[Freeswitch-users] Running freeswitch on powerpc

Wojciech Tryc wojtek at VoIPMan.ORG
Mon Mar 2 04:32:31 PST 2009


Sridhar,
PIKA's WARP is PowerPC based...AMCC but still Big Endian and PowerPC.
 From what I remember the endianness definition was broken in one or  
two places, but other than that it was effortless (native compilation).

Thanks,
Wojtek,

On Mar 2, 2009, at 7:11 AM, Giovanni Maruzzelli wrote:

> On Mon, Mar 2, 2009 at 12:52 PM, Rajagopal, Sridhar (Sridhar)
> <sridhart at alcatel-lucent.com> wrote:
>> I am planning to run freeswitch on powerpc MPC8358. Please let me  
>> know if any changes needs to be done in the code
>
> Hi Sridhar,
>
> I don't think someone has tried that. It will probably be you that let
> us all know which (if any) changes needs to be done. :-)
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> =========================================
> www.celliax.org
> via Pierlombardo 9, 20135 Milano
> Italy
> gmaruzz at celliax dot org
> Cell : +39-347-2665618
> Fax : +39-02-87390039
>
>
>
>
> On Mon, Mar 2, 2009 at 12:52 PM, Rajagopal, Sridhar (Sridhar)
> <sridhart at alcatel-lucent.com> wrote:
>> Hi all,
>>
>> I am planning to run freeswitch on powerpc MPC8358. Please let me  
>> know if any changes needs to be done in the code
>>
>> Regards
>> Sridhar
>>
>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On
>>> Behalf Of freeswitch-users-request at lists.freeswitch.org
>>> Sent: Monday, February 02, 2009 9:12 PM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Freeswitch-users Digest, Vol 32, Issue 17
>>>
>>> Send Freeswitch-users mailing list submissions to
>>>      freeswitch-users at lists.freeswitch.org
>>>
>>> To subscribe or unsubscribe via the World Wide Web, visit
>>>      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> or, via email, send a message with subject or body 'help' to
>>>      freeswitch-users-request at lists.freeswitch.org
>>>
>>> You can reach the person managing the list at
>>>      freeswitch-users-owner at lists.freeswitch.org
>>>
>>> When replying, please edit your Subject line so it is more
>>> specific than "Re: Contents of Freeswitch-users digest..."
>>>
>>>
>>> Today's Topics:
>>>
>>>   1. Re: Call Variable not available when call hangup (shehzad p)
>>>   2. Re: How do I set my FS internal ip address to a "static"
>>>      value. (clif at eugeneweb.com)
>>>   3. Re: Call Variable not available when call hangup
>>>      (Anthony Minessale)
>>>   4. Re: How do I set my FS internal ip address to a "static"
>>>      value. (Brian West)
>>>
>>>
>>> ----------------------------------------------------------------------
>>>
>>> Message: 1
>>> Date: Mon, 2 Feb 2009 07:21:32 -0800 (PST)
>>> From: shehzad p <pmhshz at gmail.com>
>>> Subject: Re: [Freeswitch-users] Call Variable not available when  
>>> call
>>>      hangup
>>> To: freeswitch-users at lists.freeswitch.org
>>> Message-ID: <21791503.post at talk.nabble.com>
>>> Content-Type: text/plain; charset=us-ascii
>>>
>>>
>>>
>>> one question is that when javascript is being called from
>>> dial plan, I get the session object already available, It is
>>> for A leg of channel, So when javascript is called after
>>> Bridge how can I get the session object for B leg also?
>>>
>>>
>>> Anthony Minessale-2 wrote:
>>>>
>>>> the leg you are running the script on is not hungup, the
>>> other leg of the
>>>> call is.
>>>>
>>>> If it was hungup you would not be executing the script.
>>>>
>>>> Asterisk and the h ext and the whole dead-agi thing are all
>>> poor design
>>>> showing it's teeth.
>>>> We do not support anything like it.
