[Freeswitch-users] CISCO 2811 Freeswitch IVR

[Oliver Schenk]

Also, maybe I should be doing something like this:

sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)'

instead of:

sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)'


but, I don't really have the CISCO configured as a gateway, nor do I
know how really...probably not on the right track there.




On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk <olimonkey at gmail.com> wrote:
> *bump*
>
>
> So I think maybe the way I'm doing the originate is the problem? In my
> call string I'm creating a connection directly from the CISCO
> (192.168.x.x) to the managed application, which may be why it starts
> playing straight away?
>
> Maybe I should be originating a call first and then only once I know
> the other side has picked up will I bridge the call to the IVR managed
> application.
>
> Problem is I dunno how to tell whether the other person has picked up
> (or even if the cisco is going to tell me) and I don't know how to do
> things to a call once it has been established.
>
>
> I'm currently reading the Dialplan wiki page, hoping to get something
> out of it there.
>
>
> Cheers
>
> Oliver
>
>
> On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk <olimonkey at gmail.com> wrote:
>> I've been battling while creating an IVR using FreeSWITCH mod_managed
>> and connecting through a CISCO 2811. Most things now work quite well,
>> but I am having a few issues with the way the system answers calls (or
>> doesn't answer calls...).
>>
>> I have FreeSWITCH running as a windows service on Windows Server 2008,
>> which is connected via LAN to a CISCO 2811 with a 4 port FXO card,
>> which is then connected to a POTS phone line.
>>
>>
>> Take the following scenario:
>>
>> 1. Managed .NET application creates a call string and uses ESL to talk
>> to freeswitch and originate a call:
>>
>> string callstring =
>> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x
>> '&managed(ivrAppName)'";
>> eslConnection.API("originate", callstring);
>>
>> where 192.168.x.x is the CISCO IP.
>>
>> 2. The CISCO sees that the phone number (1091234567) starts with a 1
>> so it uses FXO port 1 and strips the 1 and uses the remaining phone
>> number (091234567) to make the call.
>>
>> 3. My phone rings, I pick up and I can hear my IVR playing.
>>
>>
>>
>> These are my current problems:
>>
>> - IVR starts playing before I even pick up the phone. This means that
>> if the system calls a mobile phone and the person doesn't pick up, the
>> IVR will start playing and eventually the mobile phone will divert to
>> voice mail. Obviously I then get a missed call and an sms saying I
>> have a new voice mail, which is annoying. Instead I would like it to
>> KNOW that no one has picked up, but I don't know how to do this.
>> Somehow the CISCO needs to be able to tell FreeSWITCH that the call
>> has not yet been answered. For some reason however as soon as the
>> CISCO starts calling FreeSWITCH thinks the call is already connected.
>> It doesn't know that the CISCO is actually still ringing. Maybe I'm
>> doing originate the wrong way or something ...
>>
>> - The phone only rings for about 10 seconds before hanging up. I've
>> tried "call_timeout", "bridge_answer_timeout". I've also tried setting
>> CISCO "ring number". Nothing works, my phone still only rings for
>> about 10 seconds. I don't know if this is a FreeSWITCH issue or a
>> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just
>> starts playing even if no one answers the phone.
>>
>>
>>
>>
>>
>> CISCO Config for relevant FXO port:
>>
>> voice service voip
>>  allow-connections h323 to h323
>>  allow-connections h323 to sip
>>  allow-connections sip to h323
>>  allow-connections sip to sip
>>  no supplementary-service h450.2
>>  no supplementary-service h450.3
>>  supplementary-service h450.12
>>  no supplementary-service sip moved-temporarily
>>  no supplementary-service sip refer
>>  fax protocol cisco
>>  sip
>>  registrar server expires max 3600 min 3600
>>  no update-callerid
>>  no call service stop
>>
>> voice-port 0/3/2
>>  output attenuation -3
>>  no comfort-noise
>>  cptone AU
>>  impedance complex1
>>  caller-id enable
>> !
>> dial-peer voice 100 pots
>>  preference 1
>>  destination-pattern 1T
>>  port 0/3/2
>> !
>>
>>
>>
>> Many Thanks,
>>
>> Oliver