How can I perform click-to-call or click-to-dial in FreeSWITCH?
Do you have any recommendations on programs capable of click-to-call or
click-to-dial from Microsoft Outlook or Microsoft Excel?
...@domain bridge(u...@domain)';
As for programs able to do that from a Microsoft Product, That I am not sure
of.
Jonathan Pitcher
On 12/16/09 9:59 AM, John Platts wrote:
How can I perform click-to-call or click-to-dial in FreeSWITCH?
Do you have any recommendations on programs
I have uploaded the dialplan and JavaScript files used to process calls to
MODENDP-272. I have even done a make current to revision 15755, and the blind
transfer is still failing.
_
I attempted to do a make current with revision 15739, but some of the Sofia
source files will not compile with revision 15739. Those source files were not
changed between revisions 15738 and 15739. I am using GCC 4.1.2 to compile
FreeSWITCH. I used the following to get revision 15738, which
I have tried to do a blind transfer from a phone that is registered with
FreeSWITCH, and it will fail, even when proxying and media bypass are enabled.
Details about this issue can be found here:
http://jira.freeswitch.org/browse/MODENDP-272
, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 183
X-FS-Support: update_display
Remote-Party-ID: John Platts
sip:19725357...@168.75.202.212;party=calling
:04 PM, John Platts wrote:
I have considered writing JavaScript code to bridge two calls together.
However, I would like to perform custom handling of the 302 Moved
Temporarily response. How do I handle the 302 Moved Temporarily response if
I use JavaScript
I have modified sofia.c in mod_sofia so that I can define gateways without
having to specify the password parameter. This is because I am using a SIP
gateway that does not require SIP registration. The modified version still
requires the password to be set on any gateway for which register is
I was having trouble doing call forwarding from my SIP phone that is connected
to FreeSWITCH. It turns out that my SIP phone is actually sending 302 Moved
Temporarily responses, but my SIP gateway does not support 302 Moved
Temporarily or SIP REFER messages. How do I get FreeSWITCH to forward
-users] Problems with proxy media and bypass media in
FreeSWITCH
This was fixed in trunk yesterday about 8 hrs before you sent this message.
(15619). Please update and try again.
Mike
On Nov 23, 2009, at 11:33 PM, John Platts wrote:
I was using revision 15586
I have considered writing JavaScript code to bridge two calls together.
However, I would like to perform custom handling of the 302 Moved Temporarily
response. How do I handle the 302 Moved Temporarily response if I use
JavaScript?
...@freeswitch.org
Date: Tue, 24 Nov 2009 15:32:44 -0600
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Call forwarding problem
You'll have to hairpin the media thru your machine usually if they
won't accept either of those.
/b
On Nov 24, 2009, at 3:05 PM, John Platts wrote
I actually checked out the latest version of FreeSWITCH in the SVN repository.
I have the following configured in
/usr/local/freeswitch/conf/dialplan/default.xml:
extension name=setup_media continue=true
condition field=${sip_nat_detected} expression=true
action
On Nov 23, 2009, at 6:19 PM, John Platts wrote:
I actually checked out the latest version of FreeSWITCH in the SVN
repository.
I have the following configured in /usr/local/freeswitch/conf/
dialplan/default.xml:
___
FreeSWITCH
I have installed FreeSWITCH on our server, and need some help configuring our
FreeSWITCH instance. All of the numbers associated with our FreeSWITCH instance
are in the format: 1NPANXX (where NPA is the area code, and NXX are the
last 7 digits of the phone number).
I need
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