>>>>
>>>>
>>>> You can however try this: (see the link below)
>>>>
>>>>
>>> http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks
>>> -giving-me-headaches-p21614840.html
>>>>
>>>>
>>>>
>>>> On Mon, Feb 2, 2009 at 6:53 AM, shehzad p <pmhshz at gmail.com> wrote:
>>>>
>>>>>
>>>>> Is there any settings that when call hangup control can be
>>> transferred to
>>>>> another context and these CDR values can be accessible
>>> there? (just like
>>>>> in
>>>>> Asterisk, h extension)
>>>>>
>>>>> shehzad p wrote:
>>>>>>
>>>>>> Hi all,
>>>>>>
>>>>>> I need to process some CDR variables in Dialplan, like
>>> call duration,
>>>>>> Answered time etc.
>>>>>> but when I place info application after bridge, it is
>>> not listing them
>>>>>> properly as below:
>>>>>> ===========================================
>>>>>> Caller-Channel-Created-Time: [1233573341672157]
>>>>>> Caller-Channel-Answered-Time: [1233573342712939]
>>>>>> Caller-Channel-Hangup-Time: [0]
>>>>>> ==========================================
>>>>>> Here Hangup time is 0, So how can I find actual values?
>>>>>>
>>>>>> --I know that we can use xml_cdr or cdr_csv, but my
>>> current need is to
>>>>> get
>>>>>> those values from dialplan itself so that can be passed to some
>>>>> script...
>>>>>>
>>>>>>
>>>>>> thanks,
>>>>>> msp
>>>>>>
>>>>>
>>>>> --
>>>>> View this message in context:
>>>>>
>>> http://www.nabble.com/Call-Variable-not-available-when-call-ha
>>> ngup-tp21788550p21789152.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/freesw
>>> itch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>> <MSN%3Aanthony_minessale at hotmail.com>
>>>>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aantho
>>> ny.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>> <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>>
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Acon
>>> f%2B888 at conference.freeswitch.org>
>>>> pstn:213-799-1400
>>>>
>>>> _______________________________________________
>>>> Freeswitch-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
>>>>
>>>>
>>>
>>> --
>>> View this message in context:
>>> http://www.nabble.com/Call-Variable-not-available-when-call-ha
>>> ngup-tp21788550p21791503.html
>>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>>
>>>
>>>
>>>
>>> ------------------------------
>>>
>>> Message: 2
>>> Date: Sun, 1 Feb 2009 13:24:20 -0800 (PST)
>>> From: clif at eugeneweb.com
>>> Subject: Re: [Freeswitch-users] How do I set my FS internal ip  
>>> address
>>>      to a    "static" value.
>>> To: freeswitch-users at lists.freeswitch.org
>>> Message-ID: <Pine.LNX.4.62.0902011323230.2342 at redwall>
>>> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
>>>
>>> Hi Gang,
>>>
>>> I've been struggleing with this also. Actually I can get it
>>> to bind to my
>>> address, the problem is it randomly drops my calls. :-(
>>>
>>> I have a FS running on a box with a static IP and I can start
>>> a call between
>>> two extensions and it will go for hours. Then I add anther
>>> interface say eth0:0
>>> with a new static IP and reconfigure my phones and FS to use
>>> that, and the
>>> calls drop after about 15-20 mins. Though it's pretty random.
>>>
>>> Here is my setup. I have Debian Linux 2.6.23.1 kernel, and
>>> freeswitch-1.0.1.
