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| Author |
Message |
sridhart at alcatel-lu...
Guest
|
 Posted: Mon Mar 02, 2009 7:01 am Post subject: [Freeswitch-users] Running freeswitch on powerpc |
 |
|
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
| Quote: | -----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] On
Behalf Of freeswitch-users-request@lists.freeswitch.org
Sent: Monday, February 02, 2009 9:12 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Freeswitch-users Digest, Vol 32, Issue 17
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When replying, please edit your Subject line so it is more
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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@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@gmail.com>
Subject: Re: [Freeswitch-users] Call Variable not available when call
hangup
To: freeswitch-users@lists.freeswitch.org
Message-ID: <21791503.post@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:
| Quote: |
the leg you are running the script on is not hungup, the
|
other leg of the
| Quote: | 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
| Quote: | 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
| Quote: |
On Mon, Feb 2, 2009 at 6:53 AM, shehzad p <pmhshz@gmail.com> wrote:
| Quote: |
Is there any settings that when call hangup control can be
|
|
transferred to
| Quote: | | Quote: | another context and these CDR values can be accessible
|
|
there? (just like
| Quote: | | Quote: | in
Asterisk, h extension)
shehzad p wrote:
| Quote: |
Hi all,
I need to process some CDR variables in Dialplan, like
|
|
|
call duration,
| Quote: | | Quote: | | Quote: | Answered time etc.
but when I place info application after bridge, it is
|
|
|
not listing them
| Quote: | | Quote: | | Quote: | 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
| Quote: | | Quote: | get
| Quote: | those values from dialplan itself so that can be passed to some
|
script...
--
View this message in context:
|
|
http://www.nabble.com/Call-Variable-not-available-when-call-ha
ngup-tp21788550p21789152.html
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
itch-users
<MSN%3Aanthony_minessale@hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aantho
ny.minessale@gmail.com>
| Quote: | IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
|
<sip%3A888@conference.freeswitch.org>
| Quote: | iax:guest@conference.freeswitch.org/888
|
googletalk:conf+888@conference.freeswitch.org<googletalk%3Acon
f%2B888@conference.freeswitch.org>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
itch-users
| Quote: | 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@eugeneweb.com
Subject: Re: [Freeswitch-users] How do I set my FS internal ip address
to a "static" value.
To: freeswitch-users@lists.freeswitch.org
Message-ID: <Pine.LNX.4.62.0902011323230.2342@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( , ifdown(
# 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@archive> sofia status
API CALL [sofia(status)] output:
Name Type
Data
State
==============================================================
===================================
external profile
sip:mod_sofia@67.171.158.226:5080
RUNNING (0)
internal profile
sip:mod_sofia@192.168.0.249:5060
RUNNING (2)
nat profile
sip:mod_sofia@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@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@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@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@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@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@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@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@67.171.158.226:5080
RUNNING (0)
internal profile
sip:mod_sofia@192.168.0.249:5060
RUNNING (0)
outbound alias
external
ALIASED
192.168.0.249 alias
internal
ALIASED
nat profile
sip:mod_sofia@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@gmail.com>
Subject: Re: [Freeswitch-users] Call Variable not available when call
hangup
To: freeswitch-users@lists.freeswitch.org
Message-ID:
<191c3a030902020741k779e2488o38ca578a3b40e9ad@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@gmail.com> wrote:
| Quote: |
one question is that when javascript is being called from
|
dial plan, I get
| Quote: | 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
| Quote: | for B leg also?
