This question can be separated into two part:
1.Pass a call to another FS
2.Receive a call from another FS
Somebody can tell me how to do these??
Please.
-Brad
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* This question can be separated into two part:
** 1.Pass a call to another FS
*
uuid_deflect or uuid_transfer, depending on whether the call has been
answered
by the first FS instance or not. See the wiki.
*
** 2.Receive a call from another FS
*
Provide a dial plan entry in the second FS that
I have tried
extension name=remoteFreeswitch
condition field=destination_number expression=^014(\d+)$
action application=bridge data=sofia/external/$1 at
192.168.141.187http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
:5080/
/condition
/extension
But the console
Like this??
* extension name=remoteFreeswitch
** condition field=destination_number expression=^014(\d+)$
** *action application=bridge
data=sofia/${use_profile}/$1%192.168.141.187/
* /condition
** /extension
*
I have tried,but Fs still return:
[INFO] mod_dialplan_xml.c:233
I have tried
extension name=transfer_to_MeiLan
condition field=destination_number expression=^014(\d+)$
action application=bridge data=sofia/internal/$1%192.168.141.187/
/condition
/extension
But FS still return the same message
[WARNING] mod_sofia.c:2534
As title
How to reload xml without using console command line??
-Brad
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()
Originate
Failed. Cause: USER_NOT_REGISTERED
2009/6/2 Jason White ja...@jasonjgw.net
Brad Tuan brad.t...@gmail.com wrote:
I have tried
extension name=transfer_to_MeiLan
condition field=destination_number expression=^014(\d+)$
action application=bridge
data=sofia/internal/$1
? You should have the IP
address of the remote FS machine
--
*From: *Brad Tuan brad.t...@gmail.com
*Reply-To: *freeswitch-users@lists.freeswitch.org
*Date: *Tue, 2 Jun 2009 17:00:29 +0800
*To: *freeswitch-users@lists.freeswitch.org
*Subject: *Re: [Freeswitch-users
Why the Caller-Username is 97719006 but the Caller-Caller-ID-Name is
Extension 97730002??
EVENT DUMP:
Channel-State: [CS_ROUTING]
Channel-State-Number: [2]
Channel-Name: [sofia/internal/sip:97730...@210.68.184.192:62101
;rinstance=16b8076934af7da9]
Unique-ID:
I don't have a 97719006 User in my FS.
It was passed from another sip proxy.
2009/6/2 Brian West br...@freeswitch.org
Chances are that is what you set it to on the user. Verify the users
settings in the directory.
/b
On Jun 2, 2009, at 6:28 AM, Brad Tuan wrote:
Why the Caller-Username
=U3QF8QUp1F3tQ
To: sip:97730...@210.68.184.192:62113;rinstance=d0ed6224d69efcf1
Call-ID: 8dcc44c8-ca18-122c-2780-39a48cb53b8d
CSeq: 11586 INVITE
2009/6/2 Brad Tuan brad.t...@gmail.com
I don't have a 97719006 User in my FS.
It was passed from another sip proxy.
2009/6/2 Brian West br
So I need to new a User(97719006) in directory\default ??
2009/6/2 Brian West br...@freeswitch.org
I would update if I were you! :) Anyway something had to have changed it
it won't magically do it.
/b
On Jun 2, 2009, at 8:28 AM, Brad Tuan wrote:
User-Agent: FreeSWITCH-mod_sofia
User1(FS1) ,
User2(FS2) display call established,but User1(FS1) still display
calling.
Why??
(I think maybe that I need to do some setting on FS2.)
2009/6/2 Brad Tuan brad.t...@gmail.com
FS1's IP is 192.168.141.182 and FS2's IP is 192.168.141.187
These two FS are in the same LAN.
I just try
a=rtpmap:13 CN/8000
a=ptime:20
2009/6/3 Michael Collins m...@freeswitch.org
On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan brad.t...@gmail.com wrote:
How to update FreeSWITCH-mod_sofia/1.0.3-12163??
Your best bet is to use SVN trunk. It is the most stable version available,
even more stable
or console
loglevel 7) and also do the SIP trace. Make a few test calls and capture
all the output.
-MC
On Tue, Jun 2, 2009 at 9:38 PM, Brad Tuan brad.t...@gmail.com wrote:
..I've update my FS by SVN..
but the User-Agent became to FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
extension name=group-intercept
condition field=destination_number expression=^\*8$
action application=answer/
action application=intercept
data=${hash(select/${domain_name}-last_dial/${callgroup})}/
action application=sleep data=2000/
/condition
/extension
What is the ${callgroup}
or console
loglevel 7) and also do the SIP trace. Make a few test calls and capture
all the output.