>>> Here is my /etc/network/interfaces:
>>>
>>> # /etc/network/interfaces -- configuration file for ifup(8),  
>>> ifdown(8)
>>>
>>> # The loopback interface
>>> auto lo
>>> iface lo inet loopback
>>>
>>> # The first network card - this entry was created during the Debian
>>> installation
>>> auto eth0 eth0:0
>>> iface eth0 inet dhcp
>>> iface eth0:0 inet static
>>>      address 192.168.0.249
>>>      netmask 255.255.255.0
>>>      gateway 192.168.0.254
>>>
>>> The only change I made to the FS config is in Vars.xml. I
>>> added this line close
>>> to the top:
>>>
>>> <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.0.249"/>
>>>
>>> Here is the console log of the call being dropped:
>>>
>>> freeswitch at archive> sofia status
>>> API CALL [sofia(status)] output:
>>>                      Name       Type
>>>      Data
>>> State
>>> ==============================================================
>>> ===================================
>>>                  external    profile
>>> sip:mod_sofia at 67.171.158.226:5080
>>> RUNNING (0)
>>>                  internal    profile
>>> sip:mod_sofia at 192.168.0.249:5060
>>> RUNNING (2)
>>>                       nat    profile
>>> sip:mod_sofia at 67.171.158.226:5070
>>> RUNNING (0)
>>>                   default      alias
>>>  internal
>>> ALIASED
>>>                  outbound      alias
>>>  external
>>> ALIASED
>>>             192.168.0.249      alias
>>>  internal
>>> ALIASED
>>> ==============================================================
>>> ===================================
>>> 3 profiles 3 aliases
>>>
>>> freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634
>>> sofia_glue_restart_all_profiles() Reload XML [Success]
>>> 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler()
>>> ENUM Reloaded
>>> 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568
>>> sofia_read_frame() Hangup
>>> sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp
>>> ;fs_nat=yes
>>> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>>> 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820
>>> switch_ivr_multi_threaded_bridge() Hangup
>>> sofia/internal/1001 at 192.168.0.249
>>> [CS_EXECUTE] [NORMAL_CLEARING]
>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807
>>> switch_core_session_thread() Session 6
>>> (sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=ud
>>> p;fs_nat=yes)
>>> Ended
>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809
>>> switch_core_session_thread() Close Channel
>>> sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp
>>> ;fs_nat=yes
>>> [CS_HANGUP]
>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807
>>> switch_core_session_thread() Session 5
>>> (sofia/internal/1001 at 192.168.0.249)
>>> Ended
>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809
>>> switch_core_session_thread() Close Channel
>>> sofia/internal/1001 at 192.168.0.249
>>> [CS_HANGUP]
>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:645
>>> sofia_profile_thread_run() waiting for
>>> worker thread
>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:645
>>> sofia_profile_thread_run() waiting for
>>> worker thread
>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding  
>>> Alias
>>> [192.168.0.249] for profile [internal]
>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia()
>>> Adding Alias [default]
>>> for profile [internal]
>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia()
>>> Started Profile
>>> internal [sofia_reg_internal]
>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() Adding  
>>> Alias
>>> [outbound] for profile [external]
>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia()
>>> Started Profile
>>> external [sofia_reg_external]
>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:645
>>> sofia_profile_thread_run() waiting for
>>> worker thread
>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia()
>>> Started Profile nat
>>> [sofia_reg_nat]
>>> sofia status
>>> API CALL [sofia(status)] output:
>>>                      Name       Type
>>>      Data
>>> State
>>> ==============================================================
>>> ===================================
>>>                  external    profile
>>> sip:mod_sofia at 67.171.158.226:5080
>>> RUNNING (0)
>>>                  internal    profile
>>> sip:mod_sofia at 192.168.0.249:5060
>>> RUNNING (0)
>>>                  outbound      alias
>>>  external
>>> ALIASED
>>>             192.168.0.249      alias
>>>  internal
>>> ALIASED
>>>                       nat    profile
>>> sip:mod_sofia at 67.171.158.226:5070
>>> RUNNING (0)
>>>                   default      alias
>>>  internal
>>> ALIASED
>>> ==============================================================
>>> ===================================
>>> 3 profiles 3 aliases
>>>
>>> There is an older thread that says one should set
>>> <X-PRE-PROCESS cmd="set" data="bind_server_ip=192.168.0.249"/>
>>> but in this (later) thread is says only Jingleling usese that
>>> variable.
>>> ie. see:
>>> http://www.mail-archive.com/freeswitch-users@lists.freeswitch.
>>> org/msg00695.html
>>> http://www.mail-archive.com/freeswitch-users@lists.freeswitch.