Anthony Minessale-2 wrote:
| Quote: |
the leg you are running the script on is not hungup, the
|
|
other leg of the
| Quote: | | Quote: | 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
| Quote: | | Quote: | 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
| Quote: | | Quote: |
On Mon, Feb 2, 2009 at 6:53 AM, shehzad p
|
|
<pmhshz@gmail.com> wrote:
| Quote: | | Quote: |
| Quote: |
Is there any settings that when call hangup control can
|
|
|
be transferred
| Quote: | to
| Quote: | | Quote: | another context and these CDR values can be accessible
|
|
|
there? (just like
| Quote: | | Quote: | | Quote: | in
Asterisk, h extension)
shehzad p wrote:
| Quote: |
Hi all,
I need to process some CDR variables in Dialplan, like
|
|
|
|
call duration,
| Quote: | | Quote: | | Quote: | | Quote: | Answered time etc.
but when I place info application after bridge, it is
|
|
|
|
not listing them
| Quote: | | Quote: | | Quote: | | Quote: | 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
| Quote: | | Quote: | | Quote: | get
| Quote: | those values from dialplan itself so that can be passed to some
|
script...
--
View this message in context:
|
|
|
http://www.nabble.com/Call-Variable-not-available-when-call-ha
ngup-tp21788550p21789152.html
| Quote: | | Quote: | | Quote: | Sent from the Freeswitch-users mailing list archive at
|
|
|
Nabble.com.
| Quote: | http://lists.freeswitch.org/mailman/options/freeswitch-users
|
<MSN%3Aanthony_minessale@hotmail.com><
MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@
hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aantho
ny.minessale@gmail.com>
<PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.mines
sale@gmail.com>
| Quote: | | Quote: |
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
|
|
<sip%3A888@conference.freeswitch.org><
sip%3A888@conference.freeswitch.org<sip%253A888@conference.fre
eswitch.org>
| Quote: | | Quote: |
iax:guest@conference.freeswitch.org/888
|
|
googletalk:conf+888@conference.freeswitch.org<googletalk%3Acon
f%2B888@conference.freeswitch.org>
<googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%
253Aconf%252B888@conference.freeswitch.org>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
itch-users
| Quote: | | Quote: | http://www.freeswitch.org
|
--
View this message in context:
|
http://www.nabble.com/Call-Variable-not-available-when-call-ha
ngup-tp21788550p21791503.html
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
itch-users
| Quote: | http://www.freeswitch.org
|
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com
<MSN%3Aanthony_minessale@hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aantho
ny.minessale@gmail.com>
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
<sip%3A888@conference.freeswitch.org>
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org<googletalk%3Acon
f%2B888@conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
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------------------------------
Message: 4
Date: Mon, 2 Feb 2009 09:41:39 -0600
From: Brian West <brian@freeswitch.org>
Subject: Re: [Freeswitch-users] How do I set my FS internal ip address
to a "static" value.
To: freeswitch-users@lists.freeswitch.org
Message-ID: <05AC6B23-7F34-4C89-8BD0-1744BEF20B4C@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@eugeneweb.com wrote:
| Quote: | Hi Gang,
I've been struggleing with this also. Actually I can get it
|
to bind
| Quote: | 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
| Quote: | between
two extensions and it will go for hours. Then I add anther
|
interface
| Quote: | 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(, ifdown(
| Quote: |
# 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
| Quote: | 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@archive> sofia status
API CALL [sofia(status)] output:
Name Type
|
Data
| Quote: | State
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
|
======================================================================
sip:mod_sofia@67.171.158.226:5080
| Quote: | RUNNING (0)
internal profile
|
sip:mod_sofia@192.168.0.249:5060
| Quote: | RUNNING (2)
nat profile
|
sip:mod_sofia@67.171.158.226:5070
| Quote: | RUNNING (0)
default alias
|
internal
| Quote: | ALIASED
outbound alias
|
external
| Quote: | ALIASED
192.168.0.249 alias
|
internal
| Quote: | ALIASED
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
|
======================================================================
| Quote: | 3 profiles 3 aliases
freeswitch@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
| Quote: | sofia/internal/
1003@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@192.168.0.249
