Okay , thanks for the info . So then I can proceed with my requirement. ________________________________ From: Steven Ayre <steveayre@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Monday, 17 June 2013, 13:47 Subject: Re: [Freeswitch-users] FreeSwitch as an IVR gateway for External Voicexml services FAS = False Answer Supervision It means answering the call before the call has actually been answered. That's an error for many phone calls as billing starts when the call is answered so it means overbilling during call setup / ringing and for failed calls. For IVR that then forwards the call somewhere that might not be such an issue, but it would be a problem for a SIP provider allowing its users to call PSTN numbers. On Monday, June 17, 2013, luke devon wrote: Hi Jeff
May I know what is FAS error ?
I already install successfully FreeSwitch on CentOS 6.4 and my idea is, use the FreeSwitch with a mobile/telecom operator.
Could you please help me to success this implementation ?
( when I was browsing the web and found Voiceglue software which work as a VXML interpreter. however it is very difficult to install. There is so many dependencies with openvxi-3.4+vglue installation and it is only for asterisk. Hence I gave up struggling with Voiceglue and switch to FreeSwitch. )
Thanks in advance Luke.
________________________________ From: Jeffkingsley Kigg <jeffkingsleykigg@yahoo.com> To: "freeswitch-users@lists.freeswitch.org" <freeswitch-users@lists.freeswitch.org> Sent: Sunday, 16 June 2013, 15:56 Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 84, Issue 154
Hello Luke,
Free switch can do that but remember you will have FAS Error, its OK if you don't use it in Enterprise.
Best regards jeff
________________________________ From: "freeswitch-users-request@lists.freeswitch.org" <freeswitch-users-request@lists.freeswitch.org> To: freeswitch-users@lists.freeswitch.org Sent: Sunday, June 16, 2013 7:04 AM Subject: FreeSWITCH-users Digest, Vol 84, Issue 154
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Contents of FreeSWITCH-users digest..."
Today's Topics:
1. Re: FreeSwitch as a IVR gateway for External Voicexml services (Cal Leeming [Simplicity Media Ltd]) 2. Re: User variables are not inserted when bridging to user (Stanislav Sinyagin) 3. Re: User variables are not inserted when bridging to user (Michael Collins)
Yes, you can achieve this, take some time to review [1] and [2] to get a good understanding of what FS can do.
Hope this helps
Cal
[1] http://wiki.freeswitch.org/wiki/Examples [2] http://wiki.freeswitch.org/wiki/Getting_Started_Guide
On Sat, Jun 15, 2013 at 2:17 PM, luke devon <luke_devon@yahoo.com> wrote:
Dear All,
Mobile User -----Dial IVR Code of Service X ---> Mobile Network --->FreeSwitch platform ------> Re-Directed to IVR Service X
Is this scenario can be done via FreeSwitch ? Please help .
Thanks in advance Luke.
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I solved the problem by enabling only G722 on the client. But still, the user variables seem to be applied in calls from the user only, and not when a call is bridged to the user. After bridging, only the variables in dial-string are taken into account.
So, either Wiki is wrong, or there's a bug in the code.
________________________________ From: Michael Collins <msc@freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Sunday, June 16, 2013 2:18 AM Subject: Re: [Freeswitch-users] User variables are not inserted when bridging to user
Look at the discussion on the wiki about inbound late negotiation - that may help you put all the pieces together.
-MC
http://wiki.freeswitch.org/wiki/Codec_Negotiation
On Fri, Jun 14, 2013 at 6:15 PM, Stanislav Sinyagin <ssinyagin@yahoo.com> wrote:
I'm using the latest 1.2 stable branch.
I want one of my directory users to have specific codec preferences, so I do in my XML directory:
<user id="7012"> <params> <param name="a1-hash"
value="538db5a1dcf95cd9df44bf2ff0430c4b"/>
</params> <variables> <variable name="absolute_codec_string" value="G722,PCMA,PCMU"/> </variables> </user>
But when the call is bridged to user/7012, this variable is not taken into account, and the session inherits the codec from A-leg.
When the user dials out, I see this variable in the "info" output, but not when the user extension gets called.
Any idea why? Wiki says, the user variables should be used in both in- and out- calls, and the source code looks quite right.
thanks
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On Sat, Jun 15, 2013 at 6:12 PM
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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