>>> org/msg07345.html
>>>
>>> So what do you think causes this? What is the correct way? ;-)
>>>
>>>
>>>      Thanks,
>>>      Clif
>>>
>>>
>>>
>>>
>>> ------------------------------
>>>
>>> Message: 3
>>> Date: Mon, 2 Feb 2009 09:41:05 -0600
>>> From: Anthony Minessale <anthony.minessale at gmail.com>
>>> Subject: Re: [Freeswitch-users] Call Variable not available when  
>>> call
>>>      hangup
>>> To: freeswitch-users at lists.freeswitch.org
>>> Message-ID:
>>>      <191c3a030902020741k779e2488o38ca578a3b40e9ad at mail.gmail.com>
>>> Content-Type: text/plain; charset="iso-8859-1"
>>>
>>> you can't that's why i said it was a horrible approach.
>>> That's also why i posted you the instructions on the only
>>> elegant solution
>>> to your problem.
>>>
>>>
>>> On Mon, Feb 2, 2009 at 9:21 AM, shehzad p <pmhshz at gmail.com> wrote:
>>>
>>>>
>>>>
>>>> one question is that when javascript is being called from
>>> dial plan, I get
>>>> the session object already available, It is for A leg of channel,
>>>> So when javascript is called after Bridge how can I get the
>>> session object
>>>> for B leg also?
>>>>
>>>>
>>>> Anthony Minessale-2 wrote:
>>>>>
>>>>> the leg you are running the script on is not hungup, the
>>> other leg of the
>>>>> call is.
>>>>>
>>>>> If it was hungup you would not be executing the script.
>>>>>
>>>>> Asterisk and the h ext and the whole dead-agi thing are
>>> all poor design
>>>>> showing it's teeth.
>>>>> We do not support anything like it.
>>>>>
>>>>>
>>>>> You can however try this: (see the link below)
>>>>>
>>>>>
>>>>
>>> http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks
>>> -giving-me-headaches-p21614840.html
>>>>>
>>>>>
>>>>>
>>>>> On Mon, Feb 2, 2009 at 6:53 AM, shehzad p
>>> <pmhshz at gmail.com> wrote:
>>>>>
>>>>>>
>>>>>> Is there any settings that when call hangup control can
>>> be transferred
>>>> to
>>>>>> another context and these CDR values can be accessible
>>> there? (just like
>>>>>> in
>>>>>> Asterisk, h extension)
>>>>>>
>>>>>> shehzad p wrote:
>>>>>>>
>>>>>>> Hi all,
>>>>>>>
>>>>>>> I need to process some CDR variables in Dialplan, like
>>> call duration,
>>>>>>> Answered time etc.
>>>>>>> but when I place info application after bridge, it is
>>> not listing them
>>>>>>> properly as below:
>>>>>>> ===========================================
>>>>>>> Caller-Channel-Created-Time: [1233573341672157]
>>>>>>> Caller-Channel-Answered-Time: [1233573342712939]
>>>>>>> Caller-Channel-Hangup-Time: [0]
>>>>>>> ==========================================
>>>>>>> Here Hangup time is 0, So how can I find actual values?
>>>>>>>
>>>>>>> --I know that we can use xml_cdr or cdr_csv, but my
>>> current need is to
>>>>>> get
>>>>>>> those values from dialplan itself so that can be passed to some
>>>>>> script...