| Quote: | [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@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@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@192.168.0.249
| Quote: | )
Ended
2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809
switch_core_session_thread() Close Channel
|
sofia/internal/1001@192.168.0.249
| Quote: | [CS_HANGUP]
2009-02-01 13:23:19 [NOTICE] sofia.c:645
|
sofia_profile_thread_run()
| Quote: | waiting for
worker thread
2009-02-01 13:23:19 [NOTICE] sofia.c:645
|
sofia_profile_thread_run()
| Quote: | waiting for
worker thread
2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia()
|
Adding Alias
| Quote: | [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
| Quote: | [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()
| Quote: | 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
| Quote: | State
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
|
======================================================================
sip:mod_sofia@67.171.158.226:5080
| Quote: | RUNNING (0)
internal profile
|
sip:mod_sofia@192.168.0.249:5060
| Quote: | RUNNING (0)
outbound alias
|
external
| Quote: | ALIASED
192.168.0.249 alias
|
internal
| Quote: | ALIASED
nat profile
|
sip:mod_sofia@67.171.158.226:5070
| Quote: | RUNNING (0)
default alias
|
internal
| Quote: | ALIASED
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
|
======================================================================
| Quote: | 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
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
itch-users
| Quote: | http://www.freeswitch.org
|
------------------------------
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@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@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org |
|
| Back to top |
|
 |
gmaruzz at celliax.org
Guest
|
| Posted: Mon Mar 02, 2009 7:19 am Post subject: [Freeswitch-users] Running freeswitch on powerpc |
|
|
On Mon, Mar 2, 2009 at 12:52 PM, Rajagopal, Sridhar (Sridhar)
<sridhart@alcatel-lucent.com> wrote:
| Quote: | 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@alcatel-lucent.com> wrote:
| Quote: | 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
| Quote: | -----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] On
Behalf Of freeswitch-users-request@lists.freeswitch.org
Sent: Monday, February 02, 2009 9:12 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Freeswitch-users Digest, Vol 32, Issue 17
Send Freeswitch-users mailing list submissions to
freeswitch-users@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
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You can reach the person managing the list at
freeswitch-users-owner@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@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@gmail.com>
Subject: Re: [Freeswitch-users] Call Variable not available when call
hangup
To: freeswitch-users@lists.freeswitch.org
Message-ID: <21791503.post@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:
| Quote: |
the leg you are running the script on is not hungup, the
|
other leg of the
| Quote: | 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
| Quote: | 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
| Quote: |
On Mon, Feb 2, 2009 at 6:53 AM, shehzad p <pmhshz@gmail.com> wrote:
| Quote: |
Is there any settings that when call hangup control can be
|
|
transferred to
| Quote: | | Quote: | another context and these CDR values can be accessible
|
|
there? (just like
| Quote: | | Quote: | in
Asterisk, h extension)
shehzad p wrote:
| Quote: |
Hi all,
I need to process some CDR variables in Dialplan, like
|
|
|
call duration,
| Quote: | | Quote: | | Quote: | Answered time etc.
but when I place info application after bridge, it is
|
|
|
not listing them
| Quote: | | Quote: | | Quote: | 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
| Quote: | | Quote: | get
| Quote: | those values from dialplan itself so that can be passed to some
|
script...
--
View this message in context:
|
|
http://www.nabble.com/Call-Variable-not-available-when-call-ha
ngup-tp21788550p21789152.html
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
itch-users
<MSN%3Aanthony_minessale@hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aantho
ny.minessale@gmail.com>
| Quote: | IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
|
<sip%3A888@conference.freeswitch.org>
| Quote: | iax:guest@conference.freeswitch.org/888
|
googletalk:conf+888@conference.freeswitch.org<googletalk%3Acon
f%2B888@conference.freeswitch.org>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
itch-users
| Quote: | 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@eugeneweb.com
Subject: Re: [Freeswitch-users] How do I set my FS internal ip address
to a "static" value.
