	<AANLkTikC052SeKz6=1yoxa9A3HD-CnhdjwfUmWLfqV0J@mail.gmail.com>
Message-ID: <AANLkTi=8g_CGOGW_9PumBKGzOagbkhKa6o70Af97mRxY@mail.gmail.com>

Ok, so I've finally found the problem. I'm connecting to the softphone first
then transferring to the gateway, but in the a-leg connection to the
softphone I had set ignore_early_media=true. I tried removing that and it
works now.

Presumably it's correct that SDP wouldn't be sent at 180 Ringing with that
flag on - but that's above my pay-grade, I'm just happy to have it working.

Thanks for your help Anthony,

Regards,
Fraser




On Fri, Oct 29, 2010 at 11:38 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> Ah, ok, thanks, I'll do some more tests with the trace running, and see
> whats happening with different softphones and gateways.
>
> Thanks,
> Fraser
>
>
>
>
>
> On Fri, Oct 29, 2010 at 10:53 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> I'm not sure what you mean it doesn't fit.
>>
>> Did you look at your trace you sent me in wireshark ?  (see bad_call.png)
>> You can see the bria phone answer 180 ringing with no sdp in packet 7
>> Then it starts sending RTP .. ??? That is completely wrong.
>>
>> It never told us it's sending RTP how can we receive it?  We don't have a
>> SDP.
>>
>>
>> Then in bad_call2.png it sends a 200+sdp (yes that's what we wanted)
>> Now you have 2 way audio.
>>
>>
>>
>>
>> On Fri, Oct 29, 2010 at 4:23 PM, Fraser Redmond <fraserredmond at gmail.com>
>> wrote:
>> > Thanks, but that doesn't fit - I've tried it with Bria, and with the
>> recent
>> > version of XLite 4, and its happened on both. They're both made by
>> > Counterpath, so I've also tried 3CXphone, and it's happening on that
>> too.
>> >
>> > More suspicious though is that it only happens with some gateways, and
>> not
>> > others. Or did you mean that you think it's a bug with the gateway's
>> > systems? The two I've tried that have this problem are both big outfits,
>> not
>> > some guy operating out of his garage :-) Also, both were already listed
>> in
>> > the Gateways on the wiki, so I'm guessing someone else is already using
>> them
>> > with Freeswitch.
>> >
>> > My server is running on Amazon AWS - could it be a timing/virtualization
>> > type problem, or something like that?
>> >
>> > Any other ideas?
>> >
>> > Cheers,
>> > Fraser
>> >
>> >
>> >
>> >
>> > On Fri, Oct 29, 2010 at 9:39 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> OK so,
>> >> The phone sends a 180 ringing with NO SDP
>> >> then it starts sending RTP
>> >> That's is not right.  It's a bug in the phone.
>> >>
>> >>
>> >>
>> >> On Fri, Oct 29, 2010 at 1:35 PM, Anthony Minessale
>> >> <anthony.minessale at gmail.com> wrote:
>> >> > can you try another one with just udp and not "port 5060"
>> >> > so I can see the rtp too
>> >> >
>> >> >
>> >> > On Fri, Oct 29, 2010 at 12:28 PM, Fraser Redmond
>> >> > <fraserredmond at gmail.com> wrote:
>> >> >> Thanks Anthony, it's here:
>> >> >> http://pastebin.freeswitch.org/14350
>> >> >>
>> >> >> And pcap is attached.
>> >> >>
>> >> >> The call connects around (or just before) the 16:58:35 mark (line
>> 558
>> >> >> is
>> >> >> what I see in the terminal while waiting for it to connect - both
>> >> >> early-media and the missing start of the media)
>> >> >>
>> >> >> Cheers,
>> >> >> Fraser
>> >> >>
>> >> >>
>> >> >>
>> >> >>
>> >> >> On Thu, Oct 28, 2010 at 3:59 PM, Anthony Minessale
>> >> >> <anthony.minessale at gmail.com> wrote:
>> >> >>>
>> >> >>> Can you do this trace with debug level logging in addition to the
>> sip
>> >> >>> trace
>> >> >>> console loglevel debug
>> >> >>>
>> >> >>> you also may want to get a pcap of it
>> >> >>>
>> >> >>> tshark udp and port 5060 -w test.pcap
>> >> >>>
>> >> >>>
>> >> >>>
>> >> >>> On Sat, Oct 23, 2010 at 11:44 AM, Fraser Redmond
>> >> >>> <fraserredmond at gmail.com> wrote:
>> >> >>> > Thanks Anthony,
>> >> >>> >
>> >> >>> > Finally managed to get a sip trace - could you do me a favor and
>> >> >>> > take a
>> >> >>> > look
>> >> >>> > and/or give me some ideas of what to look for?
>> >> >>> >
>> >> >>> > http://pastebin.freeswitch.org/14300
>> >> >>> >
>> >> >>> > I've highlighted lines 168 and 193. In between these lines is
>> where
>> >> >>> > the
>> >> >>> > number is dialed and rings once, then picks up, then theres
>> silence
>> >> >>> > for
>> >> >>> > a
>> >> >>> > second or two, and that second SIP message is when I start
>> hearing
>> >> >>> > audio.
>> >> >>> >
>> >> >>> > Thanks,
>> >> >>> > Fraser
>> >> >>> >
>> >> >>> >
>> >> >>> >
>> >> >>> >
>> >> >>> > On Thu, Oct 21, 2010 at 6:44 PM, Anthony Minessale
>> >> >>> > <anthony.minessale at gmail.com> wrote:
>> >> >>> >>
>> >> >>> >> its a blue message on cli
>> >> >>> >>
>> >> >>> >> It could also be the other side expecting us to send media first
>> or
>> >> >>> >> something silly.
>> >> >>> >> try getting a sip trace and figure out when the rtp starts
>> >> >>> >> arriving.
>> >> >>> >>
>> >> >>> >>
>> >> >>> >> On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond
>> >> >>> >> <fraserredmond at gmail.com> wrote:
>> >> >>> >> > Sorry, yes, I am setting ignore_early_media=true in the first
>> >> >>> >> > area.
>> >> >>> >> > (Or
>> >> >>> >> > are
>> >> >>> >> > you saying that should be off? I forget now why I needed it
>> on,
>> >> >>> >> > but
>> >> >>> >> > there
>> >> >>> >> > was a reason I added it.)
>> >> >>> >> >
>> >> >>> >> > Yes, the bridge doesn't start until after the A-leg has
>> answered.
>> >> >>> >> >
>> >> >>> >> > Thanks for the suggestion about nat/auto-changing port, I'll
>> have
>> >> >>> >> > a
>> >> >>> >> > look
>> >> >>> >> > into that - would that be in the cli output or in a sip trace?
>> >> >>> >> > I've
>> >> >>> >> > already
>> >> >>> >> > looked and it's not appearing in the CLI output (with
>> >> >>> >> > loglevel=debug),
>> >> >>> >> > haven't looked in the sip trace yet.
>> >> >>> >> >
>> >> >>> >> > Cheers,
>> >> >>> >> > Fraser
>> >> >>> >> >
>> >> >>> >> >
>> >> >>> >> >
>> >> >>> >> >
>> >> >>> >> > On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale
>> >> >>> >> > <anthony.minessale at gmail.com> wrote:
>> >> >>> >> >>
>> >> >>> >> >> are you setting ignore_early_media=true in the first
>> vars=values
>> >> >>> >> >> area?
>> >> >>> >> >>
>> >> >>> >> >> This looks like you could be calling one leg who is still not
>> >> >>> >> >> answered
>> >> >>> >> >> and then bridging it to another dest.  The bridge app will
>> wait
>> >> >>> >> >> for
>> >> >>> >> >> the first leg to answer before bridging.
>> >> >>> >> >>
>> >> >>> >> >> Also if you have any NAT anywhere, look for an "auto-changing
>> >> >>> >> >> port"
>> >> >>> >> >> type message which can also be attributed to this due to a
>> >> >>> >> >> detection
>> >> >>> >> >> period for incorrect ports.
>> >> >>> >> >>
>> >> >>> >> >>
>> >> >>> >> >>
>> >> >>> >> >> On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond
>> >> >>> >> >> <fraserredmond at gmail.com> wrote:
>> >> >>> >> >> > event_socket:
>> >> >>> >> >> > api originate {vars=values}user/$fromExtn at Domain
>> >> >>> >> >> > 'set:bLegVars=values,transfer:$toNum xml outbound_call'
>> inline
>> >> >>> >> >> >
>> >> >>> >> >> > then
>> >> >>> >> >> >
>> >> >>> >> >> > dialplan:
>> >> >>> >> >> > <action application="set"
>> >> >>> >> >> > data="effective_caller_id_number=+1800number"/>
>> >> >>> >> >> > <action application="set"
>> data="effective_caller_id_name="/>
>> >> >>> >> >> > (set and/or export a bunch of other vars too)
>> >> >>> >> >> > <action application="set"
>> >> >>> >> >> >
>> >> >>> >> >> >
>> >> >>> >> >> >
>> >> >>> >> >> >
>> >> >>> >> >> > data="dial_string=sofia/gateway/
>> gatewayname.com/00${destination_number}<http://gatewayname.com/00$%7Bdestination_number%7D>
>> "/>
>> >> >>> >> >> > <action application="bridge" data="${dial_string}"/>
>> >> >>> >> >> >
>> >> >>> >> >> > Cheers,
>> >> >>> >> >> > Fraser
>> >> >>> >> >> >
>> >> >>> >> >> >
>> >> >>> >> >> >
>> >> >>> >> >> >
>> >> >>> >> >> > On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale
>> >> >>> >> >> > <anthony.minessale at gmail.com> wrote:
>> >> >>> >> >> >>
>> >> >>> >> >> >> how are you accomplishing that? by which technique?
>> >> >>> >> >> >>
>> >> >>> >> >> >> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond
>> >> >>> >> >> >> <fraserredmond at gmail.com> wrote:
>> >> >>> >> >> >> > The call is originated from Freeswitch (via CLI) to a
>> >> >>> >> >> >> > softphone,
>> >> >>> >> >> >> > then
>> >> >>> >> >> >> > when
>> >> >>> >> >> >> > that is connected it bridges out to the gateway.
>> >> >>> >> >> >> >
>> >> >>> >> >> >> > Cheers,
>> >> >>> >> >> >> > Fraser
>> >> >>> >> >> >> >
>> >> >>> >> >> >> >
>> >> >>> >> >> >> >
>> >> >>> >> >> >> >
>> >> >>> >> >> >> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale
>> >> >>> >> >> >> > <anthony.minessale at gmail.com> wrote:
>> >> >>> >> >> >> >>
>> >> >>> >> >> >> >> Where is the other side of this call coming from?
>> >> >>> >> >> >> >>
>> >> >>> >> >> >> >> [ (   ) ] -> FS -> (PSTN via SIP)
>> >> >>> >> >> >> >>
>> >> >>> >> >> >> >> What goes in the empty space above?
>> >> >>> >> >> >
>> >> >>> >> >> >
>> >> >>> >> >> > _______________________________________________
>> >> >>> >> >> > FreeSWITCH-users mailing list
>> >> >>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >>> >> >> >
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>> >> >> >
>> >> >>> >> >> >
>> >> >>> >> >> >
>> >> >>> >> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> >> >> > 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<MSN%3Aanthony_minessale at hotmail.com>
>> >> >>> >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> >> >>> >> >> IRC: irc.freenode.net #freeswitch
>> >> >>> >> >>
>> >> >>> >> >> FreeSWITCH Developer Conference
>> >> >>> >> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >> >>> >> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >> >>> >> >> pstn:+19193869900
>> >> >>> >> >>
>> >> >>> >> >> _______________________________________________
>> >> >>> >> >> FreeSWITCH-users mailing list
>> >> >>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >>> >> >>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>> >> >>
>> >> >>> >> >>
>> >> >>> >> >>
>> >> >>> >> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> >> >> http://www.freeswitch.org
>> >> >>> >> >
>> >> >>> >> >
>> >> >>> >> > _______________________________________________
>> >> >>> >> > FreeSWITCH-users mailing list
>> >> >>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>> >> >
>> >> >>> >> >
>> >> >>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> >> > 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<MSN%3Aanthony_minessale at hotmail.com>
>> >> >>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> >> >>> >> IRC: irc.freenode.net #freeswitch
>> >> >>> >>
>> >> >>> >> FreeSWITCH Developer Conference
>> >> >>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >> >>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >> >>> >> pstn:+19193869900
>> >> >>> >>
>> >> >>> >> _______________________________________________
>> >> >>> >> FreeSWITCH-users mailing list
>> >> >>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>> >>
>> >> >>> >>
>> >> >>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> >> http://www.freeswitch.org
>> >> >>> >
>> >> >>> >
>> >> >>> > _______________________________________________
>> >> >>> > FreeSWITCH-users mailing list
>> >> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>> >
>> >> >>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> > 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<MSN%3Aanthony_minessale at hotmail.com>
>> >> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> >> >>> IRC: irc.freenode.net #freeswitch
>> >> >>>
>> >> >>> FreeSWITCH Developer Conference
>> >> >>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >> >>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >> >>> pstn:+19193869900
>> >> >>>
>> >> >>> _______________________________________________
>> >> >>> FreeSWITCH-users mailing list
>> >> >>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>
>> >> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> http://www.freeswitch.org
>> >> >>
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> 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<MSN%3Aanthony_minessale at hotmail.com>
>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> >> > IRC: irc.freenode.net #freeswitch
>> >> >
>> >> > FreeSWITCH Developer Conference
>> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >> > pstn:+19193869900
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> 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<MSN%3Aanthony_minessale at hotmail.com>
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >> pstn:+19193869900
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > 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 <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From mario_fs at mgtech.com  Sat Oct 30 09:37:17 2010
From: mario_fs at mgtech.com (Mario G)
Date: Sat, 30 Oct 2010 09:37:17 -0700
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <AF7D33D9-8198-45FB-8C5E-48121D90C3D2@mgtech.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
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	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
	<AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
	<6D9FD5E2-21FA-4630-8CF4-A87ADB30AC87@mgtech.com>
	<AANLkTi=BiDiXLgEoj_MfwmZbGjvXoOuNJgUYdetciNCj@mail.gmail.com>
	<9BAFDAF8-1E66-444C-A024-A88FF65D8BCA@mgtech.com>
	<3447E946-0BF8-4F7E-B5E9-E28F53DD0968@mgtech.com>
	<AANLkTi=LRbz6ADhLvBKkjGxOywAVLf--GTkzXEpUoDbi@mail.gmail.com>
	<BE754258-BE27-454F-B740-527A35DBDFFE@mgtech.com>
	<AANLkTi=3Kd-bCHJSED7VEh-cvtpdXMs_EkWZL=s8J4Ux@mail.gmail.com>
	<14205291-75B3-470F-ABA1-EE7F08A695E9@mgtech.com>
	<AANLkTi=D6siEAquSq6gYh-RnZeJJWrk11mHWK8De2tLh@mail.gmail.com>
	<2C35DBE8-13B3-4264-BB43-7496A1054AD1@mgtech.com>
	<86E8AE3F-E945-4952-A438-C674F23881D6@mgtech.com>
	<AANLkTin5ni_P-D5+XnCKugoRLgoExACV=eBWo3496AXq@mail.gmail.com>
	<317A03A2-FE67-4D0D-BBA9-0661079DC61E@mgtech.com>
	<AANLkTik62y_uSx42tTsb9W6wGOFZ=bd7AV2pBcb=cAjg@mail.gmail.com>
	<AF7D33D9-8198-45FB-8C5E-48121D90C3D2@mgtech.com>
Message-ID: <3CAA3F8C-C2DC-48E2-89C5-9BC352C4B449@mgtech.com>

Setting sofia loglevel to 9 made Terminal app memory go from 20meg to eventually using all memory and locking the machine up. I spent all night trying to get a good trace. I need to get one when the failure occurs early before the lockup. Will be working all w/e on this. Thanks for you help.

I got this with only 1 gateway defined but did not catch it early. Notice how there is no SIP trace activity:
2010-10-30 09:30:15.046900 [WARNING] sofia_reg.c:387 mvvyl Failed Registration, setting retry to 15 seconds.
nta: timer K fired, terminate NOTIFY (3892042)
outgoing_reclaim_all(0x0, 0x0, 0x102309cb0)
nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
nta: timer not set
2010-10-30 09:30:31.926382 [NOTICE] sofia_reg.c:342 Registering mvvyl
nua: nua_register: entering
nua(0x1003c3750): sent signal r_register
nua(0x1003c3750): recv signal r_register
2010-10-30 09:30:47.685212 [WARNING] sofia_reg.c:387 mvvyl Failed Registration, setting retry to 15 seconds.
2010-10-30 09:31:03.478941 [NOTICE] sofia_reg.c:342 Registering mvvyl
nua: nua_register: entering
nua(0x1003c3750): sent signal r_register
nua(0x1003c3750): recv signal r_register
2010-10-30 09:31:19.219346 [WARNING] sofia_reg.c:387 mvvyl Failed Registration, setting retry to 15 seconds.
2010-10-30 09:31:36.208448 [NOTICE] sofia_reg.c:342 Registering mvvyl
nua: nua_register: entering
nua(0x1003c3750): sent signal r_register
nua(0x1003c3750): recv signal r_register

