[Freeswitch-users] CISCO 2811 Freeswitch IVR

Fri Jan 6 18:21:00 MSK 2012

Are you still using ignore_early_media=true - this must be set for this to work correctly.

You will see a EXECUTE log line when FS executes the application, with ignore_early_media enabled it shouldn't execute until the call has been answered. I just tried it myself, and it works as expected.

Example "originate {ignore_early_media=true}sofia/internal/number at host &park()"

Park application is only executed after the call was answered.

/Peter

________________________________________
Från: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Oliver Schenk [olimonkey at gmail.com]
Skickat: den 6 januari 2012 12:04
Till: FreeSWITCH Users Help
Ämne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR

Because I'm using an FXO card with voice, I added something to my
CISCO conf. Many others had the same thing:

voice-port 0/3/0
   ...
   supervisory disconnect dualtone mid-call
   supervisory answer dualtone    <---- ADDED THIS ONE
   ...

Once I added this, the FS output now just showed the following while
the phone was ringing:

2012-01-05 16:19:31.644440 [NOTICE] switch_channel.c:816 New Channel
sofia/internal/109212xxxx at 192.168.x.x
[69e3f13d-1e2a-409e-97a4-b5526ea6e4ec]
2012-01-05 16:19:35.124882 [INFO] sofia.c:740
sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "Outbound
Call" <109212xxxx>
2012-01-05 16:19:35.126883 [NOTICE] sofia_glue.c:3793 Pre-Answer
sofia/internal/109212xxxx at 192.168.x.x!

Where as previous it would show the above and also show the following:

2012-01-05 16:19:35.127883 [INFO] switch_channel.c:2456
sofia/internal/109212xxxx at 192.168.x.x Flipping CID from ""
<0000000000> to "Outbound Call" <109212xxxx>
2012-01-05 16:19:35.137384 [INFO] sofia.c:740
sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "109212xxxx"
<1092122856>
2012-01-05 16:19:35.138384 [NOTICE] sofia.c:5296 Channel
[sofia/internal/109212xxxx at 192.168.x.x] has been answered

BUT, the IVR still started playing even before I pick up the phone.
Hmmmm.....so why is FS still starting the managed application when the
call has not been answered yet. Are we all sure that the managed
application should not be executed until the call "has been answered"
shows up in the log file?

Will have to keep testing on monday as I don't have access to the
CISCO from where i am now. I'll have to see whether the CISCO changes
had any impact on the times at which the SIP messages are sent back
and forth. Especially the 200 OK message.

Thanks again for help, maybe getting somewhere now......

Oliver

On Fri, Jan 6, 2012 at 4:20 PM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> If it sends 200 OK right after 183, this IS the problem.
>
> 200 OK means that the call was answered, it should not be sent until the call was actually picked up in the remote end. When 200 OK arrives to FS it will execute your app, and you will start playing the files.
>
> Seems to me there is something broken in the Cisco.
>
> /Peter
>
> ________________________________________
> Från: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Oliver Schenk [olimonkey at gmail.com]
> Skickat: den 6 januari 2012 06:55
> Till: FreeSWITCH Users Help
> Ämne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR
>
> I've tried looking at disable-early-media configuration command, but
> that didn't work and I doubt that has anything to do with the CISCO
> sending a 200 OK right after a 183 SESSION PROGRESS.
>
>
>
>
> On Fri, Jan 6, 2012 at 9:20 AM, Brian West <brian at freeswitch.org> wrote:
>> 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
>>
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>>
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>>
>>
>> _________________________________________________________________________
>> 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
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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
>>
>>
>> _________________________________________________________________________
>> 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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>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
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>
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>
>
>
>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
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>
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