Hello, dear support team.
This is a jabber calling -> CUCM 11.5su3 -> SIP-> 2921 154-3.M -> E1 -> PSTN to Cell Phone.
MTP are registered at the old CUCM I migrated from. However, I have checked - they are not used during calls (only for E1 time-slots).
I am trying to configure outbound calls without MTP. It worked correctly.
However, after no sccp command ( I only guess this might be a problem) CUCM to GW started to drop some calls.
I have two traces (Called Number doesn't matter in this case, because both calls are using the same CUCM rules and GW dial-peers to in/out for these two calls).
Each of the 2nd or 3d call is dropped.
What may be a problem of failed call ?
voice service voip
allow-connections sip to sip
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 2 hs-redundancy 0 fallback none
no fax-relay sg3-to-g3
modem passthrough nse codec g711alaw
sip
no update-callerid
Incoming dial-peer:
voice class uri 4051 sip
host 10.98.197.138
voice class codec 4051
codec preference 1 g711alaw
codec preference 2 g711ulaw
dial-peer voice 4051 voip
description -= Incoming from CUCM =-
session protocol sipv2
incoming uri via 4051
voice-class codec 4051
dtmf-relay cisco-rtp rtp-nte
fax rate 14400
fax protocol t38 version 0 ls-redundancy 2 hs-redundancy 0 fallback pass-through g711alaw
no vad
Outgoing dial-peer:
voice translation-rule 30501
rule 1 /^9834936\(.....\)/ /\1/ type any subscriber plan any isdn
rule 2 /^9/ //
voice translation-rule 30502
rule 60 /^4990$/ /3634460/
rule 100 /.*/ /3634455/
voice translation-profile TP3050OUT
translate calling 30502
translate called 30501
dial-peer voice 3050 pots
description -= OUT PSTN =-
translation-profile outgoing TP3050OUT
destination-pattern ^9T
progress_ind alert enable 8
direct-inward-dial
port 0/0/1:15
forward-digits all
no sip-register