Mario

> 
> On Oct 29, 2010, at 5:43 PM, Anthony Minessale wrote:
> 
>> also raise loglevel to debug
>> console loglevel debug
>> 
>> 
>> On Fri, Oct 29, 2010 at 7:29 PM, Mario G <mario_fs at mgtech.com> wrote:
>>> Will so, BTW, I mentioned below one is dns and another uses IP, I tested that theory using an IP  does not help. I also updated to todays git version and no help there. Will post when the trace is done.
>>> 
>>> On Oct 29, 2010, at 5:17 PM, Anthony Minessale wrote:
>>> 
>>>> can you repeat that trace with sofia debug on
>>>> sofia loglevel all 9
>>>> 
>>>> Are you doing DNS by any chance in the gateway "proxy" param?
>>>> you could try filling in the register-proxy param in your gateway to
>>>> sip:<ip> <-- not dns but ip that dns resolves to
>>>> <param name="register-proxy" value="sip:1.2.3.4"/>
>>>> 
>>>> I'm just guessing but Its possible some bad dns query could be throwing FS off.
>>>> so this test would force all the packets to the exact IP of your host
>>>> instead of looking it up.
>>>> 
>>>> 
>>>> 
>>>> On Fri, Oct 29, 2010 at 6:30 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>> Oh Boy: Not only is everything set to the static route but I turned
>>>>> off/disconnected the dynamic DSL line so I only had 1 static line to the
>>>>> router. The second wan is set in the router off. Even turned off router uPnP
>>>>> even though I am using -nonat. Guess what.... I still have the problem. Look
>>>>> like when FS says is going to retry it actually does not.
>>>>> Here is a short one: http://pastebin.freeswitch.org/14359 - I caught one
>>>>> right after I started FS, failure occurred in minutes (lucky). Look at the
>>>>> bottom of the trace, you see SIP trace activity and then when it fails no
>>>>> SIP trace activity. Could this possibly be a FS bug? (I am a mainframe
>>>>> assembler systems programmer and I might think so if there were error retry
>>>>> messages but nothing showing in one of my traces...)
>>>>> Notes:
>>>>> idone is gateway 1
>>>>> idtwo is gateway 2  I had to trace both because it was impossible to figure
>>>>> out which one would fail first. Ran several times but kept missing the right
>>>>> one.
>>>>> I use a url for one gateway and ip for another but it makes no difference
>>>>> since both eventually fail.
>>>>> 10. is local lan
>>>>> 210. is external ip
>>>>> 216. is itsp
>>>>> 
>>>>> Here is a longer one from earlier   http://pastebin.freeswitch.org/14357
>>>>> Notes:
>>>>> A call was received and hung up for idtwo - beginning of trace
>>>>> 11 minutes later idtwo failed - see last line of trace
>>>>> 
>>>>> Thank you very much!
>>>>> Mario
>>>>> 
>>>>> On Oct 29, 2010, at 12:11 PM, Anthony Minessale wrote:
>>>>> 
>>>>> stun-enabled must be true in your profile XML to see what you pasted.
>>>>> 
>>>>> Get me a sip trace of this from when it works until when it fails
>>>>> 
>>>>> only enable the sip trace on the profile with the gateway to reduce traffic
>>>>> 
>>>>> 
>>>>> 
>>>>> On Fri, Oct 29, 2010 at 1:56 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>> 
>>>>> Oh my...  looks it was not NAT after all? Please help! I changed to all
>>>>> profiles to static per instructions below and still have the problem:
>>>>> 
>>>>> 2010-10-29 11:15:18.536446 [NOTICE] sofia_reg.c:342 Registering uuid1
>>>>> 
>>>>> 2010-10-29 11:15:34.313150 [WARNING] sofia_reg.c:387 uuid1 Failed
>>>>> Registration, setting retry to 15 seconds.
>>>>> 
>>>>> sofia global siptrace on did not show any activity for this gateway in or
>>>>> out, others were fine but eventually fail. I setup static:
>>>>> 
>>>>> 1. set the params ext-sip-ip and ext-rtp-ip to my external static IP
>>>>> 
>>>>> 2. map the sip ports (5060-5080) and all of the rtp ports (16384-32767)  to
>>>>> FS lan addr.
>>>>> 
>>>>> 3. set sip-ip and rtp-ip to the lan addr of FS
>>>>> 
>>>>> 4. start FS with -nonat
>>>>> 
>>>>> I don't know what to try next. BTW, the sofia status for the profiles shows
>>>>> stun enabled but I did not set it up anywhere:
>>>>> 
>>>>> Name                    uuid1
>>>>> 
>>>>> Domain Name             N/A
>>>>> 
>>>>> Auto-NAT                false
>>>>> 
>>>>> DBName                  sofia_reg_mvvyl
>>>>> 
>>>>> Pres Hosts
>>>>> 
>>>>> Dialplan                XML
>>>>> 
>>>>> Context                 public
>>>>> 
>>>>> Challenge Realm         auto_to
>>>>> 
>>>>> RTP-IP                  10.x.x.20
>>>>> 
>>>>> Ext-RTP-IP              210.x.x.100
>>>>> 
>>>>> SIP-IP                  10.x.x.20
>>>>> 
>>>>> Ext-SIP-IP              210.x.x.100
>>>>> 
>>>>> URL                     sip:mod_sofia at 210.x.x.100:5068
>>>>> 
>>>>> BIND-URL                sip:mod_sofia at 210.x.x.100:5068;maddr=10.x.x.20
>>>>> 
>>>>> HOLD-MUSIC              local_stream://moh
>>>>> 
>>>>> OUTBOUND-PROXY          N/A
>>>>> 
>>>>> CODECS IN               G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
>>>>> 
>>>>> CODECS OUT              PCMU,PCMA,GSM
>>>>> 
>>>>> TEL-EVENT               101
>>>>> 
>>>>> DTMF-MODE               rfc2833
>>>>> 
>>>>> CNG                     13
>>>>> 
>>>>> SESSION-TO              0
>>>>> 
>>>>> MAX-DIALOG              0
>>>>> 
>>>>> NOMEDIA                 false
>>>>> 
>>>>> LATE-NEG                false
>>>>> 
>>>>> PROXY-MEDIA             false
>>>>> 
>>>>> AGGRESSIVENAT           false
>>>>> 
>>>>> STUN-ENABLED            true
>>>>> 
>>>>> STUN-AUTO-DISABLE       false
>>>>> 
>>>>> CALLS-IN                0
>>>>> 
>>>>> FAILED-CALLS-IN         0
>>>>> 
>>>>> CALLS-OUT               0
>>>>> 
>>>>> FAILED-CALLS-OUT        0
>>>>> 
>>>>> 
>>>>> On Oct 27, 2010, at 10:04 AM, Anthony Minessale wrote:
>>>>> 
>>>>> if you map it or not, a scanner would penetrate it.
>>>>> 
>>>>> There are lot of sip scanners out there now, you just need to beware of
>>>>> them.
>>>>> 
>>>>> 
>>>>> On Wed, Oct 27, 2010 at 11:50 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>>> 
>>>>> Thanks so much! I am sure many others will find this info invaluable. I will
>>>>> try the static route again but have one question: When I started with FS I
>>>>> found a "sip scanner"  in FS and someone on this group said not to use port
>>>>> mapping since it was a security risk. Is that true?
>>>>> 
>>>>> On Oct 27, 2010, at 9:10 AM, Anthony Minessale wrote:
>>>>> 
>>>>> you are completely guessing at things.
>>>>> 
>>>>> I want you to understand that the only reason you are having problems
>>>>> 
>>>>> with this is because you don't understand how it works enough to know
>>>>> 
>>>>> what you are doing 100%
>>>>> 
>>>>> Its a given that the pnp stuff is only for your dynamic IP.
>>>>> 
>>>>> aggressive-nat-detection and sip-force-expires are all related to
>>>>> 
>>>>> inbound calls when the things who are registering to you may be behind
>>>>> 
>>>>> nat.
>>>>> 
>>>>> You need to learn the difference between which nat tools are
>>>>> 
>>>>> *) designed for your FS to run behind nat
>>>>> 
>>>>> *) designed for FS to run public and accept connections from devices behind
>>>>> nat.
>>>>> 
>>>>> If you have a static IP, you don't need the pnp stuff so -nonat is fine
>>>>> 
>>>>> What you need to do is set
>>>>> 
>>>>> 1) set the params ext-sip-ip and ext-rtp-ip to your external static IP
>>>>> 
>>>>> 2) map the sip ports and all of the rtp ports from your static IP to FS lan
>>>>> addr
>>>>> 
>>>>> 3) set sip-ip and rtp-ip to the lan addr you forwarded through.
>>>>> 
>>>>> 
>>>>> If you don't do this: your outbound registration will use NAT to your
>>>>> 
>>>>> provider and if there is no activity for the expire time on your NAT
>>>>> 
>>>>> mapping the reverse port mapping from your provider back to you is
>>>>> 
>>>>> lost.  This is why you set your register expires to a very low number,
>>>>> 
>>>>> (you need to make sure the provider does not turn the expires back up
>>>>> 
>>>>> in the reply because it will beat your choice *see sip trace) if this
>>>>> 
>>>>> is the case then you need the "ping" option set to 30, to continuously
>>>>> 
>>>>> send an options to your provider.
>>>>> 
>>>>> The static mapping is obviously the better, easier and more reliable
>>>>> solution.
>>>>> 
>>>>> So I want you to understand that the only way to keep a nat mapped
>>>>> 
>>>>> port alive is to continuously send traffic, all the other methods that
>>>>> 
>>>>> you are mentioning are to detect that phones registered to your are
>>>>> 
>>>>> behind nat, I gave you that force-expires option before because your
>>>>> 
>>>>> trace was full of inbound reg so I thought that is what you wanted
>>>>> 
>>>>> help with.
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> On Wed, Oct 27, 2010 at 10:43 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>>> 
>>>>> I should mention that I did not have this problem with an SPA9000 PBX
>>>>> 
>>>>> (asterisk based) for over two years so FS may be pickier about upnp and/or
>>>>> 
>>>>> nat, or just better at it exposing a problem in the router.
>>>>> 
>>>>> I made different changes to the gateways to test different things. One
>>>>> 
>>>>> failed after 17 hours, the other two stayed up.  What did not work:
>>>>> 
>>>>> added <variable name="sip-force-expires" value="30"/> to the directory
>>>>> 
>>>>> entries as suggested.
>>>>> 
>>>>> set the gateway expire times to 30 seconds.
>>>>> 
>>>>> What worked (could be coincidental) for the two gateways that stayed up:
>>>>> 
>>>>> I Added <param name="aggressive-nat-detection" value="true"/>
>>>>> 
>>>>> I originally setup FS to use the static ip by setting external sip/rtp to
>>>>> 
>>>>> just the static ip (no autonat:) and ran with -nonat but I could not get
>>>>> 
>>>>> incoming calls. The only way it worked was to use autonat:1.2.3.4. The
>>>>> 
>>>>> router has 1 static public address and 1 dynamic external IP, this is the
>>>>> 
>>>>> root of the problem, upnp only tells FS about the dynamic ip  Will keep this
>>>>> 
>>>>> thread up-to-date for anyone who may be in the same boat someday. Thanks
>>>>> 
>>>>> again for looking at the trace.
>>>>> 
>>>>> Mario
>>>>> 
>>>>> You should be setting the req freq to a low number on the outbound gateways
>>>>> 
>>>>> The examples you showed had a series of inbound reg
>>>>> 
>>>>> also set expire-seconds to 30 in your gateway xml
>>>>> 
>>>>> 
>>>>> The problem is if you are not constantly sending traffic to the box
>>>>> 
>>>>> the nat mapping will go away.
>>>>> 
>>>>> If you are in production you should be using a static ip with a static
>>>>> 
>>>>> mapping, any trouble you are having is your own fault for playing with
>>>>> 
>>>>> fire.  The best we can do is tell you how to keep it contained.
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>> 
>>>>> I made the change. I had no idea the settings for the inside phones effected
>>>>> 
>>>>> nat for the outside sip accounts. I was looking into aggressive-nat-
>>>>> 
>>>>> detection since the internal profile status always shows the right external
>>>>> 
>>>>> static IP but the nat_ap status always shows the dynamic ip. Crossing
>>>>> 
>>>>> fingers/etc since this problem is 85% of time (weeks!) into FS changeover.
>>>>> 
>>>>> Thanks!
>>>>> 
>>>>> Mario
>>>>> 
>>>>> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
>>>>> 
>>>>> add
>>>>> 
>>>>> <variable name="sip-force-expires" value="30"/>
>>>>> 
>>>>> to the <variables> section of your <user>
>>>>> 
>>>>> you have it at 600 and the nat mapping is timing out while the 600
>>>>> 
>>>>> seconds is ticking away
>>>>> 
>>>>> 
>>>>> 
>>>>> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>> 
>>>>> From the TSP:
>>>>> 
>>>>> "I have enabled the SIP trace on your account. We are not currently seeing
>>>>> 
>>>>> any registration attempts to your account within the last 15 minutes. Please
>>>>> 
>>>>> restart FreeSwitch so that registration attempts begin again. Thank you. ".
>>>>> 
>>>>> So FS is not getting past router.
>>>>> 
>>>>> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>>>>> 
>>>>> I ran the global trace during the problem and it is
>>>>> 
>>>>> at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
>>>>> 
>>>>> "accttwo", "acct3". The trace includes phones since it was global. I am
>>>>> 
>>>>> using:
>>>>> 
>>>>>  <param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>>>>> 
>>>>>  <param name="ext-sip-ip" value="autonat:my-static.ip"/>
>>>>> 
>>>>> I tried dumping nat and removing the autonat: above and using -nonat but
>>>>> 
>>>>> that did not work, registration proceeded but no calls inbound.
>>>>> 
>>>>> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>>>>> 
>>>>> Whoops, I am using an IP address for at least one gateway so that is not the
>>>>> 
>>>>> problem:
>>>>> 
>>>>> They look outbound to me and I am using dns for 2 and an IP for one so that
>>>>> 
>>>>> is not the issue. I was able to get FS to clear this up by doing "nat_map
>>>>> 
>>>>> reinit" which is why I think this is a nat problem. I will do the trace you
>>>>> 
>>>>> mentioned. I will plug an ip address into one of the gateways to see what
>>>>> 
>>>>> happens, they all fail at once. Thanks for responding!
>>>>> 
>>>>> Mario
>>>>> 
>>>>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>>>>> 
>>>>> I really need help on this as I have weeks into this problem. I thought I
>>>>> 
>>>>> had it nailed but I guess not. After 5.5 hours I get:
>>>>> 
>>>>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>>>>> 
>>>>> Registration, setting retry to 15 seconds.
>>>>> 
>>>>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>>> 
>>>>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>>> 
>>>>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
>>>>> 
>>>>> Registration, setting retry to 30 seconds.
>>>>> 
>>>>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>>>>> 
>>>>> Registration, setting retry to 15 seconds.
>>>>> 
>>>>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>>>>> 
>>>>> Registration, setting retry to 15 seconds.
>>>>> 
>>>>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>>> 
>>>>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>>> 
>>>>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>>>>> 
>>>>> and no way to make/get calls until I restart FS. I did this:
>>>>> 
>>>>> 1. log 7
>>>>> 
>>>>> 2. sofia profile xxxx siptrace on   for each profile/gateway
>>>>> 
>>>>> 3. restarted router
>>>>> 
>>>>> All three did not solve the problem. The trace and log produced no
>>>>> 
>>>>> additional lines which is why I am wondering if FS has a problem since the
>>>>> 
>>>>> trace shows no SIP activity.
>>>>> 
>>>>> 3 gateways with 2 ITSPs
>>>>> 
>>>>> 2 DSL/WAN lines, 1 static and 1 dynamic
>>>>> 
>>>>> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
>>>>> 
>>>>> external static ip.
>>>>> 
>>>>> sofia status profile ... has the right ext ip
>>>>> 
>>>>> nat_map status shows the dynamic (wrong) IP
>>>>> 
>>>>> I tried starting with -nonat but that was worse
>>>>> 
>>>>> the only way to fix is restart FS.
>>>>> 
>>>>> I read the wiki on external nat, auto_nat and everything else many times.
>>>>> 
>>>>> Thanks Mario
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> 
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> 
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> 
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> 
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> 
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> 
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> _______________________________________________
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> 
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> 
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> 
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> 
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> 
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> 
>>>>> 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
>>>>> 
>>>>> _______________________________________________
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> 
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> 
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> 
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> 
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> 
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> 
>>>>> 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
>>>>> 
>>>>> _______________________________________________
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> 
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> 
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> 
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> 
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> 
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> 
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> 
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> 
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> 
>>>>> 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
>>>>> 
>>>>> _______________________________________________
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> 
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> 
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> 
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> 
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> 
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> 
>>>>> 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
>>>>> 
>>>>> _______________________________________________
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> 
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> 
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> 
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> 
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> 
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> 
>>>>> 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
>>>>> 
>>>>> _______________________________________________
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> 
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> 
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> 
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> 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
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> 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
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Sat Oct 30 12:34:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 30 Oct 2010 14:34:23 -0500
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <3CAA3F8C-C2DC-48E2-89C5-9BC352C4B449@mgtech.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
	<407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>
	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
	<AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
	<6D9FD5E2-21FA-4630-8CF4-A87ADB30AC87@mgtech.com>
	<AANLkTi=BiDiXLgEoj_MfwmZbGjvXoOuNJgUYdetciNCj@mail.gmail.com>
	<9BAFDAF8-1E66-444C-A024-A88FF65D8BCA@mgtech.com>
	<3447E946-0BF8-4F7E-B5E9-E28F53DD0968@mgtech.com>
	<AANLkTi=LRbz6ADhLvBKkjGxOywAVLf--GTkzXEpUoDbi@mail.gmail.com>
	<BE754258-BE27-454F-B740-527A35DBDFFE@mgtech.com>
	<AANLkTi=3Kd-bCHJSED7VEh-cvtpdXMs_EkWZL=s8J4Ux@mail.gmail.com>
	<14205291-75B3-470F-ABA1-EE7F08A695E9@mgtech.com>
	<AANLkTi=D6siEAquSq6gYh-RnZeJJWrk11mHWK8De2tLh@mail.gmail.com>
	<2C35DBE8-13B3-4264-BB43-7496A1054AD1@mgtech.com>
	<86E8AE3F-E945-4952-A438-C674F23881D6@mgtech.com>
	<AANLkTin5ni_P-D5+XnCKugoRLgoExACV=eBWo3496AXq@mail.gmail.com>
	<317A03A2-FE67-4D0D-BBA9-0661079DC61E@mgtech.com>
	<AANLkTik62y_uSx42tTsb9W6wGOFZ=bd7AV2pBcb=cAjg@mail.gmail.com>
	<AF7D33D9-8198-45FB-8C5E-48121D90C3D2@mgtech.com>
	<3CAA3F8C-C2DC-48E2-89C5-9BC352C4B449@mgtech.com>
Message-ID: <AANLkTinPZQwmnqUYQWwwke1OW_GBKFmVWbeXFoGoJV3n@mail.gmail.com>

ok so there must be a problem in MAC, have you compared the results on
win32 or linux?
some other people are reporting mem issues on mac lets find it.

http://valgrind.org/downloads/valgrind-3.6.0.tar.bz2

build and install valgrind

run fs in vg mode
valgrind --tool=memcheck --log-file=vg.log --leak-check=full
--leak-resolution=high --show-reachable=yes /path/to/lt-freeswitch -vg

do something that eats mem

exit and get me the vg.log



On Sat, Oct 30, 2010 at 11:37 AM, Mario G <mario_fs at mgtech.com> wrote:
> Setting sofia loglevel to 9 made Terminal app memory go from 20meg to eventually using all memory and locking the machine up. I spent all night trying to get a good trace. I need to get one when the failure occurs early before the lockup. Will be working all w/e on this. Thanks for you help.
>
> I got this with only 1 gateway defined but did not catch it early. Notice how there is no SIP trace activity:
> 2010-10-30 09:30:15.046900 [WARNING] sofia_reg.c:387 mvvyl Failed Registration, setting retry to 15 seconds.
> nta: timer K fired, terminate NOTIFY (3892042)
> outgoing_reclaim_all(0x0, 0x0, 0x102309cb0)
> nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
> nta: timer not set
> 2010-10-30 09:30:31.926382 [NOTICE] sofia_reg.c:342 Registering mvvyl
> nua: nua_register: entering
> nua(0x1003c3750): sent signal r_register
> nua(0x1003c3750): recv signal r_register
> 2010-10-30 09:30:47.685212 [WARNING] sofia_reg.c:387 mvvyl Failed Registration, setting retry to 15 seconds.
> 2010-10-30 09:31:03.478941 [NOTICE] sofia_reg.c:342 Registering mvvyl
> nua: nua_register: entering
> nua(0x1003c3750): sent signal r_register
> nua(0x1003c3750): recv signal r_register
> 2010-10-30 09:31:19.219346 [WARNING] sofia_reg.c:387 mvvyl Failed Registration, setting retry to 15 seconds.
> 2010-10-30 09:31:36.208448 [NOTICE] sofia_reg.c:342 Registering mvvyl
> nua: nua_register: entering
> nua(0x1003c3750): sent signal r_register
> nua(0x1003c3750): recv signal r_register
>
> Mario
>
>>
>> On Oct 29, 2010, at 5:43 PM, Anthony Minessale wrote:
>>
>>> also raise loglevel to debug
>>> console loglevel debug
>>>
>>>
>>> On Fri, Oct 29, 2010 at 7:29 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>> Will so, BTW, I mentioned below one is dns and another uses IP, I tested that theory using an IP ?does not help. I also updated to todays git version and no help there. Will post when the trace is done.
>>>>
>>>> On Oct 29, 2010, at 5:17 PM, Anthony Minessale wrote:
>>>>
>>>>> can you repeat that trace with sofia debug on
>>>>> sofia loglevel all 9
>>>>>
>>>>> Are you doing DNS by any chance in the gateway "proxy" param?
>>>>> you could try filling in the register-proxy param in your gateway to
>>>>> sip:<ip> <-- not dns but ip that dns resolves to
>>>>> <param name="register-proxy" value="sip:1.2.3.4"/>
>>>>>
>>>>> I'm just guessing but Its possible some bad dns query could be throwing FS off.
>>>>> so this test would force all the packets to the exact IP of your host
>>>>> instead of looking it up.
>>>>>
>>>>>
>>>>>
>>>>> On Fri, Oct 29, 2010 at 6:30 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>> Oh Boy: Not only is everything set to the static route but I turned
>>>>>> off/disconnected the dynamic DSL line so I only had 1 static line to the
>>>>>> router. The second wan is set in the router off. Even turned off router uPnP
>>>>>> even though I am using -nonat. Guess what.... I still have the problem. Look
>>>>>> like when FS says is going to retry it actually does not.
>>>>>> Here is a short one: http://pastebin.freeswitch.org/14359 - I caught one
>>>>>> right after I started FS, failure occurred in minutes (lucky). Look at the
>>>>>> bottom of the trace, you see SIP trace activity and then when it fails no
>>>>>> SIP trace activity. Could this possibly be a FS bug? (I am a mainframe
>>>>>> assembler systems programmer and I might think so if there were error retry
>>>>>> messages but nothing showing in one of my traces...)
>>>>>> Notes:
>>>>>> idone is gateway 1
>>>>>> idtwo is gateway 2 ?I had to trace both because it was impossible to figure
>>>>>> out which one would fail first. Ran several times but kept missing the right
>>>>>> one.
>>>>>> I use a url for one gateway and ip for another but it makes no difference
>>>>>> since both eventually fail.
>>>>>> 10. is local lan
>>>>>> 210. is external ip
>>>>>> 216. is itsp
>>>>>>
>>>>>> Here is a longer one from earlier ? http://pastebin.freeswitch.org/14357
>>>>>> Notes:
>>>>>> A call was received and hung up for idtwo - beginning of trace
>>>>>> 11 minutes later idtwo failed - see last line of trace
>>>>>>
>>>>>> Thank you very much!
>>>>>> Mario
>>>>>>
>>>>>> On Oct 29, 2010, at 12:11 PM, Anthony Minessale wrote:
>>>>>>
>>>>>> stun-enabled must be true in your profile XML to see what you pasted.
>>>>>>
>>>>>> Get me a sip trace of this from when it works until when it fails
>>>>>>
>>>>>> only enable the sip trace on the profile with the gateway to reduce traffic
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Fri, Oct 29, 2010 at 1:56 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>>
>>>>>> Oh my... ?looks it was not NAT after all? Please help! I changed to all
>>>>>> profiles to static per instructions below and still have the problem:
>>>>>>
>>>>>> 2010-10-29 11:15:18.536446 [NOTICE] sofia_reg.c:342 Registering uuid1
>>>>>>
>>>>>> 2010-10-29 11:15:34.313150 [WARNING] sofia_reg.c:387 uuid1 Failed
>>>>>> Registration, setting retry to 15 seconds.
>>>>>>
>>>>>> sofia global siptrace on did not show any activity for this gateway in or
>>>>>> out, others were fine but eventually fail. I setup static:
>>>>>>
>>>>>> 1. set the params ext-sip-ip and ext-rtp-ip to my external static IP
>>>>>>
>>>>>> 2. map the sip ports (5060-5080) and all of the rtp ports (16384-32767) ?to
>>>>>> FS lan addr.
>>>>>>
>>>>>> 3. set sip-ip and rtp-ip to the lan addr of FS
>>>>>>
>>>>>> 4. start FS with -nonat
>>>>>>
>>>>>> I don't know what to try next. BTW, the sofia status for the profiles shows
>>>>>> stun enabled but I did not set it up anywhere:
>>>>>>
>>>>>> Name ? ? ? ? ? ? ? ? ? ?uuid1
>>>>>>
>>>>>> Domain Name ? ? ? ? ? ? N/A
>>>>>>
>>>>>> Auto-NAT ? ? ? ? ? ? ? ?false
>>>>>>
>>>>>> DBName ? ? ? ? ? ? ? ? ?sofia_reg_mvvyl
>>>>>>
>>>>>> Pres Hosts
>>>>>>
>>>>>> Dialplan ? ? ? ? ? ? ? ?XML
>>>>>>
>>>>>> Context ? ? ? ? ? ? ? ? public
>>>>>>
>>>>>> Challenge Realm ? ? ? ? auto_to
>>>>>>
>>>>>> RTP-IP ? ? ? ? ? ? ? ? ?10.x.x.20
>>>>>>
>>>>>> Ext-RTP-IP ? ? ? ? ? ? ?210.x.x.100
>>>>>>
>>>>>> SIP-IP ? ? ? ? ? ? ? ? ?10.x.x.20
>>>>>>
>>>>>> Ext-SIP-IP ? ? ? ? ? ? ?210.x.x.100
>>>>>>
>>>>>> URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 210.x.x.100:5068
>>>>>>
>>>>>> BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 210.x.x.100:5068;maddr=10.x.x.20
>>>>>>
>>>>>> HOLD-MUSIC ? ? ? ? ? ? ?local_stream://moh
>>>>>>
>>>>>> OUTBOUND-PROXY ? ? ? ? ?N/A
>>>>>>
>>>>>> CODECS IN ? ? ? ? ? ? ? G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
>>>>>>
>>>>>> CODECS OUT ? ? ? ? ? ? ?PCMU,PCMA,GSM
>>>>>>
>>>>>> TEL-EVENT ? ? ? ? ? ? ? 101
>>>>>>
>>>>>> DTMF-MODE ? ? ? ? ? ? ? rfc2833
>>>>>>
>>>>>> CNG ? ? ? ? ? ? ? ? ? ? 13
>>>>>>
>>>>>> SESSION-TO ? ? ? ? ? ? ?0
>>>>>>
>>>>>> MAX-DIALOG ? ? ? ? ? ? ?0
>>>>>>
>>>>>> NOMEDIA ? ? ? ? ? ? ? ? false
>>>>>>
>>>>>> LATE-NEG ? ? ? ? ? ? ? ?false
>>>>>>
>>>>>> PROXY-MEDIA ? ? ? ? ? ? false
>>>>>>
>>>>>> AGGRESSIVENAT ? ? ? ? ? false
>>>>>>
>>>>>> STUN-ENABLED ? ? ? ? ? ?true
>>>>>>
>>>>>> STUN-AUTO-DISABLE ? ? ? false
>>>>>>
>>>>>> CALLS-IN ? ? ? ? ? ? ? ?0
>>>>>>
>>>>>> FAILED-CALLS-IN ? ? ? ? 0
>>>>>>
>>>>>> CALLS-OUT ? ? ? ? ? ? ? 0
>>>>>>
>>>>>> FAILED-CALLS-OUT ? ? ? ?0
>>>>>>
>>>>>>
>>>>>> On Oct 27, 2010, at 10:04 AM, Anthony Minessale wrote:
>>>>>>
>>>>>> if you map it or not, a scanner would penetrate it.
>>>>>>
>>>>>> There are lot of sip scanners out there now, you just need to beware of
>>>>>> them.
>>>>>>
>>>>>>
>>>>>> On Wed, Oct 27, 2010 at 11:50 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>>
>>>>>> Thanks so much! I am sure many others will find this info invaluable. I will
>>>>>> try the static route again but have one question: When I started with FS I
>>>>>> found a "sip scanner" ?in FS and someone on this group said not to use port
>>>>>> mapping since it was a security risk. Is that true?
>>>>>>
>>>>>> On Oct 27, 2010, at 9:10 AM, Anthony Minessale wrote:
>>>>>>
>>>>>> you are completely guessing at things.
>>>>>>
>>>>>> I want you to understand that the only reason you are having problems
>>>>>>
>>>>>> with this is because you don't understand how it works enough to know
>>>>>>
>>>>>> what you are doing 100%
>>>>>>
>>>>>> Its a given that the pnp stuff is only for your dynamic IP.
>>>>>>
>>>>>> aggressive-nat-detection and sip-force-expires are all related to
>>>>>>
>>>>>> inbound calls when the things who are registering to you may be behind
>>>>>>
>>>>>> nat.
>>>>>>
>>>>>> You need to learn the difference between which nat tools are
>>>>>>
>>>>>> *) designed for your FS to run behind nat
>>>>>>
>>>>>> *) designed for FS to run public and accept connections from devices behind
>>>>>> nat.
>>>>>>
>>>>>> If you have a static IP, you don't need the pnp stuff so -nonat is fine
>>>>>>
>>>>>> What you need to do is set
>>>>>>
>>>>>> 1) set the params ext-sip-ip and ext-rtp-ip to your external static IP
>>>>>>
>>>>>> 2) map the sip ports and all of the rtp ports from your static IP to FS lan
>>>>>> addr
>>>>>>
>>>>>> 3) set sip-ip and rtp-ip to the lan addr you forwarded through.
>>>>>>
>>>>>>
>>>>>> If you don't do this: your outbound registration will use NAT to your
>>>>>>
>>>>>> provider and if there is no activity for the expire time on your NAT
>>>>>>
>>>>>> mapping the reverse port mapping from your provider back to you is
>>>>>>
>>>>>> lost. ?This is why you set your register expires to a very low number,
>>>>>>
>>>>>> (you need to make sure the provider does not turn the expires back up
>>>>>>
>>>>>> in the reply because it will beat your choice *see sip trace) if this
>>>>>>
>>>>>> is the case then you need the "ping" option set to 30, to continuously
>>>>>>
>>>>>> send an options to your provider.
>>>>>>
>>>>>> The static mapping is obviously the better, easier and more reliable
>>>>>> solution.
>>>>>>
>>>>>> So I want you to understand that the only way to keep a nat mapped
>>>>>>
>>>>>> port alive is to continuously send traffic, all the other methods that
>>>>>>
>>>>>> you are mentioning are to detect that phones registered to your are
>>>>>>
>>>>>> behind nat, I gave you that force-expires option before because your
>>>>>>
>>>>>> trace was full of inbound reg so I thought that is what you wanted
>>>>>>
>>>>>> help with.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Wed, Oct 27, 2010 at 10:43 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>>
>>>>>> I should mention that I did not have this problem with an SPA9000 PBX
>>>>>>
>>>>>> (asterisk based) for over two years so FS may be pickier about upnp and/or
>>>>>>
>>>>>> nat, or just better at it exposing a problem in the router.
>>>>>>
>>>>>> I made different changes to the gateways to test different things. One
>>>>>>
>>>>>> failed after 17 hours, the other two stayed up. ?What did not work:
>>>>>>
>>>>>> added <variable name="sip-force-expires" value="30"/> to the directory
>>>>>>
>>>>>> entries as suggested.
>>>>>>
>>>>>> set the gateway expire times to 30 seconds.
>>>>>>
>>>>>> What worked (could be coincidental) for the two gateways that stayed up:
>>>>>>
>>>>>> I Added <param name="aggressive-nat-detection" value="true"/>
>>>>>>
>>>>>> I originally setup FS to use the static ip by setting external sip/rtp to
>>>>>>
>>>>>> just the static ip (no autonat:) and ran with -nonat but I could not get
>>>>>>
>>>>>> incoming calls. The only way it worked was to use autonat:1.2.3.4. The
>>>>>>
>>>>>> router has 1 static public address and 1 dynamic external IP, this is the
>>>>>>
>>>>>> root of the problem, upnp only tells FS about the dynamic ip ?Will keep this
>>>>>>
>>>>>> thread up-to-date for anyone who may be in the same boat someday. Thanks
>>>>>>
>>>>>> again for looking at the trace.
>>>>>>
>>>>>> Mario
>>>>>>
>>>>>> You should be setting the req freq to a low number on the outbound gateways
>>>>>>
>>>>>> The examples you showed had a series of inbound reg
>>>>>>
>>>>>> also set expire-seconds to 30 in your gateway xml
>>>>>>
>>>>>>
>>>>>> The problem is if you are not constantly sending traffic to the box
>>>>>>
>>>>>> the nat mapping will go away.
>>>>>>
>>>>>> If you are in production you should be using a static ip with a static
>>>>>>
>>>>>> mapping, any trouble you are having is your own fault for playing with
>>>>>>
>>>>>> fire. ?The best we can do is tell you how to keep it contained.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>>
>>>>>> I made the change. I had no idea the settings for the inside phones effected
>>>>>>
>>>>>> nat for the outside sip accounts. I was looking into aggressive-nat-
>>>>>>
>>>>>> detection since the internal profile status always shows the right external
>>>>>>
>>>>>> static IP but the nat_ap status always shows the dynamic ip. Crossing
>>>>>>
>>>>>> fingers/etc since this problem is 85% of time (weeks!) into FS changeover.
>>>>>>
>>>>>> Thanks!
>>>>>>
>>>>>> Mario
>>>>>>
>>>>>> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
>>>>>>
>>>>>> add
>>>>>>
>>>>>> <variable name="sip-force-expires" value="30"/>
>>>>>>
>>>>>> to the <variables> section of your <user>
>>>>>>
>>>>>> you have it at 600 and the nat mapping is timing out while the 600
>>>>>>
>>>>>> seconds is ticking away
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>>
>>>>>> From the TSP:
>>>>>>
>>>>>> "I have enabled the SIP trace on your account. We are not currently seeing
>>>>>>
>>>>>> any registration attempts to your account within the last 15 minutes. Please
>>>>>>
>>>>>> restart FreeSwitch so that registration attempts begin again. Thank you. ".
>>>>>>
>>>>>> So FS is not getting past router.
>>>>>>
>>>>>> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>>>>>>
>>>>>> I ran the global trace during the problem and it is
>>>>>>
>>>>>> at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
>>>>>>
>>>>>> "accttwo", "acct3". The trace includes phones since it was global. I am
>>>>>>
>>>>>> using:
>>>>>>
>>>>>> ?<param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>>>>>>
>>>>>> ?<param name="ext-sip-ip" value="autonat:my-static.ip"/>
>>>>>>
>>>>>> I tried dumping nat and removing the autonat: above and using -nonat but
>>>>>>
>>>>>> that did not work, registration proceeded but no calls inbound.
>>>>>>
>>>>>> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>>>>>>
>>>>>> Whoops, I am using an IP address for at least one gateway so that is not the
>>>>>>
>>>>>> problem:
>>>>>>
>>>>>> They look outbound to me and I am using dns for 2 and an IP for one so that
>>>>>>
>>>>>> is not the issue. I was able to get FS to clear this up by doing "nat_map
>>>>>>
>>>>>> reinit" which is why I think this is a nat problem. I will do the trace you
>>>>>>
>>>>>> mentioned. I will plug an ip address into one of the gateways to see what
>>>>>>
>>>>>> happens, they all fail at once. Thanks for responding!
>>>>>>
>>>>>> Mario
>>>>>>
>>>>>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>>>>>>
>>>>>> I really need help on this as I have weeks into this problem. I thought I
>>>>>>
>>>>>> had it nailed but I guess not. After 5.5 hours I get:
>>>>>>
>>>>>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>>>>>>
>>>>>> Registration, setting retry to 15 seconds.
>>>>>>
>>>>>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>>>>
>>>>>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>>>>
>>>>>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
>>>>>>
>>>>>> Registration, setting retry to 30 seconds.
>>>>>>
>>>>>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>>>>>>
>>>>>> Registration, setting retry to 15 seconds.
>>>>>>
>>>>>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>>>>>>
>>>>>> Registration, setting retry to 15 seconds.
>>>>>>
>>>>>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>>>>
>>>>>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>>>>
>>>>>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>>>>>>
>>>>>> and no way to make/get calls until I restart FS. I did this:
>>>>>>
>>>>>> 1. log 7
>>>>>>
>>>>>> 2. sofia profile xxxx siptrace on ? for each profile/gateway
>>>>>>
>>>>>> 3. restarted router
>>>>>>
>>>>>> All three did not solve the problem. The trace and log produced no
>>>>>>
>>>>>> additional lines which is why I am wondering if FS has a problem since the
>>>>>>
>>>>>> trace shows no SIP activity.
>>>>>>
>>>>>> 3 gateways with 2 ITSPs
>>>>>>
>>>>>> 2 DSL/WAN lines, 1 static and 1 dynamic
>>>>>>
>>>>>> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
>>>>>>
>>>>>> external static ip.
>>>>>>
>>>>>> sofia status profile ... has the right ext ip
>>>>>>
>>>>>> nat_map status shows the dynamic (wrong) IP
>>>>>>
>>>>>> I tried starting with -nonat but that was worse
>>>>>>
>>>>>> the only way to fix is restart FS.
>>>>>>
>>>>>> I read the wiki on external nat, auto_nat and everything else many times.
>>>>>>
>>>>>> Thanks Mario
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>> _______________________________________________
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>
>>>>>> 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
>>>>>>
>>>>>> _______________________________________________
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>
>>>>>> 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
>>>>>>
>>>>>> _______________________________________________
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>
>>>>>> 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
>>>>>>
>>>>>> _______________________________________________
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>
>>>>>> 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
>>>>>>
>>>>>> _______________________________________________
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>
>>>>>> 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
>>>>>>
>>>>>> _______________________________________________
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 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
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> 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
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 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