>>>>>>>
>>>>>>>
>>>>>>> thanks,
>>>>>>> msp
>>>>>>>
>>>>>>
>>>>>> --
>>>>>> View this message in context:
>>>>>>
>>>>
>>> http://www.nabble.com/Call-Variable-not-available-when-call-ha
>>> ngup-tp21788550p21789152.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/
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com
>>> <MSN%3Aanthony_minessale at hotmail.com><
>>>>
>>> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale@
>>> hotmail.com>
>>>>>
>>>>>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aantho
>>> ny.minessale at gmail.com>
>>>>
>>> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.mines
>>> sale at gmail.com>
>>>>>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
>>> <sip%3A888 at conference.freeswitch.org><
>>>>
>>> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.fre
>>> eswitch.org>
>>>>>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>>
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Acon
>>> f%2B888 at conference.freeswitch.org>
>>>>
>>> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%
>>> 253Aconf%252B888 at conference.freeswitch.org>
>>>>>
>>>>> pstn:213-799-1400
>>>>>
>>>>> _______________________________________________
>>>>> Freeswitch-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
>>>>>
>>>>>
>>>>
>>>> --
>>>> View this message in context:
>>>>
>>> http://www.nabble.com/Call-Variable-not-available-when-call-ha
>>> ngup-tp21788550p21791503.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/freesw
>>> itch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aantho
>>> ny.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Acon
>>> f%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>> -------------- next part --------------
>>> An HTML attachment was scrubbed...
>>> URL:
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/attachm
>>> ents/20090202/2d430e44/attachment-0001.html
>>>
>>> ------------------------------
>>>
>>> Message: 4
>>> Date: Mon, 2 Feb 2009 09:41:39 -0600
>>> From: Brian West <brian at freeswitch.org>
>>> Subject: Re: [Freeswitch-users] How do I set my FS internal ip  
>>> address
>>>      to a    "static" value.
>>> To: freeswitch-users at lists.freeswitch.org
>>> Message-ID: <05AC6B23-7F34-4C89-8BD0-1744BEF20B4C at freeswitch.org>
>>> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes
>>>
>>> you need to add this setting to sofia.conf.xml
>>>
>>> <param name="auto-restart" value="false"/>
>>>
>>>
>>> You'll also need to edit the sofia profiles and input the
>>> exact IP you
>>> wish it to bind to.  The params are sip-ip and rtp-ip.
>>>
>>> /b
>>>
>>> On Feb 1, 2009, at 3:24 PM, clif at eugeneweb.com wrote:
>>>
>>>> Hi Gang,
>>>>
>>>> I've been struggleing with this also. Actually I can get it
>>> to bind
>>>> to my
>>>> address, the problem is it randomly drops my calls. :-(
>>>>
>>>> I have a FS running on a box with a static IP and I can
>>> start a call
>>>> between
>>>> two extensions and it will go for hours. Then I add anther
>>> interface
>>>> say eth0:0
>>>> with a new static IP and reconfigure my phones and FS to use that,
>>>> and the
>>>> calls drop after about 15-20 mins. Though it's pretty random.
>>>>
>>>> Here is my setup. I have Debian Linux 2.6.23.1 kernel, and
>>>> freeswitch-1.0.1.
>>>> Here is my /etc/network/interfaces:
>>>>
>>>> # /etc/network/interfaces -- configuration file for
>>> ifup(8), ifdown(8)
>>>>
>>>> # The loopback interface
>>>> auto lo
>>>> iface lo inet loopback
>>>>
>>>> # The first network card - this entry was created during the Debian
>>>> installation
>>>> auto eth0 eth0:0
>>>> iface eth0 inet dhcp
>>>> iface eth0:0 inet static
>>>>    address 192.168.0.249
>>>>    netmask 255.255.255.0
>>>>    gateway 192.168.0.254
>>>>
>>>> The only change I made to the FS config is in Vars.xml. I
>>> added this
>>>> line close
>>>> to the top:
>>>>
>>>> <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.0.249"/>
>>>>
>>>> Here is the console log of the call being dropped:
>>>>
>>>> freeswitch at archive> sofia status
>>>> API CALL [sofia(status)] output:
>>>>                     Name      Type
>>>      Data
>>>> State
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>>
>>> = 
>>> = 
>>> ====================================================================
>>>>                 external   profile
>>> sip:mod_sofia at 67.171.158.226:5080
>>>> RUNNING (0)
>>>>                 internal   profile
>>> sip:mod_sofia at 192.168.0.249:5060
>>>> RUNNING (2)
>>>>                      nat   profile
>>> sip:mod_sofia at 67.171.158.226:5070
>>>> RUNNING (0)
>>>>                  default     alias
