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CUCM and Cube. Some calls drop after 30 seconds

ldallapalma
Level 1
Level 1

Just for some phone calls (not all of them), my system composed by CUCM and CUBE drop calls after several unanswered UPDATE SDP (g711A telephone-event) originated by the provider. Please help !!

SDP.png

5 Replies 5

b.winter
VIP
VIP

Can you post the debug and the config of the CUBE?
Which IP is which system in your screenshot?

Hello, thank you for the answer.
10.196.42.152 = Cisco Cube version 15.7
10.196.42.126 = Provider

Here is an extract of the Cube voice service voip
ip address trusted list
ipv4 10.196.42.126
ipv4 10.196.42.100
ipv4 10.196.41.10
ipv4 10.196.41.11
ipv4 84.14.87.169
ipv4 84.14.87.164
ipv4 84.14.87.165
ipv4 10.196.41.13
ipv4 10.196.41.15
ipv4 10.196.41.50
ipv4 10.196.41.51
ipv4 10.196.42.155
ipv4 10.196.42.156
ipv4 217.147.159.210
ipv4 217.147.159.209
ipv4 217.147.159.211
address-hiding
mode border-element license capacity 500
allow-connections sip to sip
no supplementary-service sip handle-replaces
redirect ip2ip
signaling forward unconditional
fax protocol t38 version 3 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
sip
session refresh
header-passing
asserted-id pai
history-info
midcall-signaling passthru
privacy-policy passthru
g729 annexb-all
sip-profiles 100
!


dial-peer voice 1 voip
description ** INCOMING FROM PSTN TOPIX**
session protocol sipv2
incoming uri via 1
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
codec transparent
no vad"

"dial-peer voice 2 voip
description ** OUTGOING TO PSTN VIA TOPIX **
translation-profile outgoing PSTN-OUT

destination-pattern .T
session protocol sipv2
session target ipv4:10.196.42.126
session transport udp
voice-class codec 7
voice-class sip asserted-id pai
voice-class sip options-ping 60
voice-class sip early-offer forced
voice-class sip profiles 100
voice-class sip options-keepalive
voice-class sip pass-thru headers 100
voice-class sip copy-list 100
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1

 

dtmf-relay rtp-nte
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
no vad"

"dspfarm profile 20 transcode
codec g729br8
codec g729r8
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 10
associate application CUBE"
"dspfarm profile 21 transcode


codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 80
associate application SCCP"
"dspfarm profile 1 mtp
codec pass-through
codec g711ulaw
maximum sessions software 10
associate application SCCP"
"dspfarm profile 10 mtp
codec pass-through
codec g711ulaw
maximum sessions software 10
associate application SCCP"
"sccp ccm group 1
bind interface GigabitEthernet0/0
associate ccm 1 priority 1
associate ccm 2 priority 2
associate ccm 3 priority 3
associate profile 10 register MTP_CUBE2_G711
associate profile 1 register MTP_CUBE2_G729
associate profile 21 register xcd_invacube01"
"voice-card 0
dsp services dspfarm

!"
"dial-peer voice 200 voip
description ** 200 TO/FROM PUB-SUB **
huntstop
max-conn 6
session protocol sipv2
session transport tcp
session server-group 1
destination e164-pattern-map 200
incoming calling e164-pattern-map 200
voice-class codec 7
voice-class sip asserted-id pai
voice-class sip options-ping 60
voice-class sip options-keepalive
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte
no fax-relay sg3-to-g3
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none"
"voice translation-rule 10
rule 1 /^00/ /+\1/
rule 2 /^0/ /+390\1/
rule 3 // /+39\1/"
voice translation-profile PSTN-OUT
translate called 10

Why do people always just post chunks of the config and then expect to get help ... anyway

Where is the debug output?
Have you asked the provider, why he is ending the call? If you say that the IP 10.196.42.126 is provider, then he is the one sending the "BYE" and therefore ending the call.

Because unluckily at the moment our Cisco technical partner is not supporting us, so we are alone with the problem. Please let me know which Cube debug command I should use. We have a Cisco Cube just for laboratory purposes, so any debug setting will not be service-affecting. Thank you.

debug ccsip messages and debug voip ccapi inout is the most common ones to us. Please make sure you have them enabled in parallel.



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