From mario_fs at mgtech.com  Sat Oct 30 12:41:55 2010
From: mario_fs at mgtech.com (Mario G)
Date: Sat, 30 Oct 2010 12:41:55 -0700
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <3CAA3F8C-C2DC-48E2-89C5-9BC352C4B449@mgtech.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
	<407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>
	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
	<AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
	<6D9FD5E2-21FA-4630-8CF4-A87ADB30AC87@mgtech.com>
	<AANLkTi=BiDiXLgEoj_MfwmZbGjvXoOuNJgUYdetciNCj@mail.gmail.com>
	<9BAFDAF8-1E66-444C-A024-A88FF65D8BCA@mgtech.com>
	<3447E946-0BF8-4F7E-B5E9-E28F53DD0968@mgtech.com>
	<AANLkTi=LRbz6ADhLvBKkjGxOywAVLf--GTkzXEpUoDbi@mail.gmail.com>
	<BE754258-BE27-454F-B740-527A35DBDFFE@mgtech.com>
	<AANLkTi=3Kd-bCHJSED7VEh-cvtpdXMs_EkWZL=s8J4Ux@mail.gmail.com>
	<14205291-75B3-470F-ABA1-EE7F08A695E9@mgtech.com>
	<AANLkTi=D6siEAquSq6gYh-RnZeJJWrk11mHWK8De2tLh@mail.gmail.com>
	<2C35DBE8-13B3-4264-BB43-7496A1054AD1@mgtech.com>
	<86E8AE3F-E945-4952-A438-C674F23881D6@mgtech.com>
	<AANLkTin5ni_P-D5+XnCKugoRLgoExACV=eBWo3496AXq@mail.gmail.com>
	<317A03A2-FE67-4D0D-BBA9-0661079DC61E@mgtech.com>
	<AANLkTik62y_uSx42tTsb9W6wGOFZ=bd7AV2pBcb=cAjg@mail.gmail.com>
	<AF7D33D9-8198-45FB-8C5E-48121D90C3D2@mgtech.com>
	<3CAA3F8C-C2DC-48E2-89C5-9BC352C4B449@mgtech.com>
Message-ID: <A941E69D-61AC-4B59-B461-EFB8A98503EF@mgtech.com>

I am dead on the water without help so I greatly appreciate your help.

Finally got the loglevel 9 trace! Only 1 gateway (idtwo) defined and 1 phone hooked up to minimize data.
http://pastebin.freeswitch.org/14368    <- subset of trace showing registration working and then failing
http://pastebin.freeswitch.org/14365    <- this is full trace from startup minus thousands of duplicate lines
http://pastebin.freeswitch.org/14367    <- a tiny fraction of the thousands of duplicated lines removed from above, goes from 500 to 1 twice

Still, notice how there is no SIP trace activity once the registrations fail:
2010-10-30 09:30:15.046900 [WARNING] sofia_reg.c:387 idtwo Failed Registration, setting retry to 15 seconds.
nta: timer K fired, terminate NOTIFY (3892042)
outgoing_reclaim_all(0x0, 0x0, 0x102309cb0)
nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
nta: timer not set
2010-10-30 09:30:31.926382 [NOTICE] sofia_reg.c:342 Registering idtwo
nua: nua_register: entering
nua(0x1003c3750): sent signal r_register
nua(0x1003c3750): recv signal r_register
2010-10-30 09:30:47.685212 [WARNING] sofia_reg.c:387 idtwo Failed Registration, setting retry to 15 seconds.
2010-10-30 09:31:03.478941 [NOTICE] sofia_reg.c:342 Registering idtwo
nua: nua_register: entering
nua(0x1003c3750): sent signal r_register
nua(0x1003c3750): recv signal r_register
2010-10-30 09:31:19.219346 [WARNING] sofia_reg.c:387 idtwo Failed Registration, setting retry to 15 seconds.
2010-10-30 09:31:36.208448 [NOTICE] sofia_reg.c:342 Registering idtwo
nua: nua_register: entering
nua(0x1003c3750): sent signal r_register
nua(0x1003c3750): recv signal r_register

Mario

> 
>> 
>> On Oct 29, 2010, at 5:43 PM, Anthony Minessale wrote:
>> 
>>> also raise loglevel to debug
>>> console loglevel debug
>>> 
>>> 
>>> On Fri, Oct 29, 2010 at 7:29 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>> Will so, BTW, I mentioned below one is dns and another uses IP, I tested that theory using an IP  does not help. I also updated to todays git version and no help there. Will post when the trace is done.
>>>> 
>>>> On Oct 29, 2010, at 5:17 PM, Anthony Minessale wrote:
>>>> 
>>>>> can you repeat that trace with sofia debug on
>>>>> sofia loglevel all 9
>>>>> 
>>>>> Are you doing DNS by any chance in the gateway "proxy" param?
>>>>> you could try filling in the register-proxy param in your gateway to
>>>>> sip:<ip> <-- not dns but ip that dns resolves to
>>>>> <param name="register-proxy" value="sip:1.2.3.4"/>
>>>>> 
>>>>> I'm just guessing but Its possible some bad dns query could be throwing FS off.
>>>>> so this test would force all the packets to the exact IP of your host
>>>>> instead of looking it up.
>>>>> 
>>>>> 
>>>>> 
>>>>> On Fri, Oct 29, 2010 at 6:30 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>> Oh Boy: Not only is everything set to the static route but I turned
>>>>>> off/disconnected the dynamic DSL line so I only had 1 static line to the
>>>>>> router. The second wan is set in the router off. Even turned off router uPnP
>>>>>> even though I am using -nonat. Guess what.... I still have the problem. Look
>>>>>> like when FS says is going to retry it actually does not.
>>>>>> Here is a short one: http://pastebin.freeswitch.org/14359 - I caught one
>>>>>> right after I started FS, failure occurred in minutes (lucky). Look at the
>>>>>> bottom of the trace, you see SIP trace activity and then when it fails no
>>>>>> SIP trace activity. Could this possibly be a FS bug? (I am a mainframe
>>>>>> assembler systems programmer and I might think so if there were error retry
>>>>>> messages but nothing showing in one of my traces...)
>>>>>> Notes:
>>>>>> idone is gateway 1
>>>>>> idtwo is gateway 2  I had to trace both because it was impossible to figure
>>>>>> out which one would fail first. Ran several times but kept missing the right
>>>>>> one.
>>>>>> I use a url for one gateway and ip for another but it makes no difference
>>>>>> since both eventually fail.
>>>>>> 10. is local lan
>>>>>> 210. is external ip
>>>>>> 216. is itsp
>>>>>> 
>>>>>> Here is a longer one from earlier   http://pastebin.freeswitch.org/14357
>>>>>> Notes:
>>>>>> A call was received and hung up for idtwo - beginning of trace
>>>>>> 11 minutes later idtwo failed - see last line of trace
>>>>>> 
>>>>>> Thank you very much!
>>>>>> Mario
>>>>>> 
>>>>>> On Oct 29, 2010, at 12:11 PM, Anthony Minessale wrote:
>>>>>> 
>>>>>> stun-enabled must be true in your profile XML to see what you pasted.
>>>>>> 
>>>>>> Get me a sip trace of this from when it works until when it fails
>>>>>> 
>>>>>> only enable the sip trace on the profile with the gateway to reduce traffic
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> On Fri, Oct 29, 2010 at 1:56 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>> 
>>>>>> Oh my...  looks it was not NAT after all? Please help! I changed to all
>>>>>> profiles to static per instructions below and still have the problem:
>>>>>> 
>>>>>> 2010-10-29 11:15:18.536446 [NOTICE] sofia_reg.c:342 Registering uuid1
>>>>>> 
>>>>>> 2010-10-29 11:15:34.313150 [WARNING] sofia_reg.c:387 uuid1 Failed
>>>>>> Registration, setting retry to 15 seconds.
>>>>>> 
>>>>>> sofia global siptrace on did not show any activity for this gateway in or
>>>>>> out, others were fine but eventually fail. I setup static:
>>>>>> 
>>>>>> 1. set the params ext-sip-ip and ext-rtp-ip to my external static IP
>>>>>> 
>>>>>> 2. map the sip ports (5060-5080) and all of the rtp ports (16384-32767)  to
>>>>>> FS lan addr.
>>>>>> 
>>>>>> 3. set sip-ip and rtp-ip to the lan addr of FS
>>>>>> 
>>>>>> 4. start FS with -nonat
>>>>>> 
>>>>>> I don't know what to try next. BTW, the sofia status for the profiles shows
>>>>>> stun enabled but I did not set it up anywhere:
>>>>>> 
>>>>>> Name                    uuid1
>>>>>> 
>>>>>> Domain Name             N/A
>>>>>> 
>>>>>> Auto-NAT                false
>>>>>> 
>>>>>> DBName                  sofia_reg_idtwo
>>>>>> 
>>>>>> Pres Hosts
>>>>>> 
>>>>>> Dialplan                XML
>>>>>> 
>>>>>> Context                 public
>>>>>> 
>>>>>> Challenge Realm         auto_to
>>>>>> 
>>>>>> RTP-IP                  10.x.x.20
>>>>>> 
>>>>>> Ext-RTP-IP              210.x.x.100
>>>>>> 
>>>>>> SIP-IP                  10.x.x.20
>>>>>> 
>>>>>> Ext-SIP-IP              210.x.x.100
>>>>>> 
>>>>>> URL                     sip:mod_sofia at 210.x.x.100:5068
>>>>>> 
>>>>>> BIND-URL                sip:mod_sofia at 210.x.x.100:5068;maddr=10.x.x.20
>>>>>> 
>>>>>> HOLD-MUSIC              local_stream://moh
>>>>>> 
>>>>>> OUTBOUND-PROXY          N/A
>>>>>> 
>>>>>> CODECS IN               G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
>>>>>> 
>>>>>> CODECS OUT              PCMU,PCMA,GSM
>>>>>> 
>>>>>> TEL-EVENT               101
>>>>>> 
>>>>>> DTMF-MODE               rfc2833
>>>>>> 
>>>>>> CNG                     13
>>>>>> 
>>>>>> SESSION-TO              0
>>>>>> 
>>>>>> MAX-DIALOG              0
>>>>>> 
>>>>>> NOMEDIA                 false
>>>>>> 
>>>>>> LATE-NEG                false
>>>>>> 
>>>>>> PROXY-MEDIA             false
>>>>>> 
>>>>>> AGGRESSIVENAT           false
>>>>>> 
>>>>>> STUN-ENABLED            true
>>>>>> 
>>>>>> STUN-AUTO-DISABLE       false
>>>>>> 
>>>>>> CALLS-IN                0
>>>>>> 
>>>>>> FAILED-CALLS-IN         0
>>>>>> 
>>>>>> CALLS-OUT               0
>>>>>> 
>>>>>> FAILED-CALLS-OUT        0
>>>>>> 
>>>>>> 
>>>>>> On Oct 27, 2010, at 10:04 AM, Anthony Minessale wrote:
>>>>>> 
>>>>>> if you map it or not, a scanner would penetrate it.
>>>>>> 
>>>>>> There are lot of sip scanners out there now, you just need to beware of
>>>>>> them.
>>>>>> 
>>>>>> 
>>>>>> On Wed, Oct 27, 2010 at 11:50 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>> 
>>>>>> Thanks so much! I am sure many others will find this info invaluable. I will
>>>>>> try the static route again but have one question: When I started with FS I
>>>>>> found a "sip scanner"  in FS and someone on this group said not to use port
>>>>>> mapping since it was a security risk. Is that true?
>>>>>> 
>>>>>> On Oct 27, 2010, at 9:10 AM, Anthony Minessale wrote:
>>>>>> 
>>>>>> you are completely guessing at things.
>>>>>> 
>>>>>> I want you to understand that the only reason you are having problems
>>>>>> 
>>>>>> with this is because you don't understand how it works enough to know
>>>>>> 
>>>>>> what you are doing 100%
>>>>>> 
>>>>>> Its a given that the pnp stuff is only for your dynamic IP.
>>>>>> 
>>>>>> aggressive-nat-detection and sip-force-expires are all related to
>>>>>> 
>>>>>> inbound calls when the things who are registering to you may be behind
>>>>>> 
>>>>>> nat.
>>>>>> 
>>>>>> You need to learn the difference between which nat tools are
>>>>>> 
>>>>>> *) designed for your FS to run behind nat
>>>>>> 
>>>>>> *) designed for FS to run public and accept connections from devices behind
>>>>>> nat.
>>>>>> 
>>>>>> If you have a static IP, you don't need the pnp stuff so -nonat is fine
>>>>>> 
>>>>>> What you need to do is set
>>>>>> 
>>>>>> 1) set the params ext-sip-ip and ext-rtp-ip to your external static IP
>>>>>> 
>>>>>> 2) map the sip ports and all of the rtp ports from your static IP to FS lan
>>>>>> addr
>>>>>> 
>>>>>> 3) set sip-ip and rtp-ip to the lan addr you forwarded through.
>>>>>> 
>>>>>> 
>>>>>> If you don't do this: your outbound registration will use NAT to your
>>>>>> 
>>>>>> provider and if there is no activity for the expire time on your NAT
>>>>>> 
>>>>>> mapping the reverse port mapping from your provider back to you is
>>>>>> 
>>>>>> lost.  This is why you set your register expires to a very low number,
>>>>>> 
>>>>>> (you need to make sure the provider does not turn the expires back up
>>>>>> 
>>>>>> in the reply because it will beat your choice *see sip trace) if this
>>>>>> 
>>>>>> is the case then you need the "ping" option set to 30, to continuously
>>>>>> 
>>>>>> send an options to your provider.
>>>>>> 
>>>>>> The static mapping is obviously the better, easier and more reliable
>>>>>> solution.
>>>>>> 
>>>>>> So I want you to understand that the only way to keep a nat mapped
>>>>>> 
>>>>>> port alive is to continuously send traffic, all the other methods that
>>>>>> 
>>>>>> you are mentioning are to detect that phones registered to your are
>>>>>> 
>>>>>> behind nat, I gave you that force-expires option before because your
>>>>>> 
>>>>>> trace was full of inbound reg so I thought that is what you wanted
>>>>>> 
>>>>>> help with.
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> On Wed, Oct 27, 2010 at 10:43 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>> 
>>>>>> I should mention that I did not have this problem with an SPA9000 PBX
>>>>>> 
>>>>>> (asterisk based) for over two years so FS may be pickier about upnp and/or
>>>>>> 
>>>>>> nat, or just better at it exposing a problem in the router.
>>>>>> 
>>>>>> I made different changes to the gateways to test different things. One
>>>>>> 
>>>>>> failed after 17 hours, the other two stayed up.  What did not work:
>>>>>> 
>>>>>> added <variable name="sip-force-expires" value="30"/> to the directory
>>>>>> 
>>>>>> entries as suggested.
>>>>>> 
>>>>>> set the gateway expire times to 30 seconds.
>>>>>> 
>>>>>> What worked (could be coincidental) for the two gateways that stayed up:
>>>>>> 
>>>>>> I Added <param name="aggressive-nat-detection" value="true"/>
>>>>>> 
>>>>>> I originally setup FS to use the static ip by setting external sip/rtp to
>>>>>> 
>>>>>> just the static ip (no autonat:) and ran with -nonat but I could not get
>>>>>> 
>>>>>> incoming calls. The only way it worked was to use autonat:1.2.3.4. The
>>>>>> 
>>>>>> router has 1 static public address and 1 dynamic external IP, this is the
>>>>>> 
>>>>>> root of the problem, upnp only tells FS about the dynamic ip  Will keep this
>>>>>> 
>>>>>> thread up-to-date for anyone who may be in the same boat someday. Thanks
>>>>>> 
>>>>>> again for looking at the trace.
>>>>>> 
>>>>>> Mario
>>>>>> 
>>>>>> You should be setting the req freq to a low number on the outbound gateways
>>>>>> 
>>>>>> The examples you showed had a series of inbound reg
>>>>>> 
>>>>>> also set expire-seconds to 30 in your gateway xml
>>>>>> 
>>>>>> 
>>>>>> The problem is if you are not constantly sending traffic to the box
>>>>>> 
>>>>>> the nat mapping will go away.
>>>>>> 
>>>>>> If you are in production you should be using a static ip with a static
>>>>>> 
>>>>>> mapping, any trouble you are having is your own fault for playing with
>>>>>> 
>>>>>> fire.  The best we can do is tell you how to keep it contained.
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>> 
>>>>>> I made the change. I had no idea the settings for the inside phones effected
>>>>>> 
>>>>>> nat for the outside sip accounts. I was looking into aggressive-nat-
>>>>>> 
>>>>>> detection since the internal profile status always shows the right external
>>>>>> 
>>>>>> static IP but the nat_ap status always shows the dynamic ip. Crossing
>>>>>> 
>>>>>> fingers/etc since this problem is 85% of time (weeks!) into FS changeover.
>>>>>> 
>>>>>> Thanks!
>>>>>> 
>>>>>> Mario
>>>>>> 
>>>>>> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
>>>>>> 
>>>>>> add
>>>>>> 
>>>>>> <variable name="sip-force-expires" value="30"/>
>>>>>> 
>>>>>> to the <variables> section of your <user>
>>>>>> 
>>>>>> you have it at 600 and the nat mapping is timing out while the 600
>>>>>> 
>>>>>> seconds is ticking away
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>> 
>>>>>> From the TSP:
>>>>>> 
>>>>>> "I have enabled the SIP trace on your account. We are not currently seeing
>>>>>> 
>>>>>> any registration attempts to your account within the last 15 minutes. Please
>>>>>> 
>>>>>> restart FreeSwitch so that registration attempts begin again. Thank you. ".
>>>>>> 
>>>>>> So FS is not getting past router.
>>>>>> 
>>>>>> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>>>>>> 
>>>>>> I ran the global trace during the problem and it is
>>>>>> 
>>>>>> at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
>>>>>> 
>>>>>> "accttwo", "acct3". The trace includes phones since it was global. I am
>>>>>> 
>>>>>> using:
>>>>>> 
>>>>>> <param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>>>>>> 
>>>>>> <param name="ext-sip-ip" value="autonat:my-static.ip"/>
>>>>>> 
>>>>>> I tried dumping nat and removing the autonat: above and using -nonat but
>>>>>> 
>>>>>> that did not work, registration proceeded but no calls inbound.
>>>>>> 
>>>>>> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>>>>>> 
>>>>>> Whoops, I am using an IP address for at least one gateway so that is not the
>>>>>> 
>>>>>> problem:
>>>>>> 
>>>>>> They look outbound to me and I am using dns for 2 and an IP for one so that
>>>>>> 
>>>>>> is not the issue. I was able to get FS to clear this up by doing "nat_map
>>>>>> 
>>>>>> reinit" which is why I think this is a nat problem. I will do the trace you
>>>>>> 
>>>>>> mentioned. I will plug an ip address into one of the gateways to see what
>>>>>> 
>>>>>> happens, they all fail at once. Thanks for responding!
>>>>>> 
>>>>>> Mario
>>>>>> 
>>>>>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>>>>>> 
>>>>>> I really need help on this as I have weeks into this problem. I thought I
>>>>>> 
>>>>>> had it nailed but I guess not. After 5.5 hours I get:
>>>>>> 
>>>>>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>>>>>> 
>>>>>> Registration, setting retry to 15 seconds.
>>>>>> 
>>>>>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>>>> 
>>>>>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>>>> 
>>>>>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
>>>>>> 
>>>>>> Registration, setting retry to 30 seconds.
>>>>>> 
>>>>>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>>>>>> 
>>>>>> Registration, setting retry to 15 seconds.
>>>>>> 
>>>>>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>>>>>> 
>>>>>> Registration, setting retry to 15 seconds.
>>>>>> 
>>>>>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>>>> 
>>>>>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>>>> 
>>>>>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>>>>>> 
>>>>>> and no way to make/get calls until I restart FS. I did this:
>>>>>> 
>>>>>> 1. log 7
>>>>>> 
>>>>>> 2. sofia profile xxxx siptrace on   for each profile/gateway
>>>>>> 
>>>>>> 3. restarted router
>>>>>> 
>>>>>> All three did not solve the problem. The trace and log produced no
>>>>>> 
>>>>>> additional lines which is why I am wondering if FS has a problem since the
>>>>>> 
>>>>>> trace shows no SIP activity.
>>>>>> 
>>>>>> 3 gateways with 2 ITSPs
>>>>>> 
>>>>>> 2 DSL/WAN lines, 1 static and 1 dynamic
>>>>>> 
>>>>>> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
>>>>>> 
>>>>>> external static ip.
>>>>>> 
>>>>>> sofia status profile ... has the right ext ip
>>>>>> 
>>>>>> nat_map status shows the dynamic (wrong) IP
>>>>>> 
>>>>>> I tried starting with -nonat but that was worse
>>>>>> 
>>>>>> the only way to fix is restart FS.
>>>>>> 
>>>>>> I read the wiki on external nat, auto_nat and everything else many times.
>>>>>> 
>>>>>> Thanks Mario
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> 
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> 
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> 
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> 
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 
>>>>>> 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
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> 
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> 
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 
>>>>>> 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
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> 
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> 
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> 
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 
>>>>>> 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
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> 
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> 
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 
>>>>>> 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
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> 
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> 
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 
>>>>>> 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
>>>>>> 
>>>>>> _______________________________________________
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> 
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> 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
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> 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
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mario_fs at mgtech.com  Sat Oct 30 12:55:09 2010
From: mario_fs at mgtech.com (Mario G)
Date: Sat, 30 Oct 2010 12:55:09 -0700
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <AANLkTinPZQwmnqUYQWwwke1OW_GBKFmVWbeXFoGoJV3n@mail.gmail.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
	<407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>
	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
	<AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
	<6D9FD5E2-21FA-4630-8CF4-A87ADB30AC87@mgtech.com>
	<AANLkTi=BiDiXLgEoj_MfwmZbGjvXoOuNJgUYdetciNCj@mail.gmail.com>
	<9BAFDAF8-1E66-444C-A024-A88FF65D8BCA@mgtech.com>
	<3447E946-0BF8-4F7E-B5E9-E28F53DD0968@mgtech.com>
	<AANLkTi=LRbz6ADhLvBKkjGxOywAVLf--GTkzXEpUoDbi@mail.gmail.com>
	<BE754258-BE27-454F-B740-527A35DBDFFE@mgtech.com>
	<AANLkTi=3Kd-bCHJSED7VEh-cvtpdXMs_EkWZL=s8J4Ux@mail.gmail.com>
	<14205291-75B3-470F-ABA1-EE7F08A695E9@mgtech.com>
	<AANLkTi=D6siEAquSq6gYh-RnZeJJWrk11mHWK8De2tLh@mail.gmail.com>
	<2C35DBE8-13B3-4264-BB43-7496A1054AD1@mgtech.com>
	<86E8AE3F-E945-4952-A438-C674F23881D6@mgtech.com>
	<AANLkTin5ni_P-D5+XnCKugoRLgoExACV=eBWo3496AXq@mail.gmail.com>
	<317A03A2-FE67-4D0D-BBA9-0661079DC61E@mgtech.com>
	<AANLkTik62y_uSx42tTsb9W6wGOFZ=bd7AV2pBcb=cAjg@mail.gmail.com>
	<AF7D33D9-8198-45FB-8C5E-48121D90C3D2@mgtech.com>
	<3CAA3F8C-C2DC-48E2-89C5-9BC352C4B449@mgtech.com>
	<AANLkTinPZQwmnqUYQWwwke1OW_GBKFmVWbeXFoGoJV3n@mail.gmail.com>
Message-ID: <35B379BA-7259-4501-A68B-9D2715FBDC09@mgtech.com>

