[Freeswitch-users] CISCO 2811 Freeswitch IVR
Oliver Schenk
olimonkey at gmail.com
Fri Jan 6 04:13:12 MSK 2012
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
>>>>>
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>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>>
>>>>
>>>>
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>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
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