Hi forum members
I have the following scenario:
SIP PBX <--> UC540 <--> SIP trunk to service provider.
The UC540 has the phones with numbers 100 to 105 registered. The PBX has the 7XX numbers. To the PBX, the following dial-peer is used:
dial-peer voice 2 voip
description To MX-ONE
destination-pattern [2-7]..$
session protocol sipv2
session target ipv4:10.30.0.2
voice-class codec 1
dtmf-relay sip-notify
!
dial-peer voice 1 voip
description incoming from MX-ONE
session protocol sipv2
incoming called-number 10.
voice-class codec 1
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
Everything works for this dial-peer, until the ITSP SIP trunk, with the following configuration, goes up:
sip-ua
credentials username xxxxxx password 7 xxxxxxxx realm sip.elitevoip.net
authentication username xxxxxxxx password 7 xxxxxxx
sip-server ipv4:5.39.84.82
host-registrar
!
dial-peer voice 7 voip
description Internacional via SIP Trunk
translation-profile outgoing PSTN-OUT-INTERNACIONAL
destination-pattern 00.T
session protocol sipv2
session target sip-server
!
voice translation-profile PSTN-OUT-INTERNACIONAL
translate calling 3
translate called 5
!
voice translation-rule 3
rule 1 /^100/ /226434580/
rule 2 /^101/ /226434581/
rule 3 /^102/ /226434582/
rule 4 /^103/ /226434583/
rule 5 /^104/ /226434584/
rule 6 /^105/ /226434585/
!
voice translation-rule 5
rule 1 /^00/ //
Everything works perfect with the ITSP SIP trunk. It registers, and calls go out an in with no problems:
CMELDA001#sh sip-ua register status
Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
7143167791 -1 2834 yes
P.S.: the ephone-dn numbers are already with "no-reg" option, just to isolate them from registering in the sip-ua trunk.
THE PROBLEM is: when this sip-ua trunk goes up, I cant receive calls from the PBX. The call is placed, the phone on the UC540 rings, I answer but it continues ringing on the originating phone up to 20 seconds.
It starts when I put the "registrar" command. Issuing the "no registrar" under the sip-ua, I start receiving calls again from the PBX.
Do someone had a similar problem? I am attaching the debugs with the sip registrar (not working) and without (working).
Thanks all