No I was wrong, the problem was Terminal not FS, it keeps it's log in memory. See my previous post with trace info. No memory problem now.

On Oct 30, 2010, at 12:34 PM, Anthony Minessale wrote:

> ok so there must be a problem in MAC, have you compared the results on
> win32 or linux?
> some other people are reporting mem issues on mac lets find it.
> 
> http://valgrind.org/downloads/valgrind-3.6.0.tar.bz2
> 
> build and install valgrind
> 
> run fs in vg mode
> valgrind --tool=memcheck --log-file=vg.log --leak-check=full
> --leak-resolution=high --show-reachable=yes /path/to/lt-freeswitch -vg
> 
> do something that eats mem
> 
> exit and get me the vg.log
> 
> 
> 
> On Sat, Oct 30, 2010 at 11:37 AM, Mario G <mario_fs at mgtech.com> wrote:
>> Setting sofia loglevel to 9 made Terminal app memory go from 20meg to eventually using all memory and locking the machine up. I spent all night trying to get a good trace. I need to get one when the failure occurs early before the lockup. Will be working all w/e on this. Thanks for you help.
>> 
>> I got this with only 1 gateway defined but did not catch it early. Notice how there is no SIP trace activity:
>> 2010-10-30 09:30:15.046900 [WARNING] sofia_reg.c:387 mvvyl Failed Registration, setting retry to 15 seconds.
>> nta: timer K fired, terminate NOTIFY (3892042)
>> outgoing_reclaim_all(0x0, 0x0, 0x102309cb0)
>> nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
>> nta: timer not set
>> 2010-10-30 09:30:31.926382 [NOTICE] sofia_reg.c:342 Registering mvvyl
>> nua: nua_register: entering
>> nua(0x1003c3750): sent signal r_register
>> nua(0x1003c3750): recv signal r_register
>> 2010-10-30 09:30:47.685212 [WARNING] sofia_reg.c:387 mvvyl Failed Registration, setting retry to 15 seconds.
>> 2010-10-30 09:31:03.478941 [NOTICE] sofia_reg.c:342 Registering mvvyl
>> nua: nua_register: entering
>> nua(0x1003c3750): sent signal r_register
>> nua(0x1003c3750): recv signal r_register
>> 2010-10-30 09:31:19.219346 [WARNING] sofia_reg.c:387 mvvyl Failed Registration, setting retry to 15 seconds.
>> 2010-10-30 09:31:36.208448 [NOTICE] sofia_reg.c:342 Registering mvvyl
>> nua: nua_register: entering
>> nua(0x1003c3750): sent signal r_register
>> nua(0x1003c3750): recv signal r_register
>> 
>> Mario
>> 
>>> 
>>> On Oct 29, 2010, at 5:43 PM, Anthony Minessale wrote:
>>> 
>>>> also raise loglevel to debug
>>>> console loglevel debug
>>>> 
>>>> 
>>>> On Fri, Oct 29, 2010 at 7:29 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>> Will so, BTW, I mentioned below one is dns and another uses IP, I tested that theory using an IP  does not help. I also updated to todays git version and no help there. Will post when the trace is done.
>>>>> 
>>>>> On Oct 29, 2010, at 5:17 PM, Anthony Minessale wrote:
>>>>> 
>>>>>> can you repeat that trace with sofia debug on
>>>>>> sofia loglevel all 9
>>>>>> 
>>>>>> Are you doing DNS by any chance in the gateway "proxy" param?
>>>>>> you could try filling in the register-proxy param in your gateway to
>>>>>> sip:<ip> <-- not dns but ip that dns resolves to
>>>>>> <param name="register-proxy" value="sip:1.2.3.4"/>
>>>>>> 
>>>>>> I'm just guessing but Its possible some bad dns query could be throwing FS off.
>>>>>> so this test would force all the packets to the exact IP of your host
>>>>>> instead of looking it up.
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> On Fri, Oct 29, 2010 at 6:30 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>>> Oh Boy: Not only is everything set to the static route but I turned
>>>>>>> off/disconnected the dynamic DSL line so I only had 1 static line to the
>>>>>>> router. The second wan is set in the router off. Even turned off router uPnP
>>>>>>> even though I am using -nonat. Guess what.... I still have the problem. Look
>>>>>>> like when FS says is going to retry it actually does not.
>>>>>>> Here is a short one: http://pastebin.freeswitch.org/14359 - I caught one
>>>>>>> right after I started FS, failure occurred in minutes (lucky). Look at the
>>>>>>> bottom of the trace, you see SIP trace activity and then when it fails no
>>>>>>> SIP trace activity. Could this possibly be a FS bug? (I am a mainframe
>>>>>>> assembler systems programmer and I might think so if there were error retry
>>>>>>> messages but nothing showing in one of my traces...)
>>>>>>> Notes:
>>>>>>> idone is gateway 1
>>>>>>> idtwo is gateway 2  I had to trace both because it was impossible to figure
>>>>>>> out which one would fail first. Ran several times but kept missing the right
>>>>>>> one.
>>>>>>> I use a url for one gateway and ip for another but it makes no difference
>>>>>>> since both eventually fail.
>>>>>>> 10. is local lan
>>>>>>> 210. is external ip
>>>>>>> 216. is itsp
>>>>>>> 
>>>>>>> Here is a longer one from earlier   http://pastebin.freeswitch.org/14357
>>>>>>> Notes:
>>>>>>> A call was received and hung up for idtwo - beginning of trace
>>>>>>> 11 minutes later idtwo failed - see last line of trace
>>>>>>> 
>>>>>>> Thank you very much!
>>>>>>> Mario
>>>>>>> 
>>>>>>> On Oct 29, 2010, at 12:11 PM, Anthony Minessale wrote:
>>>>>>> 
>>>>>>> stun-enabled must be true in your profile XML to see what you pasted.
>>>>>>> 
>>>>>>> Get me a sip trace of this from when it works until when it fails
>>>>>>> 
>>>>>>> only enable the sip trace on the profile with the gateway to reduce traffic
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> On Fri, Oct 29, 2010 at 1:56 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>>> 
>>>>>>> Oh my...  looks it was not NAT after all? Please help! I changed to all
>>>>>>> profiles to static per instructions below and still have the problem:
>>>>>>> 
>>>>>>> 2010-10-29 11:15:18.536446 [NOTICE] sofia_reg.c:342 Registering uuid1
>>>>>>> 
>>>>>>> 2010-10-29 11:15:34.313150 [WARNING] sofia_reg.c:387 uuid1 Failed
>>>>>>> Registration, setting retry to 15 seconds.
>>>>>>> 
>>>>>>> sofia global siptrace on did not show any activity for this gateway in or
>>>>>>> out, others were fine but eventually fail. I setup static:
>>>>>>> 
>>>>>>> 1. set the params ext-sip-ip and ext-rtp-ip to my external static IP
>>>>>>> 
>>>>>>> 2. map the sip ports (5060-5080) and all of the rtp ports (16384-32767)  to
>>>>>>> FS lan addr.
>>>>>>> 
>>>>>>> 3. set sip-ip and rtp-ip to the lan addr of FS
>>>>>>> 
>>>>>>> 4. start FS with -nonat
>>>>>>> 
>>>>>>> I don't know what to try next. BTW, the sofia status for the profiles shows
>>>>>>> stun enabled but I did not set it up anywhere:
>>>>>>> 
>>>>>>> Name                    uuid1
>>>>>>> 
>>>>>>> Domain Name             N/A
>>>>>>> 
>>>>>>> Auto-NAT                false
>>>>>>> 
>>>>>>> DBName                  sofia_reg_mvvyl
>>>>>>> 
>>>>>>> Pres Hosts
>>>>>>> 
>>>>>>> Dialplan                XML
>>>>>>> 
>>>>>>> Context                 public
>>>>>>> 
>>>>>>> Challenge Realm         auto_to
>>>>>>> 
>>>>>>> RTP-IP                  10.x.x.20
>>>>>>> 
>>>>>>> Ext-RTP-IP              210.x.x.100
>>>>>>> 
>>>>>>> SIP-IP                  10.x.x.20
>>>>>>> 
>>>>>>> Ext-SIP-IP              210.x.x.100
>>>>>>> 
>>>>>>> URL                     sip:mod_sofia at 210.x.x.100:5068
>>>>>>> 
>>>>>>> BIND-URL                sip:mod_sofia at 210.x.x.100:5068;maddr=10.x.x.20
>>>>>>> 
>>>>>>> HOLD-MUSIC              local_stream://moh
>>>>>>> 
>>>>>>> OUTBOUND-PROXY          N/A
>>>>>>> 
>>>>>>> CODECS IN               G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
>>>>>>> 
>>>>>>> CODECS OUT              PCMU,PCMA,GSM
>>>>>>> 
>>>>>>> TEL-EVENT               101
>>>>>>> 
>>>>>>> DTMF-MODE               rfc2833
>>>>>>> 
>>>>>>> CNG                     13
>>>>>>> 
>>>>>>> SESSION-TO              0
>>>>>>> 
>>>>>>> MAX-DIALOG              0
>>>>>>> 
>>>>>>> NOMEDIA                 false
>>>>>>> 
>>>>>>> LATE-NEG                false
>>>>>>> 
>>>>>>> PROXY-MEDIA             false
>>>>>>> 
>>>>>>> AGGRESSIVENAT           false
>>>>>>> 
>>>>>>> STUN-ENABLED            true
>>>>>>> 
>>>>>>> STUN-AUTO-DISABLE       false
>>>>>>> 
>>>>>>> CALLS-IN                0
>>>>>>> 
>>>>>>> FAILED-CALLS-IN         0
>>>>>>> 
>>>>>>> CALLS-OUT               0
>>>>>>> 
>>>>>>> FAILED-CALLS-OUT        0
>>>>>>> 
>>>>>>> 
>>>>>>> On Oct 27, 2010, at 10:04 AM, Anthony Minessale wrote:
>>>>>>> 
>>>>>>> if you map it or not, a scanner would penetrate it.
>>>>>>> 
>>>>>>> There are lot of sip scanners out there now, you just need to beware of
>>>>>>> them.
>>>>>>> 
>>>>>>> 
>>>>>>> On Wed, Oct 27, 2010 at 11:50 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>>> 
>>>>>>> Thanks so much! I am sure many others will find this info invaluable. I will
>>>>>>> try the static route again but have one question: When I started with FS I
>>>>>>> found a "sip scanner"  in FS and someone on this group said not to use port
>>>>>>> mapping since it was a security risk. Is that true?
>>>>>>> 
>>>>>>> On Oct 27, 2010, at 9:10 AM, Anthony Minessale wrote:
>>>>>>> 
>>>>>>> you are completely guessing at things.
>>>>>>> 
>>>>>>> I want you to understand that the only reason you are having problems
>>>>>>> 
>>>>>>> with this is because you don't understand how it works enough to know
>>>>>>> 
>>>>>>> what you are doing 100%
>>>>>>> 
>>>>>>> Its a given that the pnp stuff is only for your dynamic IP.
>>>>>>> 
>>>>>>> aggressive-nat-detection and sip-force-expires are all related to
>>>>>>> 
>>>>>>> inbound calls when the things who are registering to you may be behind
>>>>>>> 
>>>>>>> nat.
>>>>>>> 
>>>>>>> You need to learn the difference between which nat tools are
>>>>>>> 
>>>>>>> *) designed for your FS to run behind nat
>>>>>>> 
>>>>>>> *) designed for FS to run public and accept connections from devices behind
>>>>>>> nat.
>>>>>>> 
>>>>>>> If you have a static IP, you don't need the pnp stuff so -nonat is fine
>>>>>>> 
>>>>>>> What you need to do is set
>>>>>>> 
>>>>>>> 1) set the params ext-sip-ip and ext-rtp-ip to your external static IP
>>>>>>> 
>>>>>>> 2) map the sip ports and all of the rtp ports from your static IP to FS lan
>>>>>>> addr
>>>>>>> 
>>>>>>> 3) set sip-ip and rtp-ip to the lan addr you forwarded through.
>>>>>>> 
>>>>>>> 
>>>>>>> If you don't do this: your outbound registration will use NAT to your
>>>>>>> 
>>>>>>> provider and if there is no activity for the expire time on your NAT
>>>>>>> 
>>>>>>> mapping the reverse port mapping from your provider back to you is
>>>>>>> 
>>>>>>> lost.  This is why you set your register expires to a very low number,
>>>>>>> 
>>>>>>> (you need to make sure the provider does not turn the expires back up
>>>>>>> 
>>>>>>> in the reply because it will beat your choice *see sip trace) if this
>>>>>>> 
>>>>>>> is the case then you need the "ping" option set to 30, to continuously
>>>>>>> 
>>>>>>> send an options to your provider.
>>>>>>> 
>>>>>>> The static mapping is obviously the better, easier and more reliable
>>>>>>> solution.
>>>>>>> 
>>>>>>> So I want you to understand that the only way to keep a nat mapped
>>>>>>> 
>>>>>>> port alive is to continuously send traffic, all the other methods that
>>>>>>> 
>>>>>>> you are mentioning are to detect that phones registered to your are
>>>>>>> 
>>>>>>> behind nat, I gave you that force-expires option before because your
>>>>>>> 
>>>>>>> trace was full of inbound reg so I thought that is what you wanted
>>>>>>> 
>>>>>>> help with.
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> On Wed, Oct 27, 2010 at 10:43 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>>> 
>>>>>>> I should mention that I did not have this problem with an SPA9000 PBX
>>>>>>> 
>>>>>>> (asterisk based) for over two years so FS may be pickier about upnp and/or
>>>>>>> 
>>>>>>> nat, or just better at it exposing a problem in the router.
>>>>>>> 
>>>>>>> I made different changes to the gateways to test different things. One
>>>>>>> 
>>>>>>> failed after 17 hours, the other two stayed up.  What did not work:
>>>>>>> 
>>>>>>> added <variable name="sip-force-expires" value="30"/> to the directory
>>>>>>> 
>>>>>>> entries as suggested.
>>>>>>> 
>>>>>>> set the gateway expire times to 30 seconds.
>>>>>>> 
>>>>>>> What worked (could be coincidental) for the two gateways that stayed up:
>>>>>>> 
>>>>>>> I Added <param name="aggressive-nat-detection" value="true"/>
>>>>>>> 
>>>>>>> I originally setup FS to use the static ip by setting external sip/rtp to
>>>>>>> 
>>>>>>> just the static ip (no autonat:) and ran with -nonat but I could not get
>>>>>>> 
>>>>>>> incoming calls. The only way it worked was to use autonat:1.2.3.4. The
>>>>>>> 
>>>>>>> router has 1 static public address and 1 dynamic external IP, this is the
>>>>>>> 
>>>>>>> root of the problem, upnp only tells FS about the dynamic ip  Will keep this
>>>>>>> 
>>>>>>> thread up-to-date for anyone who may be in the same boat someday. Thanks
>>>>>>> 
>>>>>>> again for looking at the trace.
>>>>>>> 
>>>>>>> Mario
>>>>>>> 
>>>>>>> You should be setting the req freq to a low number on the outbound gateways
>>>>>>> 
>>>>>>> The examples you showed had a series of inbound reg
>>>>>>> 
>>>>>>> also set expire-seconds to 30 in your gateway xml
>>>>>>> 
>>>>>>> 
>>>>>>> The problem is if you are not constantly sending traffic to the box
>>>>>>> 
>>>>>>> the nat mapping will go away.
>>>>>>> 
>>>>>>> If you are in production you should be using a static ip with a static
>>>>>>> 
>>>>>>> mapping, any trouble you are having is your own fault for playing with
>>>>>>> 
>>>>>>> fire.  The best we can do is tell you how to keep it contained.
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>>> 
>>>>>>> I made the change. I had no idea the settings for the inside phones effected
>>>>>>> 
>>>>>>> nat for the outside sip accounts. I was looking into aggressive-nat-
>>>>>>> 
>>>>>>> detection since the internal profile status always shows the right external
>>>>>>> 
>>>>>>> static IP but the nat_ap status always shows the dynamic ip. Crossing
>>>>>>> 
>>>>>>> fingers/etc since this problem is 85% of time (weeks!) into FS changeover.
>>>>>>> 
>>>>>>> Thanks!
>>>>>>> 
>>>>>>> Mario
>>>>>>> 
>>>>>>> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
>>>>>>> 
>>>>>>> add
>>>>>>> 
>>>>>>> <variable name="sip-force-expires" value="30"/>
>>>>>>> 
>>>>>>> to the <variables> section of your <user>
>>>>>>> 
>>>>>>> you have it at 600 and the nat mapping is timing out while the 600
>>>>>>> 
>>>>>>> seconds is ticking away
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>>> 
>>>>>>> From the TSP:
>>>>>>> 
>>>>>>> "I have enabled the SIP trace on your account. We are not currently seeing
>>>>>>> 
>>>>>>> any registration attempts to your account within the last 15 minutes. Please
>>>>>>> 
>>>>>>> restart FreeSwitch so that registration attempts begin again. Thank you. ".
>>>>>>> 
>>>>>>> So FS is not getting past router.
>>>>>>> 
>>>>>>> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>>>>>>> 
>>>>>>> I ran the global trace during the problem and it is
>>>>>>> 
>>>>>>> at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
>>>>>>> 
>>>>>>> "accttwo", "acct3". The trace includes phones since it was global. I am
>>>>>>> 
>>>>>>> using:
>>>>>>> 
>>>>>>>  <param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>>>>>>> 
>>>>>>>  <param name="ext-sip-ip" value="autonat:my-static.ip"/>
>>>>>>> 
>>>>>>> I tried dumping nat and removing the autonat: above and using -nonat but
>>>>>>> 
>>>>>>> that did not work, registration proceeded but no calls inbound.
>>>>>>> 
>>>>>>> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>>>>>>> 
>>>>>>> Whoops, I am using an IP address for at least one gateway so that is not the
>>>>>>> 
>>>>>>> problem:
>>>>>>> 
>>>>>>> They look outbound to me and I am using dns for 2 and an IP for one so that
>>>>>>> 
>>>>>>> is not the issue. I was able to get FS to clear this up by doing "nat_map
>>>>>>> 
>>>>>>> reinit" which is why I think this is a nat problem. I will do the trace you
>>>>>>> 
>>>>>>> mentioned. I will plug an ip address into one of the gateways to see what
>>>>>>> 
>>>>>>> happens, they all fail at once. Thanks for responding!
>>>>>>> 
>>>>>>> Mario
>>>>>>> 
>>>>>>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>>>>>>> 
>>>>>>> I really need help on this as I have weeks into this problem. I thought I
>>>>>>> 
>>>>>>> had it nailed but I guess not. After 5.5 hours I get:
>>>>>>> 
>>>>>>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>>>>>>> 
>>>>>>> Registration, setting retry to 15 seconds.
>>>>>>> 
>>>>>>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>>>>> 
>>>>>>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>>>>> 
>>>>>>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
>>>>>>> 
>>>>>>> Registration, setting retry to 30 seconds.
>>>>>>> 
>>>>>>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>>>>>>> 
>>>>>>> Registration, setting retry to 15 seconds.
>>>>>>> 
>>>>>>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>>>>>>> 
>>>>>>> Registration, setting retry to 15 seconds.
>>>>>>> 
>>>>>>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
>>>>>>> 
>>>>>>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
>>>>>>> 
>>>>>>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
>>>>>>> 
>>>>>>> and no way to make/get calls until I restart FS. I did this:
>>>>>>> 
>>>>>>> 1. log 7
>>>>>>> 
>>>>>>> 2. sofia profile xxxx siptrace on   for each profile/gateway
>>>>>>> 
>>>>>>> 3. restarted router
>>>>>>> 
>>>>>>> All three did not solve the problem. The trace and log produced no
>>>>>>> 
>>>>>>> additional lines which is why I am wondering if FS has a problem since the
>>>>>>> 
>>>>>>> trace shows no SIP activity.
>>>>>>> 
>>>>>>> 3 gateways with 2 ITSPs
>>>>>>> 
>>>>>>> 2 DSL/WAN lines, 1 static and 1 dynamic
>>>>>>> 
>>>>>>> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
>>>>>>> 
>>>>>>> external static ip.
>>>>>>> 
>>>>>>> sofia status profile ... has the right ext ip
>>>>>>> 
>>>>>>> nat_map status shows the dynamic (wrong) IP
>>>>>>> 
>>>>>>> I tried starting with -nonat but that was worse
>>>>>>> 
>>>>>>> the only way to fix is restart FS.
>>>>>>> 
>>>>>>> I read the wiki on external nat, auto_nat and everything else many times.
>>>>>>> 
>>>>>>> Thanks Mario
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> 
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> 
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> 
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> 
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 
>>>>>>> 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
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> 
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> 
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 
>>>>>>> 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
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> 
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> 
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> 
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 
>>>>>>> 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
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> 
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> 
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 
>>>>>>> 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
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> 
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> 
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 
>>>>>>> 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
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> 
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 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
>>>>>> 
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> 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
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> 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
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mario_fs at mgtech.com  Sat Oct 30 13:31:28 2010
From: mario_fs at mgtech.com (Mario)
Date: Sat, 30 Oct 2010 13:31:28 -0700
Subject: [Freeswitch-users] registration fails after several hours - FS
 problem?
In-Reply-To: <35B379BA-7259-4501-A68B-9D2715FBDC09@mgtech.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>	<9BAFDAF8-1E66-444C-A024-A88FF65D8BCA@mgtech.com>	<3447E946-0BF8-4F7E-B5E9-E28F53DD0968@mgtech.com>	<AANLkTi=LRbz6ADhLvBKkjGxOywAVLf--GTkzXEpUoDbi@mail.gmail.com>	<BE754258-BE27-454F-B740-527A35DBDFFE@mgtech.com>	<AANLkTi=3Kd-bCHJSED7VEh-cvtpdXMs_EkWZL=s8J4Ux@mail.gmail.com>	<14205291-75B3-470F-ABA1-EE7F08A695E9@mgtech.com>	<AANLkTi=D6siEAquSq6gYh-RnZeJJWrk11mHWK8De2tLh@mail.gmail.com>	<2C35DBE8-13B3-4264-BB43-7496A1054AD1@mgtech.com>	<86E8AE3F-E945-4952-A438-C674F23881D6@mgtech.com>	<AANLkTin5ni_P-D5+XnCKugoRLgoExACV=eBWo3496AXq@mail.gmail.com>	<317A03A2-FE67-4D0D-BBA9-0661079DC61E@mgtech.com>	<AANLkTik62y_uSx42tTsb9W6wGOFZ=bd7AV2pBcb=cAjg@mail.gmail.com>	<AF7D33D9-8198-45FB-8C5E-48121D90C3D2@mgtech.com>	<3CAA3F8C-C2DC-48E2-89C5-9BC352C4B449@mgtech.com>	<AANLkTinPZQwmnqUYQWwwke1OW_GBKFmVWbeXFoGoJV3n@mail.gmail.com>
	<35B379BA-7259-4501-A68B-9D2715FBDC09@mgtech.com>
Message-ID: <4CCC80A0.4080401@mgtech.com>

