[Freeswitch-users] Running freeswitch on powerpc
Rajagopal, Sridhar (Sridhar)
sridhart at alcatel-lucent.com
Mon Mar 2 03:52:10 PST 2009
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
>
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> 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...
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> 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
>
>
>
>
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>
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> End of Freeswitch-users Digest, Vol 32, Issue 17
> ************************************************
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