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Problem with DTMF

I am trying to establish link between A and B subscribers calling from

GSM network through the Call Controller installed in IP enviroment.

In below scenario, subscriber A makes the call to number 123456789. After this,

Call Controler sends INVITE requests to the subscriber A and Cisco Gateway ( 188888888 )

to establish voice path between them. Everythig is fine in this step, because A subscriber

is able to hear a specific prompt.

However, Call Controler sends in turn new INVITE requests to the subscriber B and

Cisco Gateway to establish voice path between them. After subscriber B is connected to

the gateway and prompt is played, when B is asked for putting DTMF code to agree to

be charged for connection with subscriber A. Unforunately, something is happening wrong,

because error is being returned during the DTMF attempt. It looks like bad stream would be read then and

the VXML script is stoped after this.

Please let me know what is wrong, if there is a possibility to catch DTMF string

only one way, when the call is originated from GSN network. However when the call is originated

from gateway and the subscriber needs to pick up call and put DTMF code, the vxml script is broken.

Please notice that a specific VoIP dial peer is configured for connecting to mobile network.

Hope, the catched log and basic script will are helpfull to find out the problem.

*Jan 30 01:48:43.746: //3541/3043089791F8/VXML:/vxml_digit_collection_done:

digits ()

*Jan 30 01:48:43.746: //3541/3043089791F8/VXML:/vxml_digit_collection_done:

name ()

*Jan 30 01:48:43.746: //3541/3043089791F8/VXML:/vxml_bind_lastprompt:

*Jan 30 01:48:43.746: //3541/3043089791F8/VXML:/vxml_digit_collect_process:

vxmlp 66680A90 status 2 async_status 60000

*Jan 30 01:48:43.746: //3541//AFW_:/vapp_session_exit_event_name: Exit Event vxml.session.error

*Jan 30 01:48:43.746: //3541/3043089791F8/VXML:/vxml_vapp_terminate:

vapp_status=2 ref_count 0

*Jan 30 01:48:43.746: //3541/3043089791F8/VXML:/vxml_vapp_terminate:

CALL_ERROR; http://xxx.yyy.zzz.aaa:10000/incomingCall.do?session.telephone.dnis=188888888&session.telephone.ani=1111

vxml session terminating with code=ERROR

vapp status=VAPP_FAIL vxml async status=VXML_ERROR_VAPP

<?xml version="1.0"?>

<vxml version="1.0">

<property name="bargein" value="true"/>

<form>

<property name="timeout" value="10s"/>

<field name="dtmfString" type="digits?length=1">

<prompt>

<audio src='http://xxx.yyy.zzz.aaa:10000/audio/press1or2_ulaw.wav'/>

</prompt>

<filled>

<submit next="questionPlayed.do;jsessionid=3nkdtmpv57b8r" namelist="dtmfString" />

</filled>

</field>

</form>

</vxml>

My dial-plan is the following:

########### Inband dial peer for the Multi-Party Call Control service #########

dial-peer voice 100 pots

service session

incoming called-number 123456789

direct-inward-dial

port 7/0:D

!

######## Call is being forwarded towards to Call Controler to manage connection and next redirected to target dial peer #################

dial-peer voice 2000 voip

service session

destination-pattern 123456789

session protocol sipv2

session target ipv4:aaa.bbb.ccc.ddd:5060

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

######## VoiceXML script is started to handle the call ################

dial-peer voice 1000 voip

service mpcc_test

session protocol sipv2

session target ipv4:aaa.bbb.ccc.ddd:5060

incoming called-number 188888888

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

########### Outband dial peer for Multi-Party Call Control service #########

dial-peer voice 101 pots

service session

destination-pattern 1........

no digit-strip

port 7/0:D

!

Thanks in advance,

1 REPLY
Silver

Re: Problem with DTMF

DTMF tones it sends an skiny packet to the ip phone or an h323 packet to the gw and the tones are generated locally. Because voice is streamed between the ip phones and gw directly if there would be a inbound dtmf ccm would not know about it as it is not involved in the stream.

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