Re: [OpenSIPS-Users] Opensips IVR

[Francjos]

I would like to use FreeSwithc as media server. Whiwh configurations do i
have  to put in place in both Opensips and Freeswitch?



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Re: [OpenSIPS-Users] Opensips IVR

[Francjos]

Thaks for your reply. I 'll use Freeswitch and opensips in front of
freeswitch



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Re: [OpenSIPS-Users] Opensips IVR

[SamyGo]

Hi,
OpenSIPS is a SIP proxy and it ma ynot be the best tool for making it do
IVR, just wondering how would opensips capture a DTMF? and play a new file
!!.
It is recommended that you use any Media-Server to handle IVR with easy,
like SEMS, FreeSwitch, Asterisk.


Regards,
Sammy


On Fri, Mar 25, 2016 at 10:09 AM, Francjos <35...@heb.be> wrote:

> Hello every body,
>
> Is it possible to implement an IVR (Interact Voice Response) with opensips?
> If yes, how is it implemented?
>
> Thanks
>
>
>
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[OpenSIPS-Users] Opensips IVR

[Francjos]

Hello every body,

Is it possible to implement an IVR (Interact Voice Response) with opensips?
If yes, how is it implemented?

Thanks



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Re: [OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL

[Terrance Devor]

Vlad, I think we could all benefit from the snippets if you know
what I mean ;).

T
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Re: [OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL

[Vlad]

I will send you my few lines code...hope to help you
Write you back asapIl 14/Mar/2015 18:11 mahan77  ha scritto:
>
> Hello again Danilo,
>
> Thank you for the quick replay.
>
> I have asterisk server running at public IP. 
>
> I have to use IVR, Voicemail, on hold message and incoming DDIs.  All
> incoming DDI send direct to asterisk IP.
>
> Some DDI will play welcome message while phone rings, others will ring
> group and after certain time it will go direct to closed message. All these
> functions are working with asterisk right now.  I’m getting high-level sip
> flood attack. Now I’m trying to secure server with OpenSIPS.  That’s why if
> I see your scripts it will help me understand more.  Looking forward to your
> email on Monday
>
> Many thanks
> sathees
>
>
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Re: [OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL

[SamyGo]

Thats sound like you need to ask for script doing the flood blocking and
security rather than IVR call control etc.



On Sat, Mar 14, 2015 at 1:11 PM, mahan77  wrote:

> Hello again Danilo,
>
> Thank you for the quick replay.
>
> I have asterisk server running at public IP.
>
> I have to use IVR, Voicemail, on hold message and incoming DDIs.  All
> incoming DDI send direct to asterisk IP.
>
>  Some DDI will play welcome message while phone rings, others will ring
> group and after certain time it will go direct to closed message. All these
> functions are working with asterisk right now.  I’m getting high-level sip
> flood attack. Now I’m trying to secure server with OpenSIPS.  That’s why if
> I see your scripts it will help me understand more.  Looking forward to
> your
> email on Monday
>
> Many thanks
> sathees
>
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/OPENSIPS-IVR-CALL-CONTROL-tp7595634p7595890.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
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Re: [OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL

[mahan77]

Hello again Danilo,

Thank you for the quick replay.

I have asterisk server running at public IP.  

I have to use IVR, Voicemail, on hold message and incoming DDIs.  All
incoming DDI send direct to asterisk IP.

 Some DDI will play welcome message while phone rings, others will ring
group and after certain time it will go direct to closed message. All these
functions are working with asterisk right now.  I’m getting high-level sip
flood attack. Now I’m trying to secure server with OpenSIPS.  That’s why if
I see your scripts it will help me understand more.  Looking forward to your
email on Monday

Many thanks
sathees




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Re: [OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL

[Stefano Pisani]

You should say something more about your issue.



Il 14/03/2015 17:22, mahan77 ha scritto:
Hi Danilo, I’m having problem with OpenSips => Asterisk connection. 
Can you able to mail me your working OpenSips scripts. mail at 
Sathees.co.uk appreciate sathees


View this message in context: Re: OPENSIPS + IVR CALL CONTROL 

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Re: [OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL

[danilo...@tin.it]

Hi
I'm currently out of office.
I will drop you an email on Monday
Actually I got it working, but please tell me more about your scenario or
post your section code

Regards
Danilo
Il 14/Mar/2015 17:22 "mahan77"  ha scritto:

> Hi Danilo, I’m having problem with OpenSips => Asterisk connection. Can
> you able to mail me your working OpenSips scripts. mail at Sathees.co.uk
> appreciate sathees
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Re: [OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL

[mahan77]

Hi Danilo,I’m having problem with OpenSips => Asterisk connection. Can you
able to mail me your working OpenSips scripts. mail at
Sathees.co.ukappreciatesathees



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Re: [OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL

[SamyGo]

Hi Danilo,
Can you just use application *Dial(SIP/OpenSips/${EXTEN})*  at the IVR to
send call back to the opensips server.Thats how I'd do to send call back to
OpenSIPS.

BR,
Sammy

On Thu, Mar 5, 2015 at 11:10 AM, danilo...@tin.it  wrote:

>  Hi there,
> I'm working on this scenario to manage some toll free call features
>
> Caller =>OpenSips => Asterisk(IVR) => OpenSips => Called
>
> A Caller dials a number(A) configured on OpenSips, the call is forwarded
> to Asterisk where is active an IVR which let caller to choose which
> extension would like to reach. After Caller select through DTMF the
> extension, the call have to be sent back to OpenSips which can manage the
> choise and go on with the call to Called(B).
> As per my script, i Can reach Asterisk and make choice, but cannot give
> back to OpenSips the call control to go on
>
> Hope I've explained my needs
>
> Thx for your reply or to point me back to an existing scenario
>
> Danilo
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[OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL

[danilo...@tin.it]

 Hi there,
I'm working on this scenario to manage some toll free call features

Caller =>OpenSips => Asterisk(IVR) => OpenSips => Called

A
 Caller dials a number(A) configured on OpenSips, the call is forwarded 
to Asterisk where is active an IVR which let caller to choose which 
extension would like to reach. After Caller select through DTMF the 
extension, the call have to be sent back to OpenSips which can manage 
the choise and go on with the call to Called(B).
As per my script, i Can reach Asterisk and make choice, but cannot give back to 
OpenSips the call control to go on

Hope I've explained my needs

Thx for your reply or to point me back to an existing scenario

Danilo  


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