I want to make sure this was clear: I have the problem sometimes after 5
minutes so memory is not an issue. Also, I checked the memory monitor.
The memory problem was only due to the loglevel 9 trace and the way
Terminal saves it. I did have this problem under Linux but at the time
we thought it was nat/upnp related. All static not and dynamic IP wan
interface is being kept inactive. I hope the traces help you see something.
Mario


From anthony.minessale at gmail.com  Sat Oct 30 17:01:18 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 30 Oct 2010 19:01:18 -0500
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <A941E69D-61AC-4B59-B461-EFB8A98503EF@mgtech.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
	<407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>
	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
	<AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
	<6D9FD5E2-21FA-4630-8CF4-A87ADB30AC87@mgtech.com>
	<AANLkTi=BiDiXLgEoj_MfwmZbGjvXoOuNJgUYdetciNCj@mail.gmail.com>
	<9BAFDAF8-1E66-444C-A024-A88FF65D8BCA@mgtech.com>
	<3447E946-0BF8-4F7E-B5E9-E28F53DD0968@mgtech.com>
	<AANLkTi=LRbz6ADhLvBKkjGxOywAVLf--GTkzXEpUoDbi@mail.gmail.com>
	<BE754258-BE27-454F-B740-527A35DBDFFE@mgtech.com>
	<AANLkTi=3Kd-bCHJSED7VEh-cvtpdXMs_EkWZL=s8J4Ux@mail.gmail.com>
	<14205291-75B3-470F-ABA1-EE7F08A695E9@mgtech.com>
	<AANLkTi=D6siEAquSq6gYh-RnZeJJWrk11mHWK8De2tLh@mail.gmail.com>
	<2C35DBE8-13B3-4264-BB43-7496A1054AD1@mgtech.com>
	<86E8AE3F-E945-4952-A438-C674F23881D6@mgtech.com>
	<AANLkTin5ni_P-D5+XnCKugoRLgoExACV=eBWo3496AXq@mail.gmail.com>
	<317A03A2-FE67-4D0D-BBA9-0661079DC61E@mgtech.com>
	<AANLkTik62y_uSx42tTsb9W6wGOFZ=bd7AV2pBcb=cAjg@mail.gmail.com>
	<AF7D33D9-8198-45FB-8C5E-48121D90C3D2@mgtech.com>
	<3CAA3F8C-C2DC-48E2-89C5-9BC352C4B449@mgtech.com>
	<A941E69D-61AC-4B59-B461-EFB8A98503EF@mgtech.com>
Message-ID: <AANLkTi=MwwajHTExYkqApTdoY1mD=i2ZqJ0kt7GRCtRr@mail.gmail.com>

Even sooner there is 1 reg sent out never answered or at least it seems.

Get a pcap at the same time from another win.
Also try to get a core dump of the running process.

I can help Monday if you can't do those things
On Oct 30, 2010 2:46 PM, "Mario G" <mario_fs at mgtech.com> wrote:
> I am dead on the water without help so I greatly appreciate your help.
>
> Finally got the loglevel 9 trace! Only 1 gateway (idtwo) defined and 1
phone hooked up to minimize data.
> http://pastebin.freeswitch.org/14368 <- subset of trace showing
registration working and then failing
> http://pastebin.freeswitch.org/14365 <- this is full trace from startup
minus thousands of duplicate lines
> http://pastebin.freeswitch.org/14367 <- a tiny fraction of the thousands
of duplicated lines removed from above, goes from 500 to 1 twice
>
> Still, notice how there is no SIP trace activity once the registrations
fail:
> 2010-10-30 09:30:15.046900 [WARNING] sofia_reg.c:387 idtwo Failed
Registration, setting retry to 15 seconds.
> nta: timer K fired, terminate NOTIFY (3892042)
> outgoing_reclaim_all(0x0, 0x0, 0x102309cb0)
> nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
> nta: timer not set
> 2010-10-30 09:30:31.926382 [NOTICE] sofia_reg.c:342 Registering idtwo
> nua: nua_register: entering
> nua(0x1003c3750): sent signal r_register
> nua(0x1003c3750): recv signal r_register
> 2010-10-30 09:30:47.685212 [WARNING] sofia_reg.c:387 idtwo Failed
Registration, setting retry to 15 seconds.
> 2010-10-30 09:31:03.478941 [NOTICE] sofia_reg.c:342 Registering idtwo
> nua: nua_register: entering
> nua(0x1003c3750): sent signal r_register
> nua(0x1003c3750): recv signal r_register
> 2010-10-30 09:31:19.219346 [WARNING] sofia_reg.c:387 idtwo Failed
Registration, setting retry to 15 seconds.
> 2010-10-30 09:31:36.208448 [NOTICE] sofia_reg.c:342 Registering idtwo
> nua: nua_register: entering
> nua(0x1003c3750): sent signal r_register
> nua(0x1003c3750): recv signal r_register
>
> Mario
>
>>
>>>
>>> On Oct 29, 2010, at 5:43 PM, Anthony Minessale wrote:
>>>
>>>> also raise loglevel to debug
>>>> console loglevel debug
>>>>
>>>>
>>>> On Fri, Oct 29, 2010 at 7:29 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>> Will so, BTW, I mentioned below one is dns and another uses IP, I
tested that theory using an IP does not help. I also updated to todays git
version and no help there. Will post when the trace is done.
>>>>>
>>>>> On Oct 29, 2010, at 5:17 PM, Anthony Minessale wrote:
>>>>>
>>>>>> can you repeat that trace with sofia debug on
>>>>>> sofia loglevel all 9
>>>>>>
>>>>>> Are you doing DNS by any chance in the gateway "proxy" param?
>>>>>> you could try filling in the register-proxy param in your gateway to
>>>>>> sip:<ip> <-- not dns but ip that dns resolves to
>>>>>> <param name="register-proxy" value="sip:1.2.3.4"/>
>>>>>>
>>>>>> I'm just guessing but Its possible some bad dns query could be
throwing FS off.
>>>>>> so this test would force all the packets to the exact IP of your host
>>>>>> instead of looking it up.
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Fri, Oct 29, 2010 at 6:30 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>>>>> Oh Boy: Not only is everything set to the static route but I turned
>>>>>>> off/disconnected the dynamic DSL line so I only had 1 static line to
the
>>>>>>> router. The second wan is set in the router off. Even turned off
router uPnP
>>>>>>> even though I am using -nonat. Guess what.... I still have the
problem. Look
>>>>>>> like when FS says is going to retry it actually does not.
>>>>>>> Here is a short one: http://pastebin.freeswitch.org/14359 - I caught
one
>>>>>>> right after I started FS, failure occurred in minutes (lucky). Look
at the
>>>>>>> bottom of the trace, you see SIP trace activity and then when it
fails no
>>>>>>> SIP trace activity. Could this possibly be a FS bug? (I am a
mainframe
>>>>>>> assembler systems programmer and I might think so if there were
error retry
>>>>>>> messages but nothing showing in one of my traces...)
>>>>>>> Notes:
>>>>>>> idone is gateway 1
>>>>>>> idtwo is gateway 2 I had to trace both because it was impossible to
figure
>>>>>>> out which one would fail first. Ran several times but kept missing
the right
>>>>>>> one.
>>>>>>> I use a url for one gateway and ip for another but it makes no
difference
>>>>>>> since both eventually fail.
>>>>>>> 10. is local lan
>>>>>>> 210. is external ip
>>>>>>> 216. is itsp
>>>>>>>
>>>>>>> Here is a longer one from earlier
http://pastebin.freeswitch.org/14357
>>>>>>> Notes:
>>>>>>> A call was received and hung up for idtwo - beginning of trace
>>>>>>> 11 minutes later idtwo failed - see last line of trace
>>>>>>>
>>>>>>> Thank you very much!
>>>>>>> Mario
>>>>>>>
>>>>>>> On Oct 29, 2010, at 12:11 PM, Anthony Minessale wrote:
>>>>>>>
>>>>>>> stun-enabled must be true in your profile XML to see what you
pasted.
>>>>>>>
>>>>>>> Get me a sip trace of this from when it works until when it fails
>>>>>>>
>>>>>>> only enable the sip trace on the profile with the gateway to reduce
traffic
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Fri, Oct 29, 2010 at 1:56 PM, Mario G <mario_fs at mgtech.com>
wrote:
>>>>>>>
>>>>>>> Oh my... looks it was not NAT after all? Please help! I changed to
all
>>>>>>> profiles to static per instructions below and still have the
problem:
>>>>>>>
>>>>>>> 2010-10-29 11:15:18.536446 [NOTICE] sofia_reg.c:342 Registering
uuid1
>>>>>>>
>>>>>>> 2010-10-29 11:15:34.313150 [WARNING] sofia_reg.c:387 uuid1 Failed
>>>>>>> Registration, setting retry to 15 seconds.
>>>>>>>
>>>>>>> sofia global siptrace on did not show any activity for this gateway
in or
>>>>>>> out, others were fine but eventually fail. I setup static:
>>>>>>>
>>>>>>> 1. set the params ext-sip-ip and ext-rtp-ip to my external static IP
>>>>>>>
>>>>>>> 2. map the sip ports (5060-5080) and all of the rtp ports
(16384-32767) to
>>>>>>> FS lan addr.
>>>>>>>
>>>>>>> 3. set sip-ip and rtp-ip to the lan addr of FS
>>>>>>>
>>>>>>> 4. start FS with -nonat
>>>>>>>
>>>>>>> I don't know what to try next. BTW, the sofia status for the
profiles shows
>>>>>>> stun enabled but I did not set it up anywhere:
>>>>>>>
>>>>>>> Name uuid1
>>>>>>>
>>>>>>> Domain Name N/A
>>>>>>>
>>>>>>> Auto-NAT false
>>>>>>>
>>>>>>> DBName sofia_reg_idtwo
>>>>>>>
>>>>>>> Pres Hosts
>>>>>>>
>>>>>>> Dialplan XML
>>>>>>>
>>>>>>> Context public
>>>>>>>
>>>>>>> Challenge Realm auto_to
>>>>>>>
>>>>>>> RTP-IP 10.x.x.20
>>>>>>>
>>>>>>> Ext-RTP-IP 210.x.x.100
>>>>>>>
>>>>>>> SIP-IP 10.x.x.20
>>>>>>>
>>>>>>> Ext-SIP-IP 210.x.x.100
>>>>>>>
>>>>>>> URL sip:mod_sofia at 210.x.x.100:5068
>>>>>>>
>>>>>>> BIND-URL sip:mod_sofia at 210.x.x.100:5068;maddr=10.x.x.20
>>>>>>>
>>>>>>> HOLD-MUSIC local_stream://moh
>>>>>>>
>>>>>>> OUTBOUND-PROXY N/A
>>>>>>>
>>>>>>> CODECS IN G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
>>>>>>>
>>>>>>> CODECS OUT PCMU,PCMA,GSM
>>>>>>>
>>>>>>> TEL-EVENT 101
>>>>>>>
>>>>>>> DTMF-MODE rfc2833
>>>>>>>
>>>>>>> CNG 13
>>>>>>>
>>>>>>> SESSION-TO 0
>>>>>>>
>>>>>>> MAX-DIALOG 0
>>>>>>>
>>>>>>> NOMEDIA false
>>>>>>>
>>>>>>> LATE-NEG false
>>>>>>>
>>>>>>> PROXY-MEDIA false
>>>>>>>
>>>>>>> AGGRESSIVENAT false
>>>>>>>
>>>>>>> STUN-ENABLED true
>>>>>>>
>>>>>>> STUN-AUTO-DISABLE false
>>>>>>>
>>>>>>> CALLS-IN 0
>>>>>>>
>>>>>>> FAILED-CALLS-IN 0
>>>>>>>
>>>>>>> CALLS-OUT 0
>>>>>>>
>>>>>>> FAILED-CALLS-OUT 0
>>>>>>>
>>>>>>>
>>>>>>> On Oct 27, 2010, at 10:04 AM, Anthony Minessale wrote:
>>>>>>>
>>>>>>> if you map it or not, a scanner would penetrate it.
>>>>>>>
>>>>>>> There are lot of sip scanners out there now, you just need to beware
of
>>>>>>> them.
>>>>>>>
>>>>>>>
>>>>>>> On Wed, Oct 27, 2010 at 11:50 AM, Mario G <mario_fs at mgtech.com>
wrote:
>>>>>>>
>>>>>>> Thanks so much! I am sure many others will find this info
invaluable. I will
>>>>>>> try the static route again but have one question: When I started
with FS I
>>>>>>> found a "sip scanner" in FS and someone on this group said not to
use port
>>>>>>> mapping since it was a security risk. Is that true?
>>>>>>>
>>>>>>> On Oct 27, 2010, at 9:10 AM, Anthony Minessale wrote:
>>>>>>>
>>>>>>> you are completely guessing at things.
>>>>>>>
>>>>>>> I want you to understand that the only reason you are having
problems
>>>>>>>
>>>>>>> with this is because you don't understand how it works enough to
know
>>>>>>>
>>>>>>> what you are doing 100%
>>>>>>>
>>>>>>> Its a given that the pnp stuff is only for your dynamic IP.
>>>>>>>
>>>>>>> aggressive-nat-detection and sip-force-expires are all related to
>>>>>>>
>>>>>>> inbound calls when the things who are registering to you may be
behind
>>>>>>>
>>>>>>> nat.
>>>>>>>
>>>>>>> You need to learn the difference between which nat tools are
>>>>>>>
>>>>>>> *) designed for your FS to run behind nat
>>>>>>>
>>>>>>> *) designed for FS to run public and accept connections from devices
behind
>>>>>>> nat.
>>>>>>>
>>>>>>> If you have a static IP, you don't need the pnp stuff so -nonat is
fine
>>>>>>>
>>>>>>> What you need to do is set
>>>>>>>
>>>>>>> 1) set the params ext-sip-ip and ext-rtp-ip to your external static
IP
>>>>>>>
>>>>>>> 2) map the sip ports and all of the rtp ports from your static IP to
FS lan
>>>>>>> addr
>>>>>>>
>>>>>>> 3) set sip-ip and rtp-ip to the lan addr you forwarded through.
>>>>>>>
>>>>>>>
>>>>>>> If you don't do this: your outbound registration will use NAT to
your
>>>>>>>
>>>>>>> provider and if there is no activity for the expire time on your NAT
>>>>>>>
>>>>>>> mapping the reverse port mapping from your provider back to you is
>>>>>>>
>>>>>>> lost. This is why you set your register expires to a very low
number,
>>>>>>>
>>>>>>> (you need to make sure the provider does not turn the expires back
up
>>>>>>>
>>>>>>> in the reply because it will beat your choice *see sip trace) if
this
>>>>>>>
>>>>>>> is the case then you need the "ping" option set to 30, to
continuously
>>>>>>>
>>>>>>> send an options to your provider.
>>>>>>>
>>>>>>> The static mapping is obviously the better, easier and more reliable
>>>>>>> solution.
>>>>>>>
>>>>>>> So I want you to understand that the only way to keep a nat mapped
>>>>>>>
>>>>>>> port alive is to continuously send traffic, all the other methods
that
>>>>>>>
>>>>>>> you are mentioning are to detect that phones registered to your are
>>>>>>>
>>>>>>> behind nat, I gave you that force-expires option before because your
>>>>>>>
>>>>>>> trace was full of inbound reg so I thought that is what you wanted
>>>>>>>
>>>>>>> help with.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Wed, Oct 27, 2010 at 10:43 AM, Mario G <mario_fs at mgtech.com>
wrote:
>>>>>>>
>>>>>>> I should mention that I did not have this problem with an SPA9000
PBX
>>>>>>>
>>>>>>> (asterisk based) for over two years so FS may be pickier about upnp
and/or
>>>>>>>
>>>>>>> nat, or just better at it exposing a problem in the router.
>>>>>>>
>>>>>>> I made different changes to the gateways to test different things.
One
>>>>>>>
>>>>>>> failed after 17 hours, the other two stayed up. What did not work:
>>>>>>>
>>>>>>> added <variable name="sip-force-expires" value="30"/> to the
directory
>>>>>>>
>>>>>>> entries as suggested.
>>>>>>>
>>>>>>> set the gateway expire times to 30 seconds.
>>>>>>>
>>>>>>> What worked (could be coincidental) for the two gateways that stayed
up:
>>>>>>>
>>>>>>> I Added <param name="aggressive-nat-detection" value="true"/>
>>>>>>>
>>>>>>> I originally setup FS to use the static ip by setting external
sip/rtp to
>>>>>>>
>>>>>>> just the static ip (no autonat:) and ran with -nonat but I could not
get
>>>>>>>
>>>>>>> incoming calls. The only way it worked was to use autonat:1.2.3.4.
The
>>>>>>>
>>>>>>> router has 1 static public address and 1 dynamic external IP, this
is the
>>>>>>>
>>>>>>> root of the problem, upnp only tells FS about the dynamic ip Will
keep this
>>>>>>>
>>>>>>> thread up-to-date for anyone who may be in the same boat someday.
Thanks
>>>>>>>
>>>>>>> again for looking at the trace.
>>>>>>>
>>>>>>> Mario
>>>>>>>
>>>>>>> You should be setting the req freq to a low number on the outbound
gateways
>>>>>>>
>>>>>>> The examples you showed had a series of inbound reg
>>>>>>>
>>>>>>> also set expire-seconds to 30 in your gateway xml
>>>>>>>
>>>>>>>
>>>>>>> The problem is if you are not constantly sending traffic to the box
>>>>>>>
>>>>>>> the nat mapping will go away.
>>>>>>>
>>>>>>> If you are in production you should be using a static ip with a
static
>>>>>>>
>>>>>>> mapping, any trouble you are having is your own fault for playing
with
>>>>>>>
>>>>>>> fire. The best we can do is tell you how to keep it contained.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com>
wrote:
>>>>>>>
>>>>>>> I made the change. I had no idea the settings for the inside phones
effected
>>>>>>>
>>>>>>> nat for the outside sip accounts. I was looking into aggressive-nat-
>>>>>>>
>>>>>>> detection since the internal profile status always shows the right
external
>>>>>>>
>>>>>>> static IP but the nat_ap status always shows the dynamic ip.
Crossing
>>>>>>>
>>>>>>> fingers/etc since this problem is 85% of time (weeks!) into FS
changeover.
>>>>>>>
>>>>>>> Thanks!
>>>>>>>
>>>>>>> Mario
>>>>>>>
>>>>>>> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
>>>>>>>
>>>>>>> add
>>>>>>>
>>>>>>> <variable name="sip-force-expires" value="30"/>
>>>>>>>
>>>>>>> to the <variables> section of your <user>
>>>>>>>
>>>>>>> you have it at 600 and the nat mapping is timing out while the 600
>>>>>>>
>>>>>>> seconds is ticking away
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com>
wrote:
>>>>>>>
>>>>>>> From the TSP:
>>>>>>>
>>>>>>> "I have enabled the SIP trace on your account. We are not currently
seeing
>>>>>>>
>>>>>>> any registration attempts to your account within the last 15
minutes. Please
>>>>>>>
>>>>>>> restart FreeSwitch so that registration attempts begin again. Thank
you. ".
>>>>>>>
>>>>>>> So FS is not getting past router.
>>>>>>>
>>>>>>> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>>>>>>>
>>>>>>> I ran the global trace during the problem and it is
>>>>>>>
>>>>>>> at http://pastebin.freeswitch.org/14324 . You can find "rnktel",
"acctone",
>>>>>>>
>>>>>>> "accttwo", "acct3". The trace includes phones since it was global. I
am
>>>>>>>
>>>>>>> using:
>>>>>>>
>>>>>>> <param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>>>>>>>
>>>>>>> <param name="ext-sip-ip" value="autonat:my-static.ip"/>
>>>>>>>
>>>>>>> I tried dumping nat and removing the autonat: above and using -nonat
but
>>>>>>>
>>>>>>> that did not work, registration proceeded but no calls inbound.
>>>>>>>
>>>>>>> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>>>>>>>
>>>>>>> Whoops, I am using an IP address for at least one gateway so that is
not the
>>>>>>>
>>>>>>> problem:
>>>>>>>
>>>>>>> They look outbound to me and I am using dns for 2 and an IP for one
so that
>>>>>>>
>>>>>>> is not the issue. I was able to get FS to clear this up by doing
"nat_map
>>>>>>>
>>>>>>> reinit" which is why I think this is a nat problem. I will do the
trace you
>>>>>>>
>>>>>>> mentioned. I will plug an ip address into one of the gateways to see
what
>>>>>>>
>>>>>>> happens, they all fail at once. Thanks for responding!
>>>>>>>
>>>>>>> Mario
>>>>>>>
>>>>>>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>>>>>>>
>>>>>>> I really need help on this as I have weeks into this problem. I
thought I
>>>>>>>
>>>>>>> had it nailed but I guess not. After 5.5 hours I get:
>>>>>>>
>>>>>>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>>>>>>>
>>>>>>> Registration, setting retry to 15 seconds.
>>>>>>>
>>>>>>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering
mvuuid
>>>>>>>
>>>>>>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering
mguuid
>>>>>>>
>>>>>>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
>>>>>>>
>>>>>>> Registration, setting retry to 30 seconds.
>>>>>>>
>>>>>>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>>>>>>>
>>>>>>> Registration, setting retry to 15 seconds.
>>>>>>>
>>>>>>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>>>>>>>
>>>>>>> Registration, setting retry to 15 seconds.
>>>>>>>
>>>>>>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering
mvuuid
>>>>>>>
>>>>>>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering
mguuid
>>>>>>>
>>>>>>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering
mynum777
>>>>>>>
>>>>>>> and no way to make/get calls until I restart FS. I did this:
>>>>>>>
>>>>>>> 1. log 7
>>>>>>>
>>>>>>> 2. sofia profile xxxx siptrace on for each profile/gateway
>>>>>>>
>>>>>>> 3. restarted router
>>>>>>>
>>>>>>> All three did not solve the problem. The trace and log produced no
>>>>>>>
>>>>>>> additional lines which is why I am wondering if FS has a problem
since the
>>>>>>>
>>>>>>> trace shows no SIP activity.
>>>>>>>
>>>>>>> 3 gateways with 2 ITSPs
>>>>>>>
>>>>>>> 2 DSL/WAN lines, 1 static and 1 dynamic
>>>>>>>
>>>>>>> I am using autonat:1.2.3.4 in internal and external profiles.
1.2.3.4 is the
>>>>>>>
>>>>>>> external static ip.
>>>>>>>
>>>>>>> sofia status profile ... has the right ext ip
>>>>>>>
>>>>>>> nat_map status shows the dynamic (wrong) IP
>>>>>>>
>>>>>>> I tried starting with -nonat but that was worse
>>>>>>>
>>>>>>> the only way to fix is restart FS.
>>>>>>>
>>>>>>> I read the wiki on external nat, auto_nat and everything else many
times.
>>>>>>>
>>>>>>> Thanks Mario
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>
>>>>>>> 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<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>>
>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>>
>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>
>>>>>>> FreeSWITCH Developer Conference
>>>>>>>
>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>>
>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>>
>>>>>>> pstn:+19193869900
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>
>>>>>>> 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<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>>
>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>>
>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>
>>>>>>> FreeSWITCH Developer Conference
>>>>>>>
>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>>
>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>>
>>>>>>> pstn:+19193869900
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>
>>>>>>> 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<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>>
>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>>
>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>
>>>>>>> FreeSWITCH Developer Conference
>>>>>>>
>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>>
>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>>
>>>>>>> pstn:+19193869900
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>
>>>>>>> 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<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>>
>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>>
>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>
>>>>>>> FreeSWITCH Developer Conference
>>>>>>>
>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>>
>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>>
>>>>>>> pstn:+19193869900
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>
>>>>>>> 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<MSN%3Aanthony_minessale at hotmail.com>
>>>>>>>
>>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>>>
>>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>>
>>>>>>> FreeSWITCH Developer Conference
>>>>>>>
>>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>>>
>>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>>>
>>>>>>> pstn:+19193869900
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> 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<MSN%3Aanthony_minessale at hotmail.com>
>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>>> IRC: irc.freenode.net #freeswitch
>>>>>>
>>>>>> FreeSWITCH Developer Conference
>>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>>> pstn:+19193869900
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> 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 <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From dujinfang at gmail.com  Sun Oct 31 01:54:58 2010
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 31 Oct 2010 16:54:58 +0800
Subject: [Freeswitch-users] ivr terminitor key?
Message-ID: <AANLkTi=7719tPQwSJU-=HAGRSvj5YYwcOcigwyd7CiWx@mail.gmail.com>

Hi,

For ivr menus that have digit-len > 1, is there a way to specify a
terminator key, say #, to stop digits collecting immediately? Default
behavior tends to wait until a timeout.