>>>  internal
>>>> ALIASED
>>>>                 outbound     alias
>>>  external
>>>> ALIASED
>>>>            192.168.0.249     alias
>>>  internal
>>>> ALIASED
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>>
>>> = 
>>> = 
>>> ====================================================================
>>>> 3 profiles 3 aliases
>>>>
>>>> freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634
>>>> sofia_glue_restart_all_profiles() Reload XML [Success]
>>>> 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() ENUM
>>>> Reloaded
>>>> 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568
>>> sofia_read_frame() Hangup
>>>> sofia/internal/
>>>> 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes
>>>> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>>>> 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820
>>>> switch_ivr_multi_threaded_bridge() Hangup
>>> sofia/internal/1001 at 192.168.0.249
>>>> [CS_EXECUTE] [NORMAL_CLEARING]
>>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807
>>>> switch_core_session_thread() Session 6
>>>> (sofia/internal/
>>>> 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes)
>>>> Ended
>>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809
>>>> switch_core_session_thread() Close Channel
>>>> sofia/internal/
>>>> 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes
>>>> [CS_HANGUP]
>>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807
>>>> switch_core_session_thread() Session 5
>>> (sofia/internal/1001 at 192.168.0.249
>>>> )
>>>> Ended
>>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809
>>>> switch_core_session_thread() Close Channel
>>> sofia/internal/1001 at 192.168.0.249
>>>> [CS_HANGUP]
>>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:645
>>> sofia_profile_thread_run()
>>>> waiting for
>>>> worker thread
>>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:645
>>> sofia_profile_thread_run()
>>>> waiting for
>>>> worker thread
>>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia()
>>> Adding Alias
>>>> [192.168.0.249] for profile [internal]
>>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding
>>>> Alias [default]
>>>> for profile [internal]
>>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() Started
>>>> Profile
>>>> internal [sofia_reg_internal]
>>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia()
>>> Adding Alias
>>>> [outbound] for profile [external]
>>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started
>>>> Profile
>>>> external [sofia_reg_external]
>>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:645
>>> sofia_profile_thread_run()
>>>> waiting for
>>>> worker thread
>>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started
>>>> Profile nat
>>>> [sofia_reg_nat]
>>>> sofia status
>>>> API CALL [sofia(status)] output:
>>>>                     Name      Type
>>>      Data
>>>> State
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>>
>>> = 
>>> = 
>>> ====================================================================
>>>>                 external   profile
>>> sip:mod_sofia at 67.171.158.226:5080
>>>> RUNNING (0)
>>>>                 internal   profile
>>> sip:mod_sofia at 192.168.0.249:5060
>>>> RUNNING (0)
>>>>                 outbound     alias
>>>  external
>>>> ALIASED
>>>>            192.168.0.249     alias
>>>  internal
>>>> ALIASED
>>>>                      nat   profile
>>> sip:mod_sofia at 67.171.158.226:5070
>>>> RUNNING (0)
>>>>                  default     alias
>>>  internal
>>>> ALIASED
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>>
>>> = 
>>> = 
>>> ====================================================================
>>>> 3 profiles 3 aliases
>>>>
>>>> There is an older thread that says one should set
>>>> <X-PRE-PROCESS cmd="set" data="bind_server_ip=192.168.0.249"/>
>>>> but in this (later) thread is says only Jingleling usese that
>>>> variable.
>>>> ie. see:
>>>>
>>> http://www.mail-archive.com/freeswitch-users@lists.freeswitch.
>>> org/msg00695.html
>>>>
>>> http://www.mail-archive.com/freeswitch-users@lists.freeswitch.
>>> org/msg07345.html
>>>>
>>>> So what do you think causes this? What is the correct way? ;-)
>>>>
>>>>
>>>>    Thanks,
>>>>    Clif
>>>>
>>>>
>>>> _______________________________________________
>>>> Freeswitch-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/freesw
>>> itch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> End of Freeswitch-users Digest, Vol 32, Issue 17
>>> ************************************************
>>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





More information about the FreeSWITCH-users mailing list