Clear Channel Codec between Cisco GW and Patton Smartnode
Hi,
i have to establish an H.323 connection between an Cisco 3745 and an Patton
Smartnode 1200 with clear-channel codec to support ISDN-Data over VoIP.
PC with ISDN-Card---S0/PattonGW----H.323----3745/E1----PSTN
Software versions:
Cisco: SPServices 12.4(5a)
Patton SN1200: R3.20 2006-03-30 H323
The call establishment is sucessfull but i am not able pass any Data over
ISDN. Does anyone got this working?
Any help is welcome.
Regards,
Robert
Below are some Konfig sniplets:
Cisco:
controller E1 0/1/0
pri-group timeslots 1-31
interface Serial0/1/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn overlap-receiving
isdn incoming-voice voice
isdn send-alerting
isdn sending-complete
no cdp enable
dial-peer voice 10 pots
description ----- Towards PSTN -----
answer-address <removed>
destination-pattern 0T
progress_ind setup enable 3
fax rate disable
direct-inward-dial
port 0/1/0:15
dial-peer voice 473997246 voip
description ----- towards Patton GW -----
destination-pattern <removed>
progress_ind setup enable 3
modem passthrough nse codec g711alaw
session target ipv4:192.168.1.28
codec clear-channel bytes 80
ip qos dscp cs5 media
no vad
dial-peer voice 473997247 voip
description ----- Incoming Dial-Peer -----
modem passthrough nse codec g711alaw
incoming called-number <removed>
codec clear-channel bytes 80
ip qos dscp cs5 media
no vad
Patton:
profile voip VOIP
codec 1 transparent-cisco rx-length 20 tx-length 20
no high-pass-filter
no post-filter
no dtmf-relay
no dtmf-mute-encoder
context ip router
rtp-port-range 16384 32767
context cs switch
no digit-collection timeout
no address-completion timeout
interface h323 IF_H323
bind gateway GW_H323
route call dest-interface IF_S0_01
remote 192.168.254.30 1720
itc rx transparent
use profile voip VOIP
interface isdn IF_S0_01
route call dest-interface IF_H323
use profile tone-set Europe
no call-waiting
gateway h323 GW_H323
faststart
early-h245
h235-security version v2
no h235-security q931-auth-int
bind interface IF_IP_LAN router
no shutdown
use profile voip VOIP
port bri 0 1
clock auto
encapsulation q921
q921
protocol pmp
uni-side auto
encapsulation q931
q931
protocol dss1
uni-side net
encapsulation cc-isdn
bind interface IF_S0_01 switch
The SIP Media Inactivity Timer feature enables Cisco gateways to monitor and disconnect VoIP calls if no Real-Time Control Protocol (RTCP) packets are received within a configurable time period.
When RTCP reports are not received by a Cisco gateway, the SIP Media Inactivity Timer feature releases the hung session and its network resources in an orderly manner. These network resources include the gateway digital signal processor (DSP) and time-division multiplexing (TDM) channel resources that are utilized by the hung sessions. Because call signaling is sent to tear down the call, any stateful SIP proxies involved in the call are also notified to clear the state that they have associated with the hung session. The call is also cleared back through the TDM port so that any attached TDM switching equipment also clears its resources.
http://www.cisco.com/en/US/products/ps6441/prod_troubleshooting_guide09186a00801fa207.html#wp2077496
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