-MC
On Tue, Jun 2, 2009 at 9:38 PM, Brad Tuan brad.t...@gmail.com wrote:
..I've update my FS by SVN..
but the User-Agent became to FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
User1(FS1) ,
User2(FS2) display call established,but User1(FS1) still display
calling.
Why??
(I think maybe that I need to do some setting on FS2.)
2009/6/2 Brad Tuan brad.t...@gmail.com
FS1's IP is 192.168.141.182 and FS2's IP is 192.168.141.187
These two FS are in the same LAN.
I just
When User(1001) calling with User(1002) ,
how to transfer User(1002) to User(1003)??
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If i don't want to use softphone function to transfer the call ,how to do
it??
2009/6/4 Brian West br...@freeswitch.org
Depends.. Press the transfer key on your phone is how I would do it.. what
kind of phone do you have?
/b
On Jun 4, 2009, at 1:40 AM, Brad Tuan wrote:
When User(1001
I know why the display name is wrong..
in conf\directory\97730002.xml
include
user id=97730002 mailbox=97730002 cidr=163.28.32.51/32
I forgot this setting...
but if I don't set cidr here ,the call from 163.28.32.51 can't come into my
FS.
How to make some setting for this??
Tuan wrote:
If i don't want to use softphone function to transfer the call ,how to do
it??
2009/6/4 Brian West br...@freeswitch.org
Depends.. Press the transfer key on your phone is how I would do it.. what
kind of phone do you have?
/b
On Jun 4, 2009, at 1:40 AM, Brad Tuan wrote
Yes I have tried it,but useless,
when 1001 and 1002 are talking to each other ,
then 1001 want to transfer 1002 to 1003,
so 1001 press *1 1003,
but nothing happen...
2009/6/4 dujinfang dujinf...@gmail.com
yes. Did you ever tried that?
On Jun 4, 2009, at 5:36 PM, Brad Tuan wrote:
I
change to a (for a-leg) or ab for both leg.
action application=bind_meta_app data=0 b s
execute_extension::dx XML features/
check bind_meta_app for detail on wiki, I bet you never tried the att_xfer
feature.
On Jun 5, 2009, at 10:43 AM, Brad Tuan wrote:
Yes I have tried
As title,
How to receive a call from another SIP proxy??
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broad question... can you tell us if you have to register to
said proxy?
/b
On Jun 7, 2009, at 8:28 PM, Brad Tuan wrote:
As title,
How to receive a call from another SIP proxy??
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
I have already set it in my FreeSwitch\conf\sip_profiles\external.xml:
gateways
gateway name=FreeSwitch
param name=username value=1008/
param name=realm value=218.210.xxx.xxx/
param name=password value=1234/
param name=extension value=1008/
I know that i need to set the dialplan,
my problem is when FS_B send a REGISTER to FS_A, FS_A will return a 403 to
FS_B
Like this:
2009-06-30 15:03:25 [NOTICE] sofia_reg.c:305 sofia_reg_check_gateway()
Registering FS_A
2009-06-30 15:03:25 [ERR] sofia_reg.c:1391 sofia_reg_handle_sip_r_register()
As title ,Does FS keep the status of gateways??
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Another question, Where does FS keep these information??
In *.db or somewhere??
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For example, send a INVITE 1001...@xxx.xxx.xxx.xxx to my FS user 1001
How to do this??
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Useless , the dialplan was changed like this:
include
extension name=1001_trans
condition field=destination_number expression=^1001$
action application=bridge
data=sofia/profile/1001123${regex(${sofia_contact(
1...@${domain})}|^...@]+(.*)|%1)}/
/condition
/extension
/include
Useless , the dialplan was changed like this:
include
extension name=1001_trans
condition field=destination_number expression=^1001$
action application=bridge
data=sofia/profile/1001123${regex(${sofia_contact(
1...@${domain})}|^...@]+(.*)|%1)}/
/condition
/extension
/include
As title, How to custom the Subject and Body and ... of the mail ??
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How to set the date format , and the IVR flow ??
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As title ,I know how to do when sending INVITE
but how to do it when fs sending REGISTER??
For example , when gateway registering , the contact is
gw+a...@xxx.xxx.xxx.xxx ,
how to change it to *a...@xxx.xxx.xxx.xxx??*
**
*Please help*
**
**
___
If my PC has two IP address, How to decide the IP which using to send
REGISTER message??
For example, I have two IP , 172.30.30.XXX for External and 192.168.60.XXX
for Internal,
and now i want to add two gateway, one is 210.XXX.XXX.XXX and the other one
is 192.168.60.30,
How to change the IP
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