To: freeswitch-users@lists.freeswitch.org
Message-ID: <Pine.LNX.4.62.0902011323230.2342@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(, ifdown(
# 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@archive> sofia status
API CALL [sofia(status)] output:
Name Type
Data
State
==============================================================
===================================
external profile
sip:mod_sofia@67.171.158.226:5080
RUNNING (0)
internal profile
sip:mod_sofia@192.168.0.249:5060
RUNNING (2)
nat profile
sip:mod_sofia@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@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@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@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@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@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@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@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@67.171.158.226:5080
RUNNING (0)
internal profile
sip:mod_sofia@192.168.0.249:5060
RUNNING (0)
outbound alias
external
ALIASED
192.168.0.249 alias
internal
ALIASED
nat profile
sip:mod_sofia@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@gmail.com>
Subject: Re: [Freeswitch-users] Call Variable not available when call
hangup
To: freeswitch-users@lists.freeswitch.org
Message-ID:
<191c3a030902020741k779e2488o38ca578a3b40e9ad@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@gmail.com> wrote:
| Quote: |
one question is that when javascript is being called from
|
dial plan, I get
| Quote: | 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
| Quote: | for B leg also?
Anthony Minessale-2 wrote:
| Quote: |
the leg you are running the script on is not hungup, the
|
|
other leg of the
| Quote: | | Quote: | 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
| Quote: | | Quote: | 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
| Quote: | | Quote: |
On Mon, Feb 2, 2009 at 6:53 AM, shehzad p
|
|
<pmhshz@gmail.com> wrote:
| Quote: | | Quote: |
| Quote: |
Is there any settings that when call hangup control can
|
|
|
be transferred
| Quote: | to
| Quote: | | Quote: | another context and these CDR values can be accessible
|
|
|
there? (just like
| Quote: | | Quote: | | Quote: | in
Asterisk, h extension)
shehzad p wrote:
| Quote: |
Hi all,
I need to process some CDR variables in Dialplan, like
|
|
|
|
call duration,
| Quote: | | Quote: | | Quote: | | Quote: | Answered time etc.
but when I place info application after bridge, it is
|
|
|
|
not listing them
| Quote: | | Quote: | | Quote: | | Quote: | 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
| Quote: | | Quote: | | Quote: | get
| Quote: | those values from dialplan itself so that can be passed to some
|
script...
--
View this message in context:
|
|
|
http://www.nabble.com/Call-Variable-not-available-when-call-ha
ngup-tp21788550p21789152.html
| Quote: | | Quote: | | Quote: | Sent from the Freeswitch-users mailing list archive at
|
|
|
Nabble.com.
| Quote: | http://lists.freeswitch.org/mailman/options/freeswitch-users
|
<MSN%3Aanthony_minessale@hotmail.com><
MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@
hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aantho
ny.minessale@gmail.com>
<PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.mines
sale@gmail.com>
| Quote: | | Quote: |
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
|
|
<sip%3A888@conference.freeswitch.org><
sip%3A888@conference.freeswitch.org<sip%253A888@conference.fre
eswitch.org>
| Quote: | | Quote: |
iax:guest@conference.freeswitch.org/888
|
|
googletalk:conf+888@conference.freeswitch.org<googletalk%3Acon
f%2B888@conference.freeswitch.org>
<googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%
253Aconf%252B888@conference.freeswitch.org>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
itch-users
| Quote: | | Quote: | http://www.freeswitch.org
|
--
View this message in context:
|
http://www.nabble.com/Call-Variable-not-available-when-call-ha
ngup-tp21788550p21791503.html
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
itch-users
| Quote: | http://www.freeswitch.org
|
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com
<MSN%3Aanthony_minessale@hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aantho
ny.minessale@gmail.com>
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
<sip%3A888@conference.freeswitch.org>
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org<googletalk%3Acon
f%2B888@conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
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------------------------------
Message: 4
Date: Mon, 2 Feb 2009 09:41:39 -0600
From: Brian West <brian@freeswitch.org>
Subject: Re: [Freeswitch-users] How do I set my FS internal ip address
to a "static" value.
