[Freeswitch-users] CISCO 2811 Freeswitch IVR

Brian West brian at freeswitch.org
Fri Jan 6 04:20:28 MSK 2012


Thats what the 183 is.. 180 vs 183 are kinda sketchy in some devices.. 180 is usually RINGING (generate ringback locally) while a 183 has media... aka early media and usually provides ringback inband.

/b

On Jan 5, 2012, at 7:13 PM, Oliver Schenk wrote:

> Shouldn't there be a  180 RINGING  somewhere in there?
> 
> 
> 
> 
> On Fri, Jan 6, 2012 at 8:25 AM, Oliver Schenk <olimonkey at gmail.com> wrote:
>> I just noticed something else, if I don't pick up the phone at all.
>> The IVR just keeps playing until the menu timeout kicks in.
>> 
>> So here is a CISCO SIP log:
>> http://pastebin.com/Y9sYkuxi
>> 
>> The FS server is 192.168.x.50 and the CISCO is 192.168.x.1.
>> I hope the CISCO log is readable, it's a bit long because I just did
>> "debug ccsip all".
>> 
>> 
>> 
>> In this test I didn't bother picking up the phone at all, but I can
>> see that FS answered anyway and the IVR kept playing until it timed
>> out.
>> I'm not an expert, but here is what I picked out of it:
>> 
>> At 00:08:10 we get a
>> Received: "INVITE sip:109212xxxx at 192.168.x.1 SIP/2.0"
>> 
>> the further down at the same timestamp we get
>> Sent: "SIP/2.0 100 Trying"
>> 
>> At 00:08:13 we get a
>> Sent: "SIP/2.0 183 Session Progress"
>> 
>> At 00:18:13 we get a
>> Sent: "SIP/2.0 200 OK"
>> 
>> Then at the same timestamp we get:
>> Received: "ACK sip:109212xxxx at 192.168.x.1:5060 SIP/2.0"
>> 
>> 
>> 
>> Once the IVR times out at 00:09:16 we get
>> Received: "BYE sip:109212xxxx at 192.168.x.1:5060 SIP/2.0"
>> 
>> And then the reply right after
>> Sent: "SIP/2.0 200 OK"
>> 
>> 
>> 
>> So I think you were right, the CISCO is sending back an "OK" 3 seconds
>> after the "INVITE" is received.
>> 
>> 
>> 
>> The part that is beyond my field of expertise so far is WHY?
>> 
>> 
>> 
>> Thanks,
>> 
>> 
>> Oliver
>> 
>> 
>> 
>> On Fri, Jan 6, 2012 at 8:04 AM, Oliver Schenk <olimonkey at gmail.com> wrote:
>>> By the way:
>>> 
>>> I tried {ignore_early_media=true} as well, but as I think we
>>> determined, my problem is probably with the CISCO telling FS that the
>>> call has been answered when really it hasn't yet.
>>> 
>>> 
>>> 
>>> On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk <olimonkey at gmail.com> wrote:
>>>> Thanks for the help so far.
>>>> 
>>>> 
>>>> Here is a pastebin of FreeSWITCH output:
>>>> http://pastebin.com/i6Qgc7ws
>>>> 
>>>> Notice how the "has been answered" log message comes immediately
>>>> (within a few milliseconds) after the call was originated. I think
>>>> this would suggest that the CISCO is immediately sending a 200 OK, as
>>>> you suggested. I also turned on CISCO debugging, but I'm just trying
>>>> to figure out how to get the information regarding SIP messages back
>>>> to Freeswitch. I'll run the test again and see if I can get some
>>>> useful CISCO debug.
>>>> 
>>>> Which "debug ccsip" commands are relevant to what I want for the CISCO
>>>> SIP debugging?
>>>> 
>>>> 
>>>> Thanks!
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 2012/1/6 Gustavo Mársico <gustavomarsico at gmail.com>:
>>>>> I think I've a similar problem related to callcenter app. When I made an originate like this:
>>>>> 
>>>>> originate loopback/2500/default/XML &bridge(user/2001)
>>>>> 
>>>>> 2500 is an extension that leads to a callcenter application. In this case, we dial first to the queue and when an agent answered we call to the customer. As far as I know
>>>>> When the A-leg reaches to the queue, without selecting an agent, the call is automatically sent to the B-leg. As far as I see, there is a pre-answer method that fs needs to send the media to A-leg.
>>>>> In order to try to avoid this, I tried using ignore_early_media=true as part of the originate in A-leg and/or B-leg, with no luck.
>>>>> 
>>>>> originate {ignore_early_media=true}loopback/2500/default/XML &bridge({ignore_early_media=true}user/2001)
>>>>> 
>>>>> Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] destination_number(2500) =~ /^(2500)$/ break=on-false
>>>>> Dialplan: loopback/2500-b Action set(ignore_early_media=true)
>>>>> Dialplan: loopback/2500-b Action callcenter(click2call)
>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE
>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send signal loopback/2500-b [BREAK]
>>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL
>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 (loopback/2500-b) State ROUTING going to sleep
>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 (loopback/2500-b) Running State Change CS_EXECUTE
>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 (loopback/2500-b) State EXECUTE
>>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b CHANNEL EXECUTE
>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 loopback/2500-b Standard EXECUTE
>>>>> EXECUTE loopback/2500-b set(open=true)
>>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [open]=[true]
>>>>> EXECUTE loopback/2500-b hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb)
>>>>> EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/0000000000/2500)
>>>>> EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb)
>>>>> EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 -0300)
>>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300]
>>>>> EXECUTE loopback/2500-b set(ignore_early_media=true)
>>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [ignore_early_media]=[true]
>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 Application callcenter Requires media! pre_answering channel loopback/2500-b
>>>>> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer loopback/2500-a!
>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-a) Callstate Change RINGING -> EARLY
>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal loopback/2500-b [BREAK]
>>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL
>>>>> 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 Pre-Answer loopback/2500-b!
>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-b) Callstate Change RINGING -> EARLY
>>>>> EXECUTE loopback/2500-b callcenter(click2call)
>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-a) Callstate Change EARLY -> ACTIVE
>>>>> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel [loopback/2500-a] has been answered
>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal loopback/2500-b [BREAK]
>>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL
>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 Originate Resulted in Success: [loopback/2500-a]
>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-b) Callstate Change EARLY -> ACTIVE
>>>>> 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 loopback/2500-a Flipping CID from "" <0000000000> to "Outbound Call" <XML>
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote:
>>>>> 
>>>>>> 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
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
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-- 
Brian West 
FreeSWITCH Solutions, LLC
Phone: +1 (918) 420-9266 
Fax:   +1 (918) 420-9267
brian at freeswitch.org
http://www.freeswitch.org

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