Does Freeswitch implement the call through service? -- I couldnt find any
relevant application.
By call through service I mean:
enables authorized corporate users outside
CallThrough services are the most common form of service. They require no
registration or pre-payment. On your existing
I'm not sure about the Nokia 6085 but I have successfully used various
mobile phones without a SIP client or wifi by using the Fring
http://www.fring.com http://www.fring.com/downloado and GPRS/3G to
connect to my FS box. If you have a good data plan this can be pretty
economical.
Hope this
Hello,
You'll need to write a script for FS in your choice of language - I
knocked together a quick demo for one in Lua the other week. Something
like this is possibly a good place to start: http://pastebin.freeswitch.org/5479
- change the bits in square brackets for your particular
I have no idea what you are talking about?
What exact dialplan are you using to test att_xfer there is a working
example in the default config.
On Thu, Sep 4, 2008 at 10:24 PM, Lee JJ [EMAIL PROTECTED] wrote:
Hello :
While the att_xfer , I collect show the calls and channels info .
After
talking riddles
On Fri, Sep 5, 2008 at 12:22 AM, Gayatri Kulkarni [EMAIL PROTECTED]wrote:
Can I register it via ENUM then?
*From:* Brian West [EMAIL PROTECTED]
*Sent:* Thursday, September 04, 2008 9:53 AM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users]
Thanks Alex!
I'll try this one
Regards,
Gayatri Kulkarni
On Fri, Sep 5, 2008 at 6:15 PM, Alex Kinch [EMAIL PROTECTED] wrote:
Hello,
You'll need to write a script for FS in your choice of language - I knocked
together a quick demo for one in Lua the other week. Something like this is
I like to try out the freeswitch in real life call, but besides the
connection through the asterisk, it seems that it has no way to directly
support by VOIP service provider.
I am sure some of the VOIP service provider do see the potential business
in this area, but not sure why we still
On Sep 5, 2008, at 11:45 AM, Dave wrote:
I like to try out the freeswitch in real life call, but besides
the connection through the asterisk, it seems that it has no way to
directly support by VOIP service provider.
Look Here:
And Asterlink :P
/b
On Sep 5, 2008, at 2:36 PM, Rupa Schomaker (lists) wrote:
On 9/5/2008 1:45 PM, Dave wrote:
[snip]
Can someone share with me the list of the VOIP service provider who
can support the freeswitch connection, or any good method to
connect to
VOIP service throguh the
I got the compile error on windows XP pro (with VS 2005).
..\..\src\switch_xml.c(2234) : error C2220: warning treated as error -
no 'object' file generated ..\..\src\switch_xml.c(2234) : warning
C4267: '=' : conversion from 'size_t' to 'int', possible loss of data
..\..\src\switch_xml.c(2266)
Is there a way to have an IVR menu take an action besides disconnecting
if the digit timeout is reached or if there is an invalid option?
An IVR that looks like this would be ideal, with two additional actions
menu-timeout and menu-invalid. This would give the IVR a great deal
more
You would put something in the dialplan after the application=ivr
and transfer it elsewhere or execute another ivr. Thats what I would
do.
On Sep 5, 2008, at 7:01 PM, Marc Lewis wrote:
An IVR that looks like this would be ideal, with two additional
actions
menu-timeout and
Hello All,
I have installed freeswitch on a computing cloud (Amazon EC2). My main
network configurations are:
(FreeSwitch ARI LAN ) --- (WAN) --{INTERNET} --- (WAN)
(SIP Soft or Hard Phone LAN)
I have pointed all of my clients to the external (5080) sip port
No matter
Did you happen to
ec2-authorize default -P udp -p 16384-32768
/b
On Sep 5, 2008, at 7:29 PM, Damon Brown wrote:
Hello All,
I have installed freeswitch on a computing cloud (Amazon EC2). My
main network configurations are:
(FreeSwitch ARI LAN ) --- (WAN) --{INTERNET} --- (WAN)
You'll have to set an ext-sip-ip and ext-rtp-ip on the internal.xml
profile on ec2 duplicate them from the external profile.
/b
On Sep 5, 2008, at 8:47 PM, Damon Brown wrote:
Yes, I have all of the valid posts open on my security group
-d
Brian West
sip:[EMAIL PROTECTED]
Try changing these lines and put your ip instead, that worked for me.
X-PRE-PROCESS cmd=set data=external_rtp_ip=stun:stun.freeswitch.org/
X-PRE-PROCESS cmd=set data=external_sip_ip=stun:stun.freeswitch.org/
Diego
On Sat, Sep 6, 2008 at 1:09 AM, Damon Brown [EMAIL PROTECTED] wrote:
Ive
Let me launch mine in 32bit and see what I can do over the weekend...
I had this working without a problem. ;/
/b
On Sep 6, 2008, at 12:09 AM, Damon Brown wrote:
Ive Tried the following with no success:
internal.xml
param name=ext-rtp-ip value=75.101.142.208/
param
Great thanks ... i look forward to your results. I installed on a deb ARI
-Original Message-
From: Brian West [EMAIL PROTECTED]
Sent: Friday, September 5, 2008 10:23pm
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2
Let me launch mine in
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