To: freeswitch-users@lists.freeswitch.org
Message-ID: <05AC6B23-7F34-4C89-8BD0-1744BEF20B4C@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@eugeneweb.com wrote:
| Quote: | Hi Gang,
I've been struggleing with this also. Actually I can get it
|
to bind
| Quote: | 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
| Quote: | between
two extensions and it will go for hours. Then I add anther
|
interface
| Quote: | 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(, ifdown(
| Quote: |
# 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
| Quote: | 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@archive> sofia status
API CALL [sofia(status)] output:
Name Type
|
Data
| Quote: | State
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
|
======================================================================
sip:mod_sofia@67.171.158.226:5080
| Quote: | RUNNING (0)
internal profile
|
sip:mod_sofia@192.168.0.249:5060
| Quote: | RUNNING (2)
nat profile
|
sip:mod_sofia@67.171.158.226:5070
| Quote: | RUNNING (0)
default alias
|
internal
| Quote: | ALIASED
outbound alias
|
external
| Quote: | ALIASED
192.168.0.249 alias
|
internal
| Quote: | ALIASED
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
|
======================================================================
| Quote: | 3 profiles 3 aliases
freeswitch@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
| Quote: | sofia/internal/
1003@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@192.168.0.249
| Quote: | [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@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@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@192.168.0.249
| Quote: | )
Ended
2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809
switch_core_session_thread() Close Channel
|
sofia/internal/1001@192.168.0.249
| Quote: | [CS_HANGUP]
2009-02-01 13:23:19 [NOTICE] sofia.c:645
|
sofia_profile_thread_run()
| Quote: | waiting for
worker thread
2009-02-01 13:23:19 [NOTICE] sofia.c:645
|
sofia_profile_thread_run()
| Quote: | waiting for
worker thread
2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia()
|
Adding Alias
| Quote: | [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
| Quote: | [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()
| Quote: | 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
| Quote: | State
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
|
======================================================================
sip:mod_sofia@67.171.158.226:5080
| Quote: | RUNNING (0)
internal profile
|
sip:mod_sofia@192.168.0.249:5060
| Quote: | RUNNING (0)
outbound alias
|
external
| Quote: | ALIASED
192.168.0.249 alias
|
internal
| Quote: | ALIASED
nat profile
|
sip:mod_sofia@67.171.158.226:5070
| Quote: | RUNNING (0)
default alias
|
internal
| Quote: | ALIASED
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
|
======================================================================
| Quote: | 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
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
itch-users
| Quote: | http://www.freeswitch.org
|
------------------------------
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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itch-users
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End of Freeswitch-users Digest, Vol 32, Issue 17
************************************************
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| Back to top |
|
|
wojtek at VoIPMan.ORG
Guest
|
| Posted: Mon Mar 02, 2009 7:41 am Post subject: [Freeswitch-users] Running freeswitch on powerpc |
|
|
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:
| Quote: | On Mon, Mar 2, 2009 at 12:52 PM, Rajagopal, Sridhar (Sridhar)
<sridhart@alcatel-lucent.com> wrote:
| Quote: | 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@alcatel-lucent.com> wrote:
| Quote: | 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
| Quote: | -----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] On
Behalf Of freeswitch-users-request@lists.freeswitch.org
Sent: Monday, February 02, 2009 9:12 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Freeswitch-users Digest, Vol 32, Issue 17
Send Freeswitch-users mailing list submissions to
freeswitch-users@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@lists.freeswitch.org
You can reach the person managing the list at
freeswitch-users-owner@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@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@gmail.com>
Subject: Re: [Freeswitch-users] Call Variable not available when
call
hangup
To: freeswitch-users@lists.freeswitch.org
Message-ID: <21791503.post@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:
| Quote: |
the leg you are running the script on is not hungup, the
|
other leg of the
| Quote: | 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
| Quote: | 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
| Quote: |
On Mon, Feb 2, 2009 at 6:53 AM, shehzad p <pmhshz@gmail.com> wrote:
| Quote: |
Is there any settings that when call hangup control can be
|
|
transferred to
| Quote: | | Quote: | another context and these CDR values can be accessible
|
|
there? (just like
| Quote: | | Quote: | in
Asterisk, h extension)
shehzad p wrote:
| Quote: |
Hi all,
I need to process some CDR variables in Dialplan, like
|
|
|
call duration,
| Quote: | | Quote: | | Quote: | Answered time etc.
but when I place info application after bridge, it is
|
|
|
not listing them
| Quote: | | Quote: | | Quote: | 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
| Quote: | | Quote: | get
| Quote: | those values from dialplan itself so that can be passed to some
|
script...