Thanks.

-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From dujinfang at gmail.com  Sun Oct 31 02:19:43 2010
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 31 Oct 2010 17:19:43 +0800
Subject: [Freeswitch-users] ivr terminitor key?
In-Reply-To: <AANLkTi=7719tPQwSJU-=HAGRSvj5YYwcOcigwyd7CiWx@mail.gmail.com>
References: <AANLkTi=7719tPQwSJU-=HAGRSvj5YYwcOcigwyd7CiWx@mail.gmail.com>
Message-ID: <AANLkTikFBEDnKBxq1f8tmWetV_4nvym6x9yi+V-Zy9Vy@mail.gmail.com>

confirm-key works that way if I enter some digits followed by #, but
not if I directly press #.

On Sun, Oct 31, 2010 at 4:54 PM, Seven Du <dujinfang at gmail.com> wrote:
> Hi,
>
> For ivr menus that have digit-len > 1, is there a way to specify a
> terminator key, say #, to stop digits collecting immediately? Default
> behavior tends to wait until a timeout.
>
> Thanks.
>
> --
> Blog: http://www.dujinfang.com
> Proj:? http://www.freeswitch.org.cn
>



-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From covici at ccs.covici.com  Sun Oct 31 10:04:23 2010
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sun, 31 Oct 2010 13:04:23 -0400
Subject: [Freeswitch-users] conference profile question
Message-ID: <16200.1288544663@ccs.covici.com>

Hi.  I have some conferences setup, not using the default profile.  Now
if the conference calls someone using the dial command, he seems to come
in with the default profile -- any variable or other thing I can set so
he comes in with the profile I called the conference originally with by
the conference app, or a better way to accomplish this?

thanks.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From mario_fs at mgtech.com  Sun Oct 31 12:24:35 2010
From: mario_fs at mgtech.com (Mario G)
Date: Sun, 31 Oct 2010 12:24:35 -0700
Subject: [Freeswitch-users] registration fails after several hours - FS
	problem?
In-Reply-To: <AANLkTi=MwwajHTExYkqApTdoY1mD=i2ZqJ0kt7GRCtRr@mail.gmail.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>
	<1BD4D198-91CC-491E-AEF4-15D716708B6B@mgtech.com>
	<407D83E5-1EBF-4B40-AB2F-E72099FB0B56@mgtech.com>
	<EA3C2BF7-F3AE-4058-B49E-35FB1F3E9C3C@mgtech.com>
	<AANLkTikpFMwUVzoSoV9E+B=FgX5+UqvGh1AzJjyCsNx=@mail.gmail.com>
	<6D9FD5E2-21FA-4630-8CF4-A87ADB30AC87@mgtech.com>
	<AANLkTi=BiDiXLgEoj_MfwmZbGjvXoOuNJgUYdetciNCj@mail.gmail.com>
	<9BAFDAF8-1E66-444C-A024-A88FF65D8BCA@mgtech.com>
	<3447E946-0BF8-4F7E-B5E9-E28F53DD0968@mgtech.com>
	<AANLkTi=LRbz6ADhLvBKkjGxOywAVLf--GTkzXEpUoDbi@mail.gmail.com>
	<BE754258-BE27-454F-B740-527A35DBDFFE@mgtech.com>
	<AANLkTi=3Kd-bCHJSED7VEh-cvtpdXMs_EkWZL=s8J4Ux@mail.gmail.com>
	<14205291-75B3-470F-ABA1-EE7F08A695E9@mgtech.com>
	<AANLkTi=D6siEAquSq6gYh-RnZeJJWrk11mHWK8De2tLh@mail.gmail.com>
	<2C35DBE8-13B3-4264-BB43-7496A1054AD1@mgtech.com>
	<86E8AE3F-E945-4952-A438-C674F23881D6@mgtech.com>
	<AANLkTin5ni_P-D5+XnCKugoRLgoExACV=eBWo3496AXq@mail.gmail.com>
	<317A03A2-FE67-4D0D-BBA9-0661079DC61E@mgtech.com>
	<AANLkTik62y_uSx42tTsb9W6wGOFZ=bd7AV2pBcb=cAjg@mail.gmail.com>
	<AF7D33D9-8198-45FB-8C5E-48121D90C3D2@mgtech.com>
	<3CAA3F8C-C2DC-48E2-89C5-9BC352C4B449@mgtech.com>
	<A941E69D-61AC-4B59-B461-EFB8A98503EF@mgtech.com>
	<AANLkTi=MwwajHTExYkqApTdoY1mD=i2ZqJ0kt7GRCtRr@mail.gmail.com>
Message-ID: <F1DAB05B-21DC-45B0-901B-0B3E6185BDBE@mgtech.com>

I have the pcap and dump to email to you and lot's of new info on this serious bug (yes it's a bug on FS for osX). The pcap is 1.1M and dump is 350M, please tell me where to send them. I don't want to put then in public areas since they contain security info. Please review my steps below. I don't know FS or Linux internals but it seems a lot like a timing issue where two processes are not communicating with each other since retry messages occur but there is no SIP tracing going on. THANKS SO MUCH!

LINUX
1. Setup FS on OpenSuse starting Sep 15. After basic initial problems there was a serious nat/upnp problems that lasted 3 weeks. Fixed with help, but still used nat.
2. Final testing was on git 2010-10-13. Ran fine for 5 days on very old 32 bit system.

OSX
3. Purchased Mac Mini and installed FS git 2010-10-23. Lasted only 3 to 17 hours. Problems looked same as nat so switched to full static.
4. With all static (-nonat) and only one DSL static connection active ITSPs go down in 5-60 minutes one by one. Still thought it was network related. Sent you traces.
5. Updated to git 10-29 but made no difference.

LINUX
6. Went back to the Linux box with git 10-13 using copy of config from mac. Pure static as osX. No problems for 6 hours!
7. Copied and updated Linux to git 10-29 to be the same as Mac box. Again, no problems for 12 hours!

OSX
8. Went back to the mac to provide you with pcap and dump. In about 15 minutes FS lost 2 ITSPs. Here are messages issues during pcap/dump, NOTE clock message which is first I have seen of it:

2010-10-31 11:35:00.593970 [WARNING] sofia_reg.c:387 idone Failed Registration, setting retry to 15 seconds.
2010-10-31 11:35:13.118634 [NOTICE] sofia_reg.c:342 Registering idtwo
2010-10-31 11:35:16.432236 [NOTICE] sofia_reg.c:342 Registering idone
2010-10-31 11:35:19.898319 [CRIT] switch_time.c:760 Forward Clock Skew Detected!
2010-10-31 11:35:25.440207 [WARNING] switch_scheduler.c:114 Task was executed late by 2 seconds 1 heartbeat (core)
2010-10-31 11:35:29.946329 [WARNING] sofia_reg.c:387 idtwo Failed Registration, setting retry to 15 seconds.
2010-10-31 11:35:32.147466 [WARNING] sofia_reg.c:387 idone Failed Registration, setting retry to 15 seconds.

I found the instruction for PCAP and TCPDUMP here in case you need them:
http://support.apple.com/kb/HT3994
http://www.osxbook.com/book/bonus/chapter8/core/

Note: I had the Mini set to no sleep even though it worked with Linux sleep. I found a couple others on the web who had the same problem and one had written a script to restart FS every 4 hours. Fried (tired) right now and cant find the URL but it was from Jan 2010.

One last thing to mention is that on osX using auto-nat:1.2.3.4 and some expiry parms, etc that may have triggered activity, FS worked much longer than on static. This is why I think it's timer or sync related and only on osX. 



On Oct 30, 2010, at 5:01 PM, Anthony Minessale wrote:

> Even sooner there is 1 reg sent out never answered or at least it seems.
> 
> Get a pcap at the same time from another win.
> Also try to get a core dump of the running process.
> 
> I can help Monday if you can't do those things
> 
> On Oct 30, 2010 2:46 PM, "Mario G" <mario_fs at mgtech.com> wrote:
> > I am dead on the water without help so I greatly appreciate your help.
> > 
> > Finally got the loglevel 9 trace! Only 1 gateway (idtwo) defined and 1 phone hooked up to minimize data.
> > http://pastebin.freeswitch.org/14368 <- subset of trace showing registration working and then failing
> > http://pastebin.freeswitch.org/14365 <- this is full trace from startup minus thousands of duplicate lines
> > http://pastebin.freeswitch.org/14367 <- a tiny fraction of the thousands of duplicated lines removed from above, goes from 500 to 1 twice
> > 
> > Still, notice how there is no SIP trace activity once the registrations fail:
> > 2010-10-30 09:30:15.046900 [WARNING] sofia_reg.c:387 idtwo Failed Registration, setting retry to 15 seconds.
> > nta: timer K fired, terminate NOTIFY (3892042)
> > outgoing_reclaim_all(0x0, 0x0, 0x102309cb0)
> > nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
> > nta: timer not set
> > 2010-10-30 09:30:31.926382 [NOTICE] sofia_reg.c:342 Registering idtwo
> > nua: nua_register: entering
> > nua(0x1003c3750): sent signal r_register
> > nua(0x1003c3750): recv signal r_register
> > 2010-10-30 09:30:47.685212 [WARNING] sofia_reg.c:387 idtwo Failed Registration, setting retry to 15 seconds.
> > 2010-10-30 09:31:03.478941 [NOTICE] sofia_reg.c:342 Registering idtwo
> > nua: nua_register: entering
> > nua(0x1003c3750): sent signal r_register
> > nua(0x1003c3750): recv signal r_register
> > 2010-10-30 09:31:19.219346 [WARNING] sofia_reg.c:387 idtwo Failed Registration, setting retry to 15 seconds.
> > 2010-10-30 09:31:36.208448 [NOTICE] sofia_reg.c:342 Registering idtwo
> > nua: nua_register: entering
> > nua(0x1003c3750): sent signal r_register
> > nua(0x1003c3750): recv signal r_register
> > 
> > Mario
> > 
> >> 
> >>> 
> >>> On Oct 29, 2010, at 5:43 PM, Anthony Minessale wrote:
> >>> 
> >>>> also raise loglevel to debug
> >>>> console loglevel debug
> >>>> 
> >>>> 
> >>>> On Fri, Oct 29, 2010 at 7:29 PM, Mario G <mario_fs at mgtech.com> wrote:
> >>>>> Will so, BTW, I mentioned below one is dns and another uses IP, I tested that theory using an IP does not help. I also updated to todays git version and no help there. Will post when the trace is done.
> >>>>> 
> >>>>> On Oct 29, 2010, at 5:17 PM, Anthony Minessale wrote:
> >>>>> 
> >>>>>> can you repeat that trace with sofia debug on
> >>>>>> sofia loglevel all 9
> >>>>>> 
> >>>>>> Are you doing DNS by any chance in the gateway "proxy" param?
> >>>>>> you could try filling in the register-proxy param in your gateway to
> >>>>>> sip:<ip> <-- not dns but ip that dns resolves to
> >>>>>> <param name="register-proxy" value="sip:1.2.3.4"/>
> >>>>>> 
> >>>>>> I'm just guessing but Its possible some bad dns query could be throwing FS off.
> >>>>>> so this test would force all the packets to the exact IP of your host
> >>>>>> instead of looking it up.
> >>>>>> 
> >>>>>> 
> >>>>>> 
> >>>>>> On Fri, Oct 29, 2010 at 6:30 PM, Mario G <mario_fs at mgtech.com> wrote:
> >>>>>>> Oh Boy: Not only is everything set to the static route but I turned
> >>>>>>> off/disconnected the dynamic DSL line so I only had 1 static line to the
> >>>>>>> router. The second wan is set in the router off. Even turned off router uPnP
> >>>>>>> even though I am using -nonat. Guess what.... I still have the problem. Look
> >>>>>>> like when FS says is going to retry it actually does not.
> >>>>>>> Here is a short one: http://pastebin.freeswitch.org/14359 - I caught one
> >>>>>>> right after I started FS, failure occurred in minutes (lucky). Look at the
> >>>>>>> bottom of the trace, you see SIP trace activity and then when it fails no
> >>>>>>> SIP trace activity. Could this possibly be a FS bug? (I am a mainframe
> >>>>>>> assembler systems programmer and I might think so if there were error retry
> >>>>>>> messages but nothing showing in one of my traces...)
> >>>>>>> Notes:
> >>>>>>> idone is gateway 1
> >>>>>>> idtwo is gateway 2 I had to trace both because it was impossible to figure
> >>>>>>> out which one would fail first. Ran several times but kept missing the right
> >>>>>>> one.
> >>>>>>> I use a url for one gateway and ip for another but it makes no difference
> >>>>>>> since both eventually fail.
> >>>>>>> 10. is local lan
> >>>>>>> 210. is external ip
> >>>>>>> 216. is itsp
> >>>>>>> 
> >>>>>>> Here is a longer one from earlier http://pastebin.freeswitch.org/14357
> >>>>>>> Notes:
> >>>>>>> A call was received and hung up for idtwo - beginning of trace
> >>>>>>> 11 minutes later idtwo failed - see last line of trace
> >>>>>>> 
> >>>>>>> Thank you very much!
> >>>>>>> Mario
> >>>>>>> 
> >>>>>>> On Oct 29, 2010, at 12:11 PM, Anthony Minessale wrote:
> >>>>>>> 
> >>>>>>> stun-enabled must be true in your profile XML to see what you pasted.
> >>>>>>> 
> >>>>>>> Get me a sip trace of this from when it works until when it fails
> >>>>>>> 
> >>>>>>> only enable the sip trace on the profile with the gateway to reduce traffic
> >>>>>>> 
> >>>>>>> 
> >>>>>>> 
> >>>>>>> On Fri, Oct 29, 2010 at 1:56 PM, Mario G <mario_fs at mgtech.com> wrote:
> >>>>>>> 
> >>>>>>> Oh my... looks it was not NAT after all? Please help! I changed to all
> >>>>>>> profiles to static per instructions below and still have the problem:
> >>>>>>> 
> >>>>>>> 2010-10-29 11:15:18.536446 [NOTICE] sofia_reg.c:342 Registering uuid1
> >>>>>>> 
> >>>>>>> 2010-10-29 11:15:34.313150 [WARNING] sofia_reg.c:387 uuid1 Failed
> >>>>>>> Registration, setting retry to 15 seconds.
> >>>>>>> 
> >>>>>>> sofia global siptrace on did not show any activity for this gateway in or
> >>>>>>> out, others were fine but eventually fail. I setup static:
> >>>>>>> 
> >>>>>>> 1. set the params ext-sip-ip and ext-rtp-ip to my external static IP
> >>>>>>> 
> >>>>>>> 2. map the sip ports (5060-5080) and all of the rtp ports (16384-32767) to
> >>>>>>> FS lan addr.
> >>>>>>> 
> >>>>>>> 3. set sip-ip and rtp-ip to the lan addr of FS
> >>>>>>> 
> >>>>>>> 4. start FS with -nonat
> >>>>>>> 
> >>>>>>> I don't know what to try next. BTW, the sofia status for the profiles shows
> >>>>>>> stun enabled but I did not set it up anywhere:
> >>>>>>> 
> >>>>>>> Name uuid1
> >>>>>>> 
> >>>>>>> Domain Name N/A
> >>>>>>> 
> >>>>>>> Auto-NAT false
> >>>>>>> 
> >>>>>>> DBName sofia_reg_idtwo
> >>>>>>> 
> >>>>>>> Pres Hosts
> >>>>>>> 
> >>>>>>> Dialplan XML
> >>>>>>> 
> >>>>>>> Context public
> >>>>>>> 
> >>>>>>> Challenge Realm auto_to
> >>>>>>> 
> >>>>>>> RTP-IP 10.x.x.20
> >>>>>>> 
> >>>>>>> Ext-RTP-IP 210.x.x.100
> >>>>>>> 
> >>>>>>> SIP-IP 10.x.x.20
> >>>>>>> 
> >>>>>>> Ext-SIP-IP 210.x.x.100
> >>>>>>> 
> >>>>>>> URL sip:mod_sofia at 210.x.x.100:5068
> >>>>>>> 
> >>>>>>> BIND-URL sip:mod_sofia at 210.x.x.100:5068;maddr=10.x.x.20
> >>>>>>> 
> >>>>>>> HOLD-MUSIC local_stream://moh
> >>>>>>> 
> >>>>>>> OUTBOUND-PROXY N/A
> >>>>>>> 
> >>>>>>> CODECS IN G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
> >>>>>>> 
> >>>>>>> CODECS OUT PCMU,PCMA,GSM
> >>>>>>> 
> >>>>>>> TEL-EVENT 101
> >>>>>>> 
> >>>>>>> DTMF-MODE rfc2833
> >>>>>>> 
> >>>>>>> CNG 13
> >>>>>>> 
> >>>>>>> SESSION-TO 0
> >>>>>>> 
> >>>>>>> MAX-DIALOG 0
> >>>>>>> 
> >>>>>>> NOMEDIA false
> >>>>>>> 
> >>>>>>> LATE-NEG false
> >>>>>>> 
> >>>>>>> PROXY-MEDIA false
> >>>>>>> 
> >>>>>>> AGGRESSIVENAT false
> >>>>>>> 
> >>>>>>> STUN-ENABLED true
> >>>>>>> 
> >>>>>>> STUN-AUTO-DISABLE false
> >>>>>>> 
> >>>>>>> CALLS-IN 0
> >>>>>>> 
> >>>>>>> FAILED-CALLS-IN 0
> >>>>>>> 
> >>>>>>> CALLS-OUT 0
> >>>>>>> 
> >>>>>>> FAILED-CALLS-OUT 0
> >>>>>>> 
> >>>>>>> 
> >>>>>>> On Oct 27, 2010, at 10:04 AM, Anthony Minessale wrote:
> >>>>>>> 
> >>>>>>> if you map it or not, a scanner would penetrate it.
> >>>>>>> 
> >>>>>>> There are lot of sip scanners out there now, you just need to beware of
> >>>>>>> them.
> >>>>>>> 
> >>>>>>> 
> >>>>>>> On Wed, Oct 27, 2010 at 11:50 AM, Mario G <mario_fs at mgtech.com> wrote:
> >>>>>>> 
> >>>>>>> Thanks so much! I am sure many others will find this info invaluable. I will
> >>>>>>> try the static route again but have one question: When I started with FS I
> >>>>>>> found a "sip scanner" in FS and someone on this group said not to use port
> >>>>>>> mapping since it was a security risk. Is that true?
> >>>>>>> 
> >>>>>>> On Oct 27, 2010, at 9:10 AM, Anthony Minessale wrote:
> >>>>>>> 
> >>>>>>> you are completely guessing at things.
> >>>>>>> 
> >>>>>>> I want you to understand that the only reason you are having problems
> >>>>>>> 
> >>>>>>> with this is because you don't understand how it works enough to know
> >>>>>>> 
> >>>>>>> what you are doing 100%
> >>>>>>> 
> >>>>>>> Its a given that the pnp stuff is only for your dynamic IP.
> >>>>>>> 
> >>>>>>> aggressive-nat-detection and sip-force-expires are all related to
> >>>>>>> 
> >>>>>>> inbound calls when the things who are registering to you may be behind
> >>>>>>> 
> >>>>>>> nat.
> >>>>>>> 
> >>>>>>> You need to learn the difference between which nat tools are
> >>>>>>> 
> >>>>>>> *) designed for your FS to run behind nat
> >>>>>>> 
> >>>>>>> *) designed for FS to run public and accept connections from devices behind
> >>>>>>> nat.
> >>>>>>> 
> >>>>>>> If you have a static IP, you don't need the pnp stuff so -nonat is fine
> >>>>>>> 
> >>>>>>> What you need to do is set
> >>>>>>> 
> >>>>>>> 1) set the params ext-sip-ip and ext-rtp-ip to your external static IP
> >>>>>>> 
> >>>>>>> 2) map the sip ports and all of the rtp ports from your static IP to FS lan
> >>>>>>> addr
> >>>>>>> 
> >>>>>>> 3) set sip-ip and rtp-ip to the lan addr you forwarded through.
> >>>>>>> 
> >>>>>>> 
> >>>>>>> If you don't do this: your outbound registration will use NAT to your
> >>>>>>> 
> >>>>>>> provider and if there is no activity for the expire time on your NAT
> >>>>>>> 
> >>>>>>> mapping the reverse port mapping from your provider back to you is
> >>>>>>> 
> >>>>>>> lost. This is why you set your register expires to a very low number,
> >>>>>>> 
> >>>>>>> (you need to make sure the provider does not turn the expires back up
> >>>>>>> 
> >>>>>>> in the reply because it will beat your choice *see sip trace) if this
> >>>>>>> 
> >>>>>>> is the case then you need the "ping" option set to 30, to continuously
> >>>>>>> 
> >>>>>>> send an options to your provider.
> >>>>>>> 
> >>>>>>> The static mapping is obviously the better, easier and more reliable
> >>>>>>> solution.
> >>>>>>> 
> >>>>>>> So I want you to understand that the only way to keep a nat mapped
> >>>>>>> 
> >>>>>>> port alive is to continuously send traffic, all the other methods that
> >>>>>>> 
> >>>>>>> you are mentioning are to detect that phones registered to your are
> >>>>>>> 
> >>>>>>> behind nat, I gave you that force-expires option before because your
> >>>>>>> 
> >>>>>>> trace was full of inbound reg so I thought that is what you wanted
> >>>>>>> 
> >>>>>>> help with.
> >>>>>>> 
> >>>>>>> 
> >>>>>>> 
> >>>>>>> 
> >>>>>>> 
> >>>>>>> 
> >>>>>>> 
> >>>>>>> 
> >>>>>>> On Wed, Oct 27, 2010 at 10:43 AM, Mario G <mario_fs at mgtech.com> wrote:
> >>>>>>> 
> >>>>>>> I should mention that I did not have this problem with an SPA9000 PBX
> >>>>>>> 
> >>>>>>> (asterisk based) for over two years so FS may be pickier about upnp and/or
> >>>>>>> 
> >>>>>>> nat, or just better at it exposing a problem in the router.
> >>>>>>> 
> >>>>>>> I made different changes to the gateways to test different things. One
> >>>>>>> 
> >>>>>>> failed after 17 hours, the other two stayed up. What did not work:
> >>>>>>> 
> >>>>>>> added <variable name="sip-force-expires" value="30"/> to the directory
> >>>>>>> 
> >>>>>>> entries as suggested.
> >>>>>>> 
> >>>>>>> set the gateway expire times to 30 seconds.
> >>>>>>> 
> >>>>>>> What worked (could be coincidental) for the two gateways that stayed up:
> >>>>>>> 
> >>>>>>> I Added <param name="aggressive-nat-detection" value="true"/>
> >>>>>>> 
> >>>>>>> I originally setup FS to use the static ip by setting external sip/rtp to
> >>>>>>> 
> >>>>>>> just the static ip (no autonat:) and ran with -nonat but I could not get
> >>>>>>> 
> >>>>>>> incoming calls. The only way it worked was to use autonat:1.2.3.4. The
> >>>>>>> 
> >>>>>>> router has 1 static public address and 1 dynamic external IP, this is the
> >>>>>>> 
> >>>>>>> root of the problem, upnp only tells FS about the dynamic ip Will keep this
> >>>>>>> 
> >>>>>>> thread up-to-date for anyone who may be in the same boat someday. Thanks
> >>>>>>> 
> >>>>>>> again for looking at the trace.
> >>>>>>> 
> >>>>>>> Mario
> >>>>>>> 
> >>>>>>> You should be setting the req freq to a low number on the outbound gateways
> >>>>>>> 
> >>>>>>> The examples you showed had a series of inbound reg
> >>>>>>> 
> >>>>>>> also set expire-seconds to 30 in your gateway xml
> >>>>>>> 
> >>>>>>> 
> >>>>>>> The problem is if you are not constantly sending traffic to the box
> >>>>>>> 
> >>>>>>> the nat mapping will go away.
> >>>>>>> 
> >>>>>>> If you are in production you should be using a static ip with a static
> >>>>>>> 
> >>>>>>> mapping, any trouble you are having is your own fault for playing with
> >>>>>>> 
> >>>>>>> fire. The best we can do is tell you how to keep it contained.
> >>>>>>> 
> >>>>>>> 
> >>>>>>> 
> >>>>>>> 
> >>>>>>> On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com> wrote:
> >>>>>>> 
> >>>>>>> I made the change. I had no idea the settings for the inside phones effected
> >>>>>>> 
> >>>>>>> nat for the outside sip accounts. I was looking into aggressive-nat-
> >>>>>>> 
> >>>>>>> detection since the internal profile status always shows the right external
> >>>>>>> 
> >>>>>>> static IP but the nat_ap status always shows the dynamic ip. Crossing
> >>>>>>> 
> >>>>>>> fingers/etc since this problem is 85% of time (weeks!) into FS changeover.
> >>>>>>> 
> >>>>>>> Thanks!
> >>>>>>> 
> >>>>>>> Mario
> >>>>>>> 
> >>>>>>> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
> >>>>>>> 
> >>>>>>> add
> >>>>>>> 
> >>>>>>> <variable name="sip-force-expires" value="30"/>
> >>>>>>> 
> >>>>>>> to the <variables> section of your <user>
> >>>>>>> 
> >>>>>>> you have it at 600 and the nat mapping is timing out while the 600
> >>>>>>> 
> >>>>>>> seconds is ticking away
> >>>>>>> 
> >>>>>>> 
> >>>>>>> 
> >>>>>>> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com> wrote:
> >>>>>>> 
> >>>>>>> From the TSP:
> >>>>>>> 
> >>>>>>> "I have enabled the SIP trace on your account. We are not currently seeing
> >>>>>>> 
> >>>>>>> any registration attempts to your account within the last 15 minutes. Please
> >>>>>>> 
> >>>>>>> restart FreeSwitch so that registration attempts begin again. Thank you. ".
> >>>>>>> 
> >>>>>>> So FS is not getting past router.
> >>>>>>> 
> >>>>>>> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
> >>>>>>> 
> >>>>>>> I ran the global trace during the problem and it is
> >>>>>>> 
> >>>>>>> at http://pastebin.freeswitch.org/14324 . You can find "rnktel", "acctone",
> >>>>>>> 
> >>>>>>> "accttwo", "acct3". The trace includes phones since it was global. I am
> >>>>>>> 
> >>>>>>> using:
> >>>>>>> 
> >>>>>>> <param name="ext-rtp-ip" value="autonat:my-static.ip"/>
> >>>>>>> 
> >>>>>>> <param name="ext-sip-ip" value="autonat:my-static.ip"/>
> >>>>>>> 
> >>>>>>> I tried dumping nat and removing the autonat: above and using -nonat but
> >>>>>>> 
> >>>>>>> that did not work, registration proceeded but no calls inbound.
> >>>>>>> 
> >>>>>>> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
> >>>>>>> 
> >>>>>>> Whoops, I am using an IP address for at least one gateway so that is not the
> >>>>>>> 
> >>>>>>> problem:
> >>>>>>> 
> >>>>>>> They look outbound to me and I am using dns for 2 and an IP for one so that
> >>>>>>> 
> >>>>>>> is not the issue. I was able to get FS to clear this up by doing "nat_map
> >>>>>>> 
> >>>>>>> reinit" which is why I think this is a nat problem. I will do the trace you
> >>>>>>> 
> >>>>>>> mentioned. I will plug an ip address into one of the gateways to see what
> >>>>>>> 
> >>>>>>> happens, they all fail at once. Thanks for responding!
> >>>>>>> 
> >>>>>>> Mario
> >>>>>>> 
> >>>>>>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
> >>>>>>> 
> >>>>>>> I really need help on this as I have weeks into this problem. I thought I
> >>>>>>> 
> >>>>>>> had it nailed but I guess not. After 5.5 hours I get:
> >>>>>>> 
> >>>>>>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
> >>>>>>> 
> >>>>>>> Registration, setting retry to 15 seconds.
> >>>>>>> 
> >>>>>>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342 Registering mvuuid
> >>>>>>> 
> >>>>>>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342 Registering mguuid
> >>>>>>> 
> >>>>>>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777 Failed
> >>>>>>> 
> >>>>>>> Registration, setting retry to 30 seconds.
> >>>>>>> 
> >>>>>>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
> >>>>>>> 
> >>>>>>> Registration, setting retry to 15 seconds.
> >>>>>>> 
> >>>>>>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
> >>>>>>> 
> >>>>>>> Registration, setting retry to 15 seconds.
> >>>>>>> 
> >>>>>>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342 Registering mvuuid
> >>>>>>> 
> >>>>>>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342 Registering mguuid
> >>>>>>> 
> >>>>>>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342 Registering mynum777
> >>>>>>> 
> >>>>>>> and no way to make/get calls until I restart FS. I did this:
> >>>>>>> 
> >>>>>>> 1. log 7
> >>>>>>> 
> >>>>>>> 2. sofia profile xxxx siptrace on for each profile/gateway
> >>>>>>> 
> >>>>>>> 3. restarted router
> >>>>>>> 
> >>>>>>> All three did not solve the problem. The trace and log produced no
> >>>>>>> 
> >>>>>>> additional lines which is why I am wondering if FS has a problem since the
> >>>>>>> 
> >>>>>>> trace shows no SIP activity.
> >>>>>>> 
> >>>>>>> 3 gateways with 2 ITSPs
> >>>>>>> 
> >>>>>>> 2 DSL/WAN lines, 1 static and 1 dynamic
> >>>>>>> 
> >>>>>>> I am using autonat:1.2.3.4 in internal and external profiles. 1.2.3.4 is the
> >>>>>>> 
> >>>>>>> external static ip.
> >>>>>>> 
> >>>>>>> sofia status profile ... has the right ext ip
> >>>>>>> 
> >>>>>>> nat_map status shows the dynamic (wrong) IP
> >>>>>>> 
> >>>>>>> I tried starting with -nonat but that was worse
> >>>>>>> 
> >>>>>>> the only way to fix is restart FS.
> >>>>>>> 
> >>>>>>> I read the wiki on external nat, auto_nat and everything else many times.
> >>>>>>> 
> >>>>>>> Thanks Mario
> >>>>>>> 
> >>>>>>> 
> >>>>>>> _______________________________________________
> >>>>>>> 
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> 
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> 
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> 
> >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> 
> >>>>>>> http://www.freeswitch.org
> >>>>>>> 
> >>>>>>> 
> >>>>>>> _______________________________________________
> >>>>>>> 
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> 
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> 
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> 
> >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> 
> >>>>>>> http://www.freeswitch.org
> >>>>>>> 
> >>>>>>> _______________________________________________
> >>>>>>> 
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> 
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> 
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> 
> >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> 
> >>>>>>> http://www.freeswitch.org
> >>>>>>> 
> >>>>>>> 
> >>>>>>> _______________________________________________
> >>>>>>> 
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> 
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> 
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> 
> >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> 
> >>>>>>> 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
> >>>>>>> 
> >>>>>>> _______________________________________________
> >>>>>>> 
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> 
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> 
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> 
> >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> 
> >>>>>>> http://www.freeswitch.org
> >>>>>>> 
> >>>>>>> 
> >>>>>>> _______________________________________________
> >>>>>>> 
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> 
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> 
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> 
> >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> 
> >>>>>>> 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
> >>>>>>> 
> >>>>>>> _______________________________________________
> >>>>>>> 
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> 
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> 
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> 
> >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> 
> >>>>>>> http://www.freeswitch.org
> >>>>>>> 
> >>>>>>> 
> >>>>>>> _______________________________________________
> >>>>>>> 
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> 
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> 
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> 
> >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> 
> >>>>>>> http://www.freeswitch.org
> >>>>>>> 
> >>>>>>> 
> >>>>>>> _______________________________________________
> >>>>>>> 
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> 
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> 
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> 
> >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> 
> >>>>>>> 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
> >>>>>>> 
> >>>>>>> _______________________________________________
> >>>>>>> 
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> 
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> 
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> 
> >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> 
> >>>>>>> http://www.freeswitch.org
> >>>>>>> 
> >>>>>>> 
> >>>>>>> _______________________________________________
> >>>>>>> 
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> 
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> 
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> 
> >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> 
> >>>>>>> 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
> >>>>>>> 
> >>>>>>> _______________________________________________
> >>>>>>> 
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> 
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> 
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> 
> >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> 
> >>>>>>> http://www.freeswitch.org
> >>>>>>> 
> >>>>>>> 
> >>>>>>> _______________________________________________
> >>>>>>> 
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> 
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> 
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> 
> >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> 
> >>>>>>> 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
> >>>>>>> 
> >>>>>>> _______________________________________________
> >>>>>>> 
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> 
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> 
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> 
> >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> 
> >>>>>>> http://www.freeswitch.org
> >>>>>>> 
> >>>>>>> 
> >>>>>>> _______________________________________________
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> http://www.freeswitch.org
> >>>>>>> 
> >>>>>>> 
> >>>>>>> _______________________________________________
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> 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
> >>>>>> 
> >>>>>> _______________________________________________
> >>>>>> FreeSWITCH-users mailing list
> >>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>> http://www.freeswitch.org
> >>>>> 
> >>>>> 
> >>>>> _______________________________________________
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> 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
> >>>> 
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>> 
> >>> 
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >> 
> >> 
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mario_fs at mgtech.com  Sun Oct 31 17:43:15 2010
From: mario_fs at mgtech.com (Mario)
Date: Sun, 31 Oct 2010 17:43:15 -0700
Subject: [Freeswitch-users] registration fails after several hours - FS
 problem?
In-Reply-To: <F1DAB05B-21DC-45B0-901B-0B3E6185BDBE@mgtech.com>
References: <02EF56A4-65D6-401B-864C-601DCF06AC72@mgtech.com>	<9BAFDAF8-1E66-444C-A024-A88FF65D8BCA@mgtech.com>	<3447E946-0BF8-4F7E-B5E9-E28F53DD0968@mgtech.com>	<AANLkTi=LRbz6ADhLvBKkjGxOywAVLf--GTkzXEpUoDbi@mail.gmail.com>	<BE754258-BE27-454F-B740-527A35DBDFFE@mgtech.com>	<AANLkTi=3Kd-bCHJSED7VEh-cvtpdXMs_EkWZL=s8J4Ux@mail.gmail.com>	<14205291-75B3-470F-ABA1-EE7F08A695E9@mgtech.com>	<AANLkTi=D6siEAquSq6gYh-RnZeJJWrk11mHWK8De2tLh@mail.gmail.com>	<2C35DBE8-13B3-4264-BB43-7496A1054AD1@mgtech.com>	<86E8AE3F-E945-4952-A438-C674F23881D6@mgtech.com>	<AANLkTin5ni_P-D5+XnCKugoRLgoExACV=eBWo3496AXq@mail.gmail.com>	<317A03A2-FE67-4D0D-BBA9-0661079DC61E@mgtech.com>	<AANLkTik62y_uSx42tTsb9W6wGOFZ=bd7AV2pBcb=cAjg@mail.gmail.com>	<AF7D33D9-8198-45FB-8C5E-48121D90C3D2@mgtech.com>	<3CAA3F8C-C2DC-48E2-89C5-9BC352C4B449@mgtech.com>	<A941E69D-61AC-4B59-B461-EFB8A98503EF@mgtech.com>	<AANLkTi=MwwajHTExYkqApTdoY1mD=i2ZqJ0kt7GRCtRr@mail.gmail.com>
	<F1DAB05B-21DC-45B0-901B-0B3E6185BDBE@mgtech.com>
Message-ID: <4CCE0D23.50800@mgtech.com>

