Hi,
Have you seen this?
http://openbts.sourceforge.net/background.html
I guess, FreeSWITCH would be better for this ;)
Regards,
Tamas
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Hi All,
How to use scheduled hangup from within a java script.
Here is the flow.
User dials 1212 from softphone call goes to default dialplan and calls
test.js
in test.js i am doing like this
session.answer();
//...
//Do some stuff like ivr play etc ...
//
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
Thanks Diego,
I have :
and im still having no luck my guess would be some issue with the elastic
ip translation? then again im drawing at newbie straws im good with the
others and nat/rtp have never been an issue ... is here.
-Original Message-
From: "Diego Viola"
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
>
>
>
>
> Ive also tried just changing the vars.xml file. I al
Try changing these lines and put your ip instead, that worked for me.
Diego
On Sat, Sep 6, 2008 at 1:09 AM, Damon Brown <[EMAIL PROTECTED]> wrote:
> Ive Tried the following with no success:
>
> internal.xml
>
>
>
>
> Ive also tried just changing the vars.xml file. I also tried for
in vars.xml
On Sat, Sep 6, 2008 at 1:21 AM, Diego Viola <[EMAIL PROTECTED]> wrote:
> Try changing these lines and put your ip instead, that worked for me.
>
>
>
>
> Diego
>
> On Sat, Sep 6, 2008 at 1:09 AM, Damon Brown <[EMAIL PROTECTED]> wrote:
>> Ive Tried the following with no success:
>>
>
Ive Tried the following with no success:
internal.xml
Ive also tried just changing the vars.xml file. I also tried forwarding the
incoming rtp connections to my test pc. all with no audio success. I am sure
im doing something wrong I jsut cant find it.
I found another suggestion
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]
__
Yes, I have all of the valid posts open on my security group
-d
-Original Message-
From: "Brian West" <[EMAIL PROTECTED]>
Sent: Friday, September 5, 2008 6:14pm
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] SIP, NAT and Amazon EC2
Did you happen to
ec2-authori
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} ---
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 ma
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 menu-i
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 flexibilit
Lets just put this thread to rest and say It will pretty much work
with ANY voip/itsp out there. ;)
/b
On Sep 5, 2008, at 4:54 PM, Gonzalo Servat wrote:
> Works fine with PennyTel, too.
>
> Gonz
Brian West
sip:[EMAIL PROTECTED]
___
Freesw
Works fine with PennyTel, too.
Gonz
On Fri, Sep 5, 2008 at 4:43 PM, Brian West <[EMAIL PROTECTED]> wrote:
> 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
the TLS issue on windows seems to be unsolvable by me and the interest got
over when i came to know my linksys device doesnt do the TLS which
freeswitch supports.
wouldnt it be better if FS could be backward compatible with the TLS dont by
other devices as i cant seem to find a update for the firm
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) :
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 thr
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 SIP.
> Thanks.
I have it working with voicepulse and vitelity.
-Rupa
http://wiki.freeswitch.org/wiki/Tested_Phone_Providers_Listing
Tested_Phone_Providers_Listing
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View this message in context:
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Sent from the Freeswitch-users mailing list archive at Nabble.
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:
http://wiki.freeswitch.org/wiki/SIP_Provider_Ex
Many VOIP services supporting SIP can work with freeswitch now, for me, like
www.voipstunt.com, www.voipraider.com, www.vbuzzer.com they are all working
fine on my freeswitch via SIP.
Thanks,
Chris
On Fri, Sep 5, 2008 at 2:45 PM, Dave <[EMAIL PROTECTED]> wrote:
>
>I like to try out the freesw
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 didn
I use a 'virtual' PSTN line (voip trunk) from (http://www.voxbone.com) as
incoming external line to my Asterisk server (192.168.1.100)
In my router I have have :
Application Start End ProtocolIP Address
---
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
> po
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-u
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 .
> A
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
confi
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 help
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 teleph
On Sep 4, 2008, at 11:37 PM, Gayatri Kulkarni wrote:
> How do I call a number that is not administered on the FS, through FS?
You need to setup a dial plan that sends the call through a gateway...
Marty
>
> Regards,
> Gayatri Kulkarni
>
> -
> Whenever you find yourself on the side of the ma
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