--
View this message in context:
|
|
http://www.nabble.com/Call-Variable-not-available-when-call-ha
ngup-tp21788550p21789152.html
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
itch-users
<MSN%3Aanthony_minessale@hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aantho
ny.minessale@gmail.com>
| Quote: | IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
|
<sip%3A888@conference.freeswitch.org>
| Quote: | iax:guest@conference.freeswitch.org/888
|
googletalk:conf+888@conference.freeswitch.org<googletalk%3Acon
f%2B888@conference.freeswitch.org>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
itch-users
| Quote: | 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@eugeneweb.com
Subject: Re: [Freeswitch-users] How do I set my FS internal ip
address
to a "static" value.
To: freeswitch-users@lists.freeswitch.org
Message-ID: <Pine.LNX.4.62.0902011323230.2342@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(,
ifdown(
# 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@archive> sofia status
API CALL [sofia(status)] output:
Name Type
Data
State
==============================================================
===================================
external profile
sip:mod_sofia@67.171.158.226:5080
RUNNING (0)
internal profile
sip:mod_sofia@192.168.0.249:5060
RUNNING (2)
nat profile
sip:mod_sofia@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@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@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@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@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@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@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@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@67.171.158.226:5080
RUNNING (0)
internal profile
sip:mod_sofia@192.168.0.249:5060
RUNNING (0)
outbound alias
external
ALIASED
192.168.0.249 alias
internal
ALIASED
nat profile
sip:mod_sofia@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@gmail.com>
Subject: Re: [Freeswitch-users] Call Variable not available when
call
hangup
To: freeswitch-users@lists.freeswitch.org
Message-ID:
<191c3a030902020741k779e2488o38ca578a3b40e9ad@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@gmail.com> wrote:
| Quote: |
one question is that when javascript is being called from
|
dial plan, I get
| Quote: | 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
| Quote: | for B leg also?
Anthony Minessale-2 wrote:
| Quote: |
the leg you are running the script on is not hungup, the
|
|
other leg of the
| Quote: | | Quote: | 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
| Quote: | | Quote: | 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
| Quote: | | Quote: |
On Mon, Feb 2, 2009 at 6:53 AM, shehzad p
|
|
<pmhshz@gmail.com> wrote:
| Quote: | | Quote: |
| Quote: |
Is there any settings that when call hangup control can
|
|
|
be transferred
| Quote: | to
| Quote: | | Quote: | another context and these CDR values can be accessible
|
|
|
there? (just like
| Quote: | | Quote: | | Quote: | in
Asterisk, h extension)
shehzad p wrote:
| Quote: |
Hi all,
I need to process some CDR variables in Dialplan, like
|
|
|
|
call duration,
| Quote: | | Quote: | | Quote: | | Quote: | Answered time etc.
but when I place info application after bridge, it is
|
|
|
|
not listing them
| Quote: | | Quote: | | Quote: | | Quote: | 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
| Quote: | | Quote: | | Quote: | get
| Quote: | those values from dialplan itself so that can be passed to some
|
script...
--
View this message in context:
|
|
|
http://www.nabble.com/Call-Variable-not-available-when-call-ha
ngup-tp21788550p21789152.html
| Quote: | | Quote: | | Quote: | Sent from the Freeswitch-users mailing list archive at
|
|
|
Nabble.com.
| Quote: | http://lists.freeswitch.org/mailman/options/freeswitch-users
|
<MSN%3Aanthony_minessale@hotmail.com><
MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@
hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aantho
ny.minessale@gmail.com>
<PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.mines
sale@gmail.com>
| Quote: | | Quote: |
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
|
|
<sip%3A888@conference.freeswitch.org><
sip%3A888@conference.freeswitch.org<sip%253A888@conference.fre
eswitch.org>
| Quote: | | Quote: |
iax:guest@conference.freeswitch.org/888
|
|
googletalk:conf+888@conference.freeswitch.org<googletalk%3Acon
f%2B888@conference.freeswitch.org>
<googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%
253Aconf%252B888@conference.freeswitch.org>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
itch-users
| Quote: | | Quote: | http://www.freeswitch.org
|
--
View this message in context:
|
http://www.nabble.com/Call-Variable-not-available-when-call-ha
ngup-tp21788550p21791503.html
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
itch-users
| Quote: | http://www.freeswitch.org
|
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com
<MSN%3Aanthony_minessale@hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aantho
ny.minessale@gmail.com>
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
<sip%3A888@conference.freeswitch.org>
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org<googletalk%3Acon
f%2B888@conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
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------------------------------
Message: 4
Date: Mon, 2 Feb 2009 09:41:39 -0600
From: Brian West <brian@freeswitch.org>
Subject: Re: [Freeswitch-users] How do I set my FS internal ip
address
to a "static" value.
To: freeswitch-users@lists.freeswitch.org
Message-ID: <05AC6B23-7F34-4C89-8BD0-1744BEF20B4C@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@eugeneweb.com wrote:
| Quote: | Hi Gang,
I've been struggleing with this also. Actually I can get it
|
to bind
| Quote: | 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
| Quote: | between
two extensions and it will go for hours. Then I add anther
|
interface
| Quote: | 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(, ifdown(
| Quote: |
# 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
| Quote: | 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@archive> sofia status
API CALL [sofia(status)] output:
Name Type
|
Data
| Quote: | State
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
|
=
=
====================================================================
sip:mod_sofia@67.171.158.226:5080
| Quote: | RUNNING (0)
internal profile
|
sip:mod_sofia@192.168.0.249:5060
| Quote: | RUNNING (2)
nat profile
|
sip:mod_sofia@67.171.158.226:5070
| Quote: | RUNNING (0)
default alias
|
internal
| Quote: | ALIASED
outbound alias
|
external
| Quote: | ALIASED
192.168.0.249 alias
|
internal
| Quote: | ALIASED
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
|
=
=
====================================================================
| Quote: | 3 profiles 3 aliases
freeswitch@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
| Quote: | sofia/internal/
1003@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@192.168.0.249
| Quote: | [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@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@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@192.168.0.249
| Quote: | )
Ended
2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809
switch_core_session_thread() Close Channel
|
sofia/internal/1001@192.168.0.249
| Quote: | [CS_HANGUP]
2009-02-01 13:23:19 [NOTICE] sofia.c:645
|
sofia_profile_thread_run()
| Quote: | waiting for
worker thread
2009-02-01 13:23:19 [NOTICE] sofia.c:645
|
sofia_profile_thread_run()
| Quote: | waiting for
worker thread
2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia()
|
Adding Alias
| Quote: | [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
| Quote: | [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()
| Quote: | 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
| Quote: | State
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
|
=
=
====================================================================
sip:mod_sofia@67.171.158.226:5080
| Quote: | RUNNING (0)
internal profile
|
sip:mod_sofia@192.168.0.249:5060
| Quote: | RUNNING (0)
outbound alias
|
external
| Quote: | ALIASED
192.168.0.249 alias
|
internal
| Quote: | ALIASED
nat profile
|
sip:mod_sofia@67.171.158.226:5070
| Quote: | RUNNING (0)
default alias
|
internal
| Quote: | ALIASED
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
=
|
=
=
====================================================================
| Quote: | 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
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
itch-users
| Quote: | http://www.freeswitch.org
|
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steveu at coppice.org
Guest
|
| Posted: Mon Mar 02, 2009 8:02 am Post subject: [Freeswitch-users] Running freeswitch on powerpc |
|
|
Rajagopal, Sridhar (Sridhar) wrote:
| Quote: | 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
|
It may be easier to say what will currently stop Freeswitch working.
The lack of an MMU is a problem right now, so Blackfins are out, which
is sad. Cores without hardware floating point may not perform all that
well, but should work. Endianness should not be a problem. I think
machines which choke on misaligned access are probably OK, too.
Checking that list, you should be OK on a PPC.
Steve
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