UPDATED: I have the pcap and dump taken during the problem on our web
server. Please send me a private email so I can send credentials. I
don't want them public since they contain security info. THANKS SO MUCH!
Here is where we are:

Lot's of new info on this serious problem that looks like a FS bug or
incompatibility for osX since it runs fine on Linux (same config/ip).
It's acting like a timing issue where processes are not communicating
with each other since retry messages occur but there is no SIP tracing
going on. Please review the steps I took:

LINUX
1. Setup FS on OpenSuse starting Sep 15. After basic initial problems
there was a serious nat/upnp problems that lasted 3 weeks. Fixed with
help, but still used nat.
2. Final testing was on git 2010-10-13. Ran fine for 5 days on very old
32 bit system.

OSX
3. Purchased Mac Mini and installed FS git 2010-10-23. Lasted only 3 to
17 hours. Problems looked same as nat so switched to full static.
4. With all static (-nonat) and only one DSL static connection active
ITSPs go down in 5-60 minutes one by one. Still thought it was network
related. Sent you traces.
5. Updated to git 10-29 but made no difference.

LINUX
6. Went back to the Linux box with git 10-13 using exact copy of config
from mac, same IPs, etc. No problems for 6 hours!
7. Copied and updated Linux to git 10-29 to be the same as Mac box.
Again, no problems for 12 hours!

OSX
8. Went back to the mac to provide you with pcap and dump. In about 15
minutes FS lost 2 ITSPs. Here are messages issues during pcap/dump, NOTE
clock message which is first I have seen of it:

2010-10-31 11:35:00.593970 [WARNING] sofia_reg.c:387 idone Failed
Registration, setting retry to 15 seconds.
2010-10-31 11:35:13.118634 [NOTICE] sofia_reg.c:342 Registering idtwo
2010-10-31 11:35:16.432236 [NOTICE] sofia_reg.c:342 Registering idone
2010-10-31 11:35:19.898319 [CRIT] switch_time.c:760 Forward Clock Skew
Detected!
2010-10-31 11:35:25.440207 [WARNING] switch_scheduler.c:114 Task was
executed late by 2 seconds 1 heartbeat (core)
2010-10-31 11:35:29.946329 [WARNING] sofia_reg.c:387 idtwo Failed
Registration, setting retry to 15 seconds.
2010-10-31 11:35:32.147466 [WARNING] sofia_reg.c:387 idone Failed
Registration, setting retry to 15 seconds.

I found the instruction for PCAP and TCPDUMP here in case you need them:
http://support.apple.com/kb/HT3994
http://www.osxbook.com/book/bonus/chapter8/core/

Note: I set the Minito no sleep although it worked on Linux sleep. I
found a couple others on the web who had the same problem and one had
written a script to restart FS every 4 hours. Fried (tired) right now
and cant find the URL but it was from Jan 2010.

One last thing to mention is that on osX using auto-nat:1.2.3.4 and some
expiry parms, etc that may have triggered activity, FS worked much
longer than on static. This is why I think it's timer or sync related
and only on osX.


