hello,
in callmanager I have 7861 sip phone configured with directory-number 3225016588.
in callmanager I have a hunt-pilot with directory-number 3225016599 with 7861 as member.
from pstn I can call both numbers and I arrive on the 7861, works fine.
------------------------------------------
I disconnect callmanager from network, and the 7861 registers on voice-gateway, I can still call 6588.
6599 is unreachable.
-------------------------------------------
config is
voice register global
timeouts interdigit 4
system message SRST | backup mode
max-dn 100
max-pool 40
!
voice register pool 3
id network 192.168.240.0 mask 255.255.255.0
preference 2
translate-outgoing called 66 normally this command should translate called number
alias 2 3225016599 to 3225016588
voice-class codec 10
no vad
!
voice translation-rule 66
rule 2 /3225016599/ /3225016588/
rule 4 /\(.*\)/ /3225016588/
!
--------------
show commands:
vgw#sh dial-peer voice summary
dial-peer hunt 0
AD PRE PASS OUT
TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT KEEPALIVE
1 pots up up 0T 1 up 0/0/0:15
2 voip up up 32250165.. 1 syst ipv4:10.10.10.10
110 pots up up 0 down
40001 voip up up 3225016588 2 syst ipv4:192.168.240.18:
40002 voip up up 3225016599 0 syst ipv4:192.168.240.18:
-------------------------
debug ccsip messages (when I call to 6599)
Sent:
INVITE sip:3225016599@192.168.240.18:5060 SIP/2.0 so you see wrong called number
Via: SIP/2.0/UDP 10.10.10.9:5060;branch=z9hG4bK13268A
Remote-Party-ID: <sip:3225016588@10.10.10.9>;party=calling;screen=yes;privacy=off
From: <sip:3225016588@10.10.10.9>;tag=679F4C-1218
To: <sip:3225016599@192.168.240.18>
Date: Tue, 03 Feb 2015 20:39:42 GMT
Call-ID: 9D8EA3B2-AB1B11E4-80BA9ED8-D977F7B0@10.10.10.9
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 2643130922-2870677988-2148859927-0250033008
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1422995982
Contact: <sip:3225016588@10.10.10.9:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 1237 2676 IN IP4 10.10.10.9
s=SIP Call
c=IN IP4 10.10.10.9
t=0 0
m=audio 16482 RTP/AVP 0 8 18
c=IN IP4 10.10.10.9
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
*Feb 3 20:39:42.395: //75/9D8AFA2A8015/SIP/Msg/ccsipDisplayMsg:
Received: it’s refused by the phone
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 10.10.10.9:5060;branch=z9hG4bK13268A
From: <sip:3225016588@10.10.10.9>;tag=679F4C-1218
To: <sip:3225016599@192.168.240.18>;tag=bc671c316fb0003a1a13ca06-61077078
Call-ID: 9D8EA3B2-AB1B11E4-80BA9ED8-D977F7B0@10.10.10.9
Date: Tue, 03 Feb 2015 20:39:42 GMT
CSeