[Freeswitch-users] CISCO 2811 Freeswitch IVR

Oliver Schenk olimonkey at gmail.com
Fri Jan 6 03:25:08 MSK 2012 

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
>>>>
>>>> _________________________________________________________________________
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>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
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>>>
>>>
>>> _________________________________________________________________________
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>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
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