> 
> 
> On Oct 30, 2010, at 5:01 PM, Anthony Minessale wrote:
> 
>> Even sooner there is 1 reg sent out never answered or at least it seems.
>>
>> Get a pcap at the same time from another win.
>> Also try to get a core dump of the running process.
>>
>> I can help Monday if you can't do those things
>>
>> On Oct 30, 2010 2:46 PM, "Mario G" <mario_fs at mgtech.com
>> <mailto:mario_fs at mgtech.com>> wrote:
>> > I am dead on the water without help so I greatly appreciate your help.
>> >
>> > Finally got the loglevel 9 trace! Only 1 gateway (idtwo) defined and
>> 1 phone hooked up to minimize data.
>> > http://pastebin.freeswitch.org/14368 <- subset of trace showing
>> registration working and then failing
>> > http://pastebin.freeswitch.org/14365 <- this is full trace from
>> startup minus thousands of duplicate lines
>> > http://pastebin.freeswitch.org/14367 <- a tiny fraction of the
>> thousands of duplicated lines removed from above, goes from 500 to 1 twice
>> >
>> > Still, notice how there is no SIP trace activity once the
>> registrations fail:
>> > 2010-10-30 09:30:15.046900 [WARNING] sofia_reg.c:387 idtwo Failed
>> Registration, setting retry to 15 seconds.
>> > nta: timer K fired, terminate NOTIFY (3892042)
>> > outgoing_reclaim_all(0x0, 0x0, 0x102309cb0)
>> > nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
>> > nta: timer not set
>> > 2010-10-30 09:30:31.926382 [NOTICE] sofia_reg.c:342 Registering idtwo
>> > nua: nua_register: entering
>> > nua(0x1003c3750): sent signal r_register
>> > nua(0x1003c3750): recv signal r_register
>> > 2010-10-30 09:30:47.685212 [WARNING] sofia_reg.c:387 idtwo Failed
>> Registration, setting retry to 15 seconds.
>> > 2010-10-30 09:31:03.478941 [NOTICE] sofia_reg.c:342 Registering idtwo
>> > nua: nua_register: entering
>> > nua(0x1003c3750): sent signal r_register
>> > nua(0x1003c3750): recv signal r_register
>> > 2010-10-30 09:31:19.219346 [WARNING] sofia_reg.c:387 idtwo Failed
>> Registration, setting retry to 15 seconds.
>> > 2010-10-30 09:31:36.208448 [NOTICE] sofia_reg.c:342 Registering idtwo
>> > nua: nua_register: entering
>> > nua(0x1003c3750): sent signal r_register
>> > nua(0x1003c3750): recv signal r_register
>> >
>> > Mario
>> >
>> >>
>> >>>
>> >>> On Oct 29, 2010, at 5:43 PM, Anthony Minessale wrote:
>> >>>
>> >>>> also raise loglevel to debug
>> >>>> console loglevel debug
>> >>>>
>> >>>>
>> >>>> On Fri, Oct 29, 2010 at 7:29 PM, Mario G <mario_fs at mgtech.com
>> <mailto:mario_fs at mgtech.com>> wrote:
>> >>>>> Will so, BTW, I mentioned below one is dns and another uses IP,
>> I tested that theory using an IP does not help. I also updated to
>> todays git version and no help there. Will post when the trace is done.
>> >>>>>
>> >>>>> On Oct 29, 2010, at 5:17 PM, Anthony Minessale wrote:
>> >>>>>
>> >>>>>> can you repeat that trace with sofia debug on
>> >>>>>> sofia loglevel all 9
>> >>>>>>
>> >>>>>> Are you doing DNS by any chance in the gateway "proxy" param?
>> >>>>>> you could try filling in the register-proxy param in your
>> gateway to
>> >>>>>> sip:<ip> <-- not dns but ip that dns resolves to
>> >>>>>> <param name="register-proxy" value="sip:1.2.3.4"/>
>> >>>>>>
>> >>>>>> I'm just guessing but Its possible some bad dns query could be
>> throwing FS off.
>> >>>>>> so this test would force all the packets to the exact IP of
>> your host
>> >>>>>> instead of looking it up.
>> >>>>>>
>> >>>>>>
>> >>>>>>
>> >>>>>> On Fri, Oct 29, 2010 at 6:30 PM, Mario G <mario_fs at mgtech.com
>> <mailto:mario_fs at mgtech.com>> wrote:
>> >>>>>>> Oh Boy: Not only is everything set to the static route but I
>> turned
>> >>>>>>> off/disconnected the dynamic DSL line so I only had 1 static
>> line to the
>> >>>>>>> router. The second wan is set in the router off. Even turned
>> off router uPnP
>> >>>>>>> even though I am using -nonat. Guess what.... I still have the
>> problem. Look
>> >>>>>>> like when FS says is going to retry it actually does not.
>> >>>>>>> Here is a short one: http://pastebin.freeswitch.org/14359 - I
>> caught one
>> >>>>>>> right after I started FS, failure occurred in minutes (lucky).
>> Look at the
>> >>>>>>> bottom of the trace, you see SIP trace activity and then when
>> it fails no
>> >>>>>>> SIP trace activity. Could this possibly be a FS bug? (I am a
>> mainframe
>> >>>>>>> assembler systems programmer and I might think so if there
>> were error retry
>> >>>>>>> messages but nothing showing in one of my traces...)
>> >>>>>>> Notes:
>> >>>>>>> idone is gateway 1
>> >>>>>>> idtwo is gateway 2 I had to trace both because it was
>> impossible to figure
>> >>>>>>> out which one would fail first. Ran several times but kept
>> missing the right
>> >>>>>>> one.
>> >>>>>>> I use a url for one gateway and ip for another but it makes no
>> difference
>> >>>>>>> since both eventually fail.
>> >>>>>>> 10. is local lan
>> >>>>>>> 210. is external ip
>> >>>>>>> 216. is itsp
>> >>>>>>>
>> >>>>>>> Here is a longer one from earlier
>> http://pastebin.freeswitch.org/14357
>> >>>>>>> Notes:
>> >>>>>>> A call was received and hung up for idtwo - beginning of trace
>> >>>>>>> 11 minutes later idtwo failed - see last line of trace
>> >>>>>>>
>> >>>>>>> Thank you very much!
>> >>>>>>> Mario
>> >>>>>>>
>> >>>>>>> On Oct 29, 2010, at 12:11 PM, Anthony Minessale wrote:
>> >>>>>>>
>> >>>>>>> stun-enabled must be true in your profile XML to see what you
>> pasted.
>> >>>>>>>
>> >>>>>>> Get me a sip trace of this from when it works until when it fails
>> >>>>>>>
>> >>>>>>> only enable the sip trace on the profile with the gateway to
>> reduce traffic
>> >>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>>> On Fri, Oct 29, 2010 at 1:56 PM, Mario G <mario_fs at mgtech.com
>> <mailto:mario_fs at mgtech.com>> wrote:
>> >>>>>>>
>> >>>>>>> Oh my... looks it was not NAT after all? Please help! I
>> changed to all
>> >>>>>>> profiles to static per instructions below and still have the
>> problem:
>> >>>>>>>
>> >>>>>>> 2010-10-29 11:15:18.536446 [NOTICE] sofia_reg.c:342
>> Registering uuid1
>> >>>>>>>
>> >>>>>>> 2010-10-29 11:15:34.313150 [WARNING] sofia_reg.c:387 uuid1 Failed
>> >>>>>>> Registration, setting retry to 15 seconds.
>> >>>>>>>
>> >>>>>>> sofia global siptrace on did not show any activity for this
>> gateway in or
>> >>>>>>> out, others were fine but eventually fail. I setup static:
>> >>>>>>>
>> >>>>>>> 1. set the params ext-sip-ip and ext-rtp-ip to my external
>> static IP
>> >>>>>>>
>> >>>>>>> 2. map the sip ports (5060-5080) and all of the rtp ports
>> (16384-32767) to
>> >>>>>>> FS lan addr.
>> >>>>>>>
>> >>>>>>> 3. set sip-ip and rtp-ip to the lan addr of FS
>> >>>>>>>
>> >>>>>>> 4. start FS with -nonat
>> >>>>>>>
>> >>>>>>> I don't know what to try next. BTW, the sofia status for the
>> profiles shows
>> >>>>>>> stun enabled but I did not set it up anywhere:
>> >>>>>>>
>> >>>>>>> Name uuid1
>> >>>>>>>
>> >>>>>>> Domain Name N/A
>> >>>>>>>
>> >>>>>>> Auto-NAT false
>> >>>>>>>
>> >>>>>>> DBName sofia_reg_idtwo
>> >>>>>>>
>> >>>>>>> Pres Hosts
>> >>>>>>>
>> >>>>>>> Dialplan XML
>> >>>>>>>
>> >>>>>>> Context public
>> >>>>>>>
>> >>>>>>> Challenge Realm auto_to
>> >>>>>>>
>> >>>>>>> RTP-IP 10.x.x.20
>> >>>>>>>
>> >>>>>>> Ext-RTP-IP 210.x.x.100
>> >>>>>>>
>> >>>>>>> SIP-IP 10.x.x.20
>> >>>>>>>
>> >>>>>>> Ext-SIP-IP 210.x.x.100
>> >>>>>>>
>> >>>>>>> URL sip:mod_sofia at 210.x.x.100:5068
>> >>>>>>>
>> >>>>>>> BIND-URL sip:mod_sofia at 210.x.x.100:5068;maddr=10.x.x.20
>> >>>>>>>
>> >>>>>>> HOLD-MUSIC local_stream://moh
>> >>>>>>>
>> >>>>>>> OUTBOUND-PROXY N/A
>> >>>>>>>
>> >>>>>>> CODECS IN G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
>> >>>>>>>
>> >>>>>>> CODECS OUT PCMU,PCMA,GSM
>> >>>>>>>
>> >>>>>>> TEL-EVENT 101
>> >>>>>>>
>> >>>>>>> DTMF-MODE rfc2833
>> >>>>>>>
>> >>>>>>> CNG 13
>> >>>>>>>
>> >>>>>>> SESSION-TO 0
>> >>>>>>>
>> >>>>>>> MAX-DIALOG 0
>> >>>>>>>
>> >>>>>>> NOMEDIA false
>> >>>>>>>
>> >>>>>>> LATE-NEG false
>> >>>>>>>
>> >>>>>>> PROXY-MEDIA false
>> >>>>>>>
>> >>>>>>> AGGRESSIVENAT false
>> >>>>>>>
>> >>>>>>> STUN-ENABLED true
>> >>>>>>>
>> >>>>>>> STUN-AUTO-DISABLE false
>> >>>>>>>
>> >>>>>>> CALLS-IN 0
>> >>>>>>>
>> >>>>>>> FAILED-CALLS-IN 0
>> >>>>>>>
>> >>>>>>> CALLS-OUT 0
>> >>>>>>>
>> >>>>>>> FAILED-CALLS-OUT 0
>> >>>>>>>
>> >>>>>>>
>> >>>>>>> On Oct 27, 2010, at 10:04 AM, Anthony Minessale wrote:
>> >>>>>>>
>> >>>>>>> if you map it or not, a scanner would penetrate it.
>> >>>>>>>
>> >>>>>>> There are lot of sip scanners out there now, you just need to
>> beware of
>> >>>>>>> them.
>> >>>>>>>
>> >>>>>>>
>> >>>>>>> On Wed, Oct 27, 2010 at 11:50 AM, Mario G <mario_fs at mgtech.com
>> <mailto:mario_fs at mgtech.com>> wrote:
>> >>>>>>>
>> >>>>>>> Thanks so much! I am sure many others will find this info
>> invaluable. I will
>> >>>>>>> try the static route again but have one question: When I
>> started with FS I
>> >>>>>>> found a "sip scanner" in FS and someone on this group said not
>> to use port
>> >>>>>>> mapping since it was a security risk. Is that true?
>> >>>>>>>
>> >>>>>>> On Oct 27, 2010, at 9:10 AM, Anthony Minessale wrote:
>> >>>>>>>
>> >>>>>>> you are completely guessing at things.
>> >>>>>>>
>> >>>>>>> I want you to understand that the only reason you are having
>> problems
>> >>>>>>>
>> >>>>>>> with this is because you don't understand how it works enough
>> to know
>> >>>>>>>
>> >>>>>>> what you are doing 100%
>> >>>>>>>
>> >>>>>>> Its a given that the pnp stuff is only for your dynamic IP.
>> >>>>>>>
>> >>>>>>> aggressive-nat-detection and sip-force-expires are all related to
>> >>>>>>>
>> >>>>>>> inbound calls when the things who are registering to you may
>> be behind
>> >>>>>>>
>> >>>>>>> nat.
>> >>>>>>>
>> >>>>>>> You need to learn the difference between which nat tools are
>> >>>>>>>
>> >>>>>>> *) designed for your FS to run behind nat
>> >>>>>>>
>> >>>>>>> *) designed for FS to run public and accept connections from
>> devices behind
>> >>>>>>> nat.
>> >>>>>>>
>> >>>>>>> If you have a static IP, you don't need the pnp stuff so
>> -nonat is fine
>> >>>>>>>
>> >>>>>>> What you need to do is set
>> >>>>>>>
>> >>>>>>> 1) set the params ext-sip-ip and ext-rtp-ip to your external
>> static IP
>> >>>>>>>
>> >>>>>>> 2) map the sip ports and all of the rtp ports from your static
>> IP to FS lan
>> >>>>>>> addr
>> >>>>>>>
>> >>>>>>> 3) set sip-ip and rtp-ip to the lan addr you forwarded through.
>> >>>>>>>
>> >>>>>>>
>> >>>>>>> If you don't do this: your outbound registration will use NAT
>> to your
>> >>>>>>>
>> >>>>>>> provider and if there is no activity for the expire time on
>> your NAT
>> >>>>>>>
>> >>>>>>> mapping the reverse port mapping from your provider back to you is
>> >>>>>>>
>> >>>>>>> lost. This is why you set your register expires to a very low
>> number,
>> >>>>>>>
>> >>>>>>> (you need to make sure the provider does not turn the expires
>> back up
>> >>>>>>>
>> >>>>>>> in the reply because it will beat your choice *see sip trace)
>> if this
>> >>>>>>>
>> >>>>>>> is the case then you need the "ping" option set to 30, to
>> continuously
>> >>>>>>>
>> >>>>>>> send an options to your provider.
>> >>>>>>>
>> >>>>>>> The static mapping is obviously the better, easier and more
>> reliable
>> >>>>>>> solution.
>> >>>>>>>
>> >>>>>>> So I want you to understand that the only way to keep a nat mapped
>> >>>>>>>
>> >>>>>>> port alive is to continuously send traffic, all the other
>> methods that
>> >>>>>>>
>> >>>>>>> you are mentioning are to detect that phones registered to
>> your are
>> >>>>>>>
>> >>>>>>> behind nat, I gave you that force-expires option before
>> because your
>> >>>>>>>
>> >>>>>>> trace was full of inbound reg so I thought that is what you wanted
>> >>>>>>>
>> >>>>>>> help with.
>> >>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>>> On Wed, Oct 27, 2010 at 10:43 AM, Mario G <mario_fs at mgtech.com
>> <mailto:mario_fs at mgtech.com>> wrote:
>> >>>>>>>
>> >>>>>>> I should mention that I did not have this problem with an
>> SPA9000 PBX
>> >>>>>>>
>> >>>>>>> (asterisk based) for over two years so FS may be pickier about
>> upnp and/or
>> >>>>>>>
>> >>>>>>> nat, or just better at it exposing a problem in the router.
>> >>>>>>>
>> >>>>>>> I made different changes to the gateways to test different
>> things. One
>> >>>>>>>
>> >>>>>>> failed after 17 hours, the other two stayed up. What did not work:
>> >>>>>>>
>> >>>>>>> added <variable name="sip-force-expires" value="30"/> to the
>> directory
>> >>>>>>>
>> >>>>>>> entries as suggested.
>> >>>>>>>
>> >>>>>>> set the gateway expire times to 30 seconds.
>> >>>>>>>
>> >>>>>>> What worked (could be coincidental) for the two gateways that
>> stayed up:
>> >>>>>>>
>> >>>>>>> I Added <param name="aggressive-nat-detection" value="true"/>
>> >>>>>>>
>> >>>>>>> I originally setup FS to use the static ip by setting external
>> sip/rtp to
>> >>>>>>>
>> >>>>>>> just the static ip (no autonat:) and ran with -nonat but I
>> could not get
>> >>>>>>>
>> >>>>>>> incoming calls. The only way it worked was to use
>> autonat:1.2.3.4. The
>> >>>>>>>
>> >>>>>>> router has 1 static public address and 1 dynamic external IP,
>> this is the
>> >>>>>>>
>> >>>>>>> root of the problem, upnp only tells FS about the dynamic ip
>> Will keep this
>> >>>>>>>
>> >>>>>>> thread up-to-date for anyone who may be in the same boat
>> someday. Thanks
>> >>>>>>>
>> >>>>>>> again for looking at the trace.
>> >>>>>>>
>> >>>>>>> Mario
>> >>>>>>>
>> >>>>>>> You should be setting the req freq to a low number on the
>> outbound gateways
>> >>>>>>>
>> >>>>>>> The examples you showed had a series of inbound reg
>> >>>>>>>
>> >>>>>>> also set expire-seconds to 30 in your gateway xml
>> >>>>>>>
>> >>>>>>>
>> >>>>>>> The problem is if you are not constantly sending traffic to
>> the box
>> >>>>>>>
>> >>>>>>> the nat mapping will go away.
>> >>>>>>>
>> >>>>>>> If you are in production you should be using a static ip with
>> a static
>> >>>>>>>
>> >>>>>>> mapping, any trouble you are having is your own fault for
>> playing with
>> >>>>>>>
>> >>>>>>> fire. The best we can do is tell you how to keep it contained.
>> >>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>>> On Tue, Oct 26, 2010 at 12:34 PM, Mario G <mario_fs at mgtech.com
>> <mailto:mario_fs at mgtech.com>> wrote:
>> >>>>>>>
>> >>>>>>> I made the change. I had no idea the settings for the inside
>> phones effected
>> >>>>>>>
>> >>>>>>> nat for the outside sip accounts. I was looking into
>> aggressive-nat-
>> >>>>>>>
>> >>>>>>> detection since the internal profile status always shows the
>> right external
>> >>>>>>>
>> >>>>>>> static IP but the nat_ap status always shows the dynamic ip.
>> Crossing
>> >>>>>>>
>> >>>>>>> fingers/etc since this problem is 85% of time (weeks!) into FS
>> changeover.
>> >>>>>>>
>> >>>>>>> Thanks!
>> >>>>>>>
>> >>>>>>> Mario
>> >>>>>>>
>> >>>>>>> On Oct 26, 2010, at 10:15 AM, Anthony Minessale wrote:
>> >>>>>>>
>> >>>>>>> add
>> >>>>>>>
>> >>>>>>> <variable name="sip-force-expires" value="30"/>
>> >>>>>>>
>> >>>>>>> to the <variables> section of your <user>
>> >>>>>>>
>> >>>>>>> you have it at 600 and the nat mapping is timing out while the 600
>> >>>>>>>
>> >>>>>>> seconds is ticking away
>> >>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>>> On Tue, Oct 26, 2010 at 12:01 PM, Mario G <mario_fs at mgtech.com
>> <mailto:mario_fs at mgtech.com>> wrote:
>> >>>>>>>
>> >>>>>>> From the TSP:
>> >>>>>>>
>> >>>>>>> "I have enabled the SIP trace on your account. We are not
>> currently seeing
>> >>>>>>>
>> >>>>>>> any registration attempts to your account within the last 15
>> minutes. Please
>> >>>>>>>
>> >>>>>>> restart FreeSwitch so that registration attempts begin again.
>> Thank you. ".
>> >>>>>>>
>> >>>>>>> So FS is not getting past router.
>> >>>>>>>
>> >>>>>>> On Oct 26, 2010, at 9:09 AM, Mario G wrote:
>> >>>>>>>
>> >>>>>>> I ran the global trace during the problem and it is
>> >>>>>>>
>> >>>>>>> at http://pastebin.freeswitch.org/14324 . You can find
>> "rnktel", "acctone",
>> >>>>>>>
>> >>>>>>> "accttwo", "acct3". The trace includes phones since it was
>> global. I am
>> >>>>>>>
>> >>>>>>> using:
>> >>>>>>>
>> >>>>>>> <param name="ext-rtp-ip" value="autonat:my-static.ip"/>
>> >>>>>>>
>> >>>>>>> <param name="ext-sip-ip" value="autonat:my-static.ip"/>
>> >>>>>>>
>> >>>>>>> I tried dumping nat and removing the autonat: above and using
>> -nonat but
>> >>>>>>>
>> >>>>>>> that did not work, registration proceeded but no calls inbound.
>> >>>>>>>
>> >>>>>>> On Oct 25, 2010, at 4:11 PM, Mario G wrote:
>> >>>>>>>
>> >>>>>>> Whoops, I am using an IP address for at least one gateway so
>> that is not the
>> >>>>>>>
>> >>>>>>> problem:
>> >>>>>>>
>> >>>>>>> They look outbound to me and I am using dns for 2 and an IP
>> for one so that
>> >>>>>>>
>> >>>>>>> is not the issue. I was able to get FS to clear this up by
>> doing "nat_map
>> >>>>>>>
>> >>>>>>> reinit" which is why I think this is a nat problem. I will do
>> the trace you
>> >>>>>>>
>> >>>>>>> mentioned. I will plug an ip address into one of the gateways
>> to see what
>> >>>>>>>
>> >>>>>>> happens, they all fail at once. Thanks for responding!
>> >>>>>>>
>> >>>>>>> Mario
>> >>>>>>>
>> >>>>>>> On Oct 25, 2010, at 3:26 PM, Mario wrote:
>> >>>>>>>
>> >>>>>>> I really need help on this as I have weeks into this problem.
>> I thought I
>> >>>>>>>
>> >>>>>>> had it nailed but I guess not. After 5.5 hours I get:
>> >>>>>>>
>> >>>>>>> 2010-10-25 15:05:43.407272 [WARNING] sofia_reg.c:387 mguuid Failed
>> >>>>>>>
>> >>>>>>> Registration, setting retry to 15 seconds.
>> >>>>>>>
>> >>>>>>> 2010-10-25 15:05:49.557478 [NOTICE] sofia_reg.c:342
>> Registering mvuuid
>> >>>>>>>
>> >>>>>>> 2010-10-25 15:05:59.206273 [NOTICE] sofia_reg.c:342
>> Registering mguuid
>> >>>>>>>
>> >>>>>>> 2010-10-25 15:06:04.923157 [WARNING] sofia_reg.c:387 mynum777
>> Failed
>> >>>>>>>
>> >>>>>>> Registration, setting retry to 30 seconds.
>> >>>>>>>
>> >>>>>>> 2010-10-25 15:06:05.358321 [WARNING] sofia_reg.c:387 mvuuid Failed
>> >>>>>>>
>> >>>>>>> Registration, setting retry to 15 seconds.
>> >>>>>>>
>> >>>>>>> 2010-10-25 15:06:16.125060 [WARNING] sofia_reg.c:387 mguuid Failed
>> >>>>>>>
>> >>>>>>> Registration, setting retry to 15 seconds.
>> >>>>>>>
>> >>>>>>> 2010-10-25 15:06:21.151240 [NOTICE] sofia_reg.c:342
>> Registering mvuuid
>> >>>>>>>
>> >>>>>>> 2010-10-25 15:06:33.060421 [NOTICE] sofia_reg.c:342
>> Registering mguuid
>> >>>>>>>
>> >>>>>>> 2010-10-25 15:06:35.392655 [NOTICE] sofia_reg.c:342
>> Registering mynum777
>> >>>>>>>
>> >>>>>>> and no way to make/get calls until I restart FS. I did this:
>> >>>>>>>
>> >>>>>>> 1. log 7
>> >>>>>>>
>> >>>>>>> 2. sofia profile xxxx siptrace on for each profile/gateway
>> >>>>>>>
>> >>>>>>> 3. restarted router
>> >>>>>>>
>> >>>>>>> All three did not solve the problem. The trace and log produced no
>> >>>>>>>
>> >>>>>>> additional lines which is why I am wondering if FS has a
>> problem since the
>> >>>>>>>
>> >>>>>>> trace shows no SIP activity.
>> >>>>>>>
>> >>>>>>> 3 gateways with 2 ITSPs
>> >>>>>>>
>> >>>>>>> 2 DSL/WAN lines, 1 static and 1 dynamic
>> >>>>>>>
>> >>>>>>> I am using autonat:1.2.3.4 in internal and external profiles.
>> 1.2.3.4 is the
>> >>>>>>>
>> >>>>>>> external static ip.
>> >>>>>>>
>> >>>>>>> sofia status profile ... has the right ext ip
>> >>>>>>>
>> >>>>>>> nat_map status shows the dynamic (wrong) IP
>> >>>>>>>
>> >>>>>>> I tried starting with -nonat but that was worse
>> >>>>>>>
>> >>>>>>> the only way to fix is restart FS.
>> >>>>>>>
>> >>>>>>> I read the wiki on external nat, auto_nat and everything else
>> many times.
>> >>>>>>>
>> >>>>>>> Thanks Mario
>> >>>>>>>
>> >>>>>>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From bwibowo at gmail.com  Sun Oct 31 18:11:34 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Mon, 1 Nov 2010 08:11:34 +0700
Subject: [Freeswitch-users] mod_nibblebill cdr
Message-ID: <AANLkTinQTigafVy7h5X4jy-6taijRMYyctE=N=VFyymf@mail.gmail.com>

dear all
i have installed mod_nibblebill and works perfectly, can nibble_bill produce
cdr with info for calling and called number?

regards

budi
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From potxoka at gmail.com  Sun Oct 31 18:52:36 2010
From: potxoka at gmail.com (Antonio)
Date: Mon, 01 Nov 2010 02:52:36 +0100
Subject: [Freeswitch-users] Fax Machine
Message-ID: <4CCE1D64.8080008@gmail.com>

Hello

I'm new to FreeSwitch and am having some doubts with the implantation of 
a fax system.

In the company I am introducing a machine with FreeSwitch that make the 
function of gateway (ISDN & SIP) with different provider. We have a 
server (SIP) for the treatment of users: registration, location, 
acounting (to know you spend each user and department, etc.). I'm having 
real problems (well, I have more since I am a rookie with FreeSwitch) 
with the configuration of sending fax. After checking FreeSwitch wiki, 
the fax receiving does not see much complication, will receive the fax 
and send it to an email address ;-), but the problem comes when I send a 
fax and make the accounting :-(.

If traffic T38, would pass by the main server and this would make the 
accounting treatment, but we will use email sending or winprint (hylafax 
virtual printer) and therefore does not pass through the main server :'( .

He had intended to create in FreeSwitch a sip provider that is the 
connection to the server and forwarding the traffic back to FreeSwitch 
(gateway), so traffic would be recorded.

FreeSwitch (gateway & fax) -> server (accounting) -> FreeSwitch (gateway 
& fax) -> providers.

What do you think this solution? You think of any better idea? Do not 
enter a loop to send traffic that is going to route it back outside? 
thank you very much.

Greetings


From mnhassan at usa.net  Sun Oct 31 21:08:13 2010
From: mnhassan at usa.net (Nyamul Hassan)
Date: Mon, 1 Nov 2010 10:08:13 +0600
Subject: [Freeswitch-users] mod_nibblebill cdr
In-Reply-To: <AANLkTinQTigafVy7h5X4jy-6taijRMYyctE=N=VFyymf@mail.gmail.com>
References: <AANLkTinQTigafVy7h5X4jy-6taijRMYyctE=N=VFyymf@mail.gmail.com>
Message-ID: <AANLkTikT6Z9SB5sjojvkTXxnRtLN9e4WmWkyanLHU-Dg@mail.gmail.com>

Nibble bill does not produce a separate CDR. However, the variables
and their values will be recorded if you are using XML CDR.

Regards
HASSAN


On 2010-11-01, budi wibowo <bwibowo at gmail.com> wrote:
> dear all
> i have installed mod_nibblebill and works perfectly, can nibble_bill produce
> cdr with info for calling and called number?
>
> regards
>
> budi
>

-- 
Sent from my mobile device


From general at bestoronto.com  Sun Oct 31 10:22:37 2010
From: general at bestoronto.com (general at bestoronto.com)
Date: Sun, 31 Oct 2010 13:22:37 -0400
Subject: [Freeswitch-users] Probably obvious but not to me - sofia hangup
Message-ID: <201010311322.37178.general@bestoronto.com>

Hi, I am on freeswitch / centOS: Linux pbx 2.6.18-194.11.3.el5.centos.plus #1 
SMP
It is a fairly straight out of the box freeswitch config, I am not so good 
with XML yet.
Issue is the line to my single SIP phone will drop after about 8 minutes, no 
matter if talking or not.
What might be the cause of this? Seems like a timeout or lack of ACK 
somewhere?
Thanks for the help....
Mark B


From majdi at ieee.org  Sun Oct 31 22:01:09 2010
From: majdi at ieee.org (Majdi Bsoul)
Date: Mon, 1 Nov 2010 05:01:09 +0000
Subject: [Freeswitch-users] How to generate ringing when dialing out through
	dingaling/gtalk
Message-ID: <SNT141-w399F962AB2CBD89AA37E6C5480@phx.gbl>


How can I generate a ringing tone treatment to the dialing extension, when dialing through dingaling/gtalk?
Currently the extension user gets dead air, until the destination answers.
I am running with the most recent GIT FreeSWITCH (10/29/2010).

Regards, 		 	   		  
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From tayeb.meftah at gmail.com  Tue Oct 26 02:09:57 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 26 Oct 2010 10:09:57 +0100
Subject: [Freeswitch-users] dingaling error
In-Reply-To: <AANLkTinGaSAQdWN_1a=zQFUHOc8_xpdgczey81bNryCN@mail.gmail.com>
References: <AANLkTinGaSAQdWN_1a=zQFUHOc8_xpdgczey81bNryCN@mail.gmail.com>
Message-ID: <4CC69AE5.5020003@gmail.com>

did you configured the acl?
check the last configuration line in your default dingaling client mode 
profile
thanks
Le 03/11/2010 04:59, budi wibowo a ?crit :
> dear all
> yesterday my fs works with dingaling module, can send and recieve call 
> from/to google. but this morning it failed.
> i got this error :
> 2010-11-03 11:56:55.814762 [ERR] switch_ivr_originate.c:2612 Cannot 
> create outgoing channel of type [dingaling] cause: 
> [DESTINATION_OUT_OF_ORDER]
>
>
> but if i dial directly from my gmail account call to usa is working
>
> regards
>
> budi
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313

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From tayeb.meftah at gmail.com  Wed Oct 27 05:14:18 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Wed, 27 Oct 2010 13:14:18 +0100
Subject: [Freeswitch-users] pre answer and playback
In-Reply-To: <4CD12AC4.7050200@tagnet.ru>
References: <4CD12AC4.7050200@tagnet.ru>
Message-ID: <4CC8179A.8040009@gmail.com>

check this:
module_exists mod_spandsp
if no try load mod_spandsp
otherwise go to your fs source tree a    nd do:
make mod_spandsp-install
thanks
Le 03/11/2010 10:26, Boris Kovalenko a ?crit :
> Hello!
>
>       When the user entered wrong number I want to redirect it to special
> extension and playback file in pre_answer mode. Is this possible? With
> this extension:
>
> <extension name="WrongNumber">
> <condition field="${destination_number}" expression="^9999$">
> <action application="set" data="bypass_media=false"/>
> <action application="pre_answer"/>
> <action application="endless_playback" data="$${sounds_dir}/misc/inv
> alid_number.wav"/>
> </condition>
> </extension>
>
> i get an error: 2010-11-03 14:22:42.235076 [ERR] sofia_glue.c:2649 No
> audio codec available
>
>    


-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313



From tayeb.meftah at gmail.com  Fri Oct 29 09:24:36 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Fri, 29 Oct 2010 17:24:36 +0100
Subject: [Freeswitch-users] transfer related question
Message-ID: <4CCAF544.4050602@gmail.com>

hi,
if i do blind or atended transfer i like my phone to pass through a 
special dialplan extension befaure thay refer to the other phone
any work arround this?
because i wanna falback to my phone if the transfer fail
using *1 in dp could do it, but using the phone transfer button no idea
thank you

-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313



From tayeb.meftah at gmail.com  Sat Oct 30 09:58:31 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 30 Oct 2010 17:58:31 +0100
Subject: [Freeswitch-users] 2010-11-07 17:17:49.876400 [CRIT]
 switch_loadable_module.c:926 Error Loading module
 /usr/local/freeswitch/mod/mod_dingaling.so ,
 **/usr/local/freeswitch/mod/mod_dingaling.so: undefined symbol:
 switch_channel_get_variable** , 2
In-Reply-To: <4CD6D4D5.9050200@gmail.com>
References: <4CD6D4D5.9050200@gmail.com>
Message-ID: <4CCC4EB7.6020106@gmail.com>

make clean && make && make install will fix it
thanks
Le 07/11/2010 17:33, Pekka Kurki a ?crit :
> get with the latest snapshot load error:
>
> 2010-11-07 17:17:49.876400 [CRIT] switch_loadable_module.c:926 Error
> Loading module /usr/local/freeswitch/mod/mod_dingaling.so
> **/usr/local/freeswitch/mod/mod_dingaling.so: undefined symbol:
> switch_channel_get_variable**
>
> what is wrong?
>
> br
>
> ---pekka---
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313



From tayeb.meftah at gmail.com  Sat Oct 30 10:02:39 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 30 Oct 2010 18:02:39 +0100
Subject: [Freeswitch-users] Problem with bind_meta_app
In-Reply-To: <AANLkTi==-QXJZJ+6gNooxyeSLPJHOzS6iiMuxEiLaLE1@mail.gmail.com>
References: <AANLkTi==-QXJZJ+6gNooxyeSLPJHOzS6iiMuxEiLaLE1@mail.gmail.com>
Message-ID: <4CCC4FAF.2050406@gmail.com>

because *1 is allready inuse
check dialplan/default.xml extension: local_extension
try *5 and see if work
because 1|2|3|4 is allready used
Le 05/11/2010 16:47, Mike van Lammeren a ?crit :
> Hello!
>
> I want *1 to trigger a Lua script, so I configured my dialplan with a 
> call to bind_meta_app like this:
>
> <extension name="my_bridge_call">
> <condition field="destination_number" expression="^11">
> <action application="bind_meta_app" data="1 b b lua::my_trigger.lua"/>
> <action application="lua" data="my_bridge_call.lua"/>
> </condition>
> </extension>
>
> With the configuration above, whenever I press *1 on the b leg, the 
> my_trigger.lua script is launched. That part works great!
>
> However, it seems like something else is also listening for the *1, 
> since notice a few oddities every time I trigger the script:
> 1. I see this error in the console:
>
> 2010-11-05 11:38:53.875766 [ERR] switch_core_file.c:122 Invalid file 
> format [local_stream] for [moh]!
>
> 2. On the a leg, I can hear a noise when dialing *1 on the b leg. I 
> don't quite hear the touch tones, but the noise is definitely 
> associated with them.
>
> 3. After triggering the script, no audio is transferred for about 5 
> seconds.
>
> It looks to me like FreeSWITCH is trying to transfer the calls to 
> music-on-hold, then gets an error, times out, and goes back to the 
> original call. There are no other calls to bind_meta_app, so what else 
> is reacting to the *1?
>
> Thanks!
>
> Mike van Lammeren
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
Phone: +13602276297
Fax: +12538020313

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