Re: [OpenSIPS-Users] Opensips IVR
I would like to use FreeSwithc as media server. Whiwh configurations do i have to put in place in both Opensips and Freeswitch? -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-IVR-tp7602372p7602426.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips IVR
Thaks for your reply. I 'll use Freeswitch and opensips in front of freeswitch -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-IVR-tp7602372p7602421.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips IVR
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 > > > > -- > View this message in context: > http://opensips-open-sip-server.1449251.n2.nabble.com/Opensips-IVR-tp7602372.html > Sent from the OpenSIPS - Users mailing list archive at Nabble.com. > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL
I will send you my few lines code...hope to help you Write you back asapIl 14/Mar/2015 18:11 mahan77 m...@sathees.co.uk 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 -- 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. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL
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 m...@sathees.co.uk 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. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL
Vlad, I think we could all benefit from the snippets if you know what I mean ;). T ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL
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. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL
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 m...@sathees.co.uk 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 http://opensips-open-sip-server.1449251.n2.nabble.com/OPENSIPS-IVR-CALL-CONTROL-tp7595634p7595887.html Sent from the OpenSIPS - Users mailing list archive http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL
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 http://opensips-open-sip-server.1449251.n2.nabble.com/OPENSIPS-IVR-CALL-CONTROL-tp7595634p7595887.html Sent from the OpenSIPS - Users mailing list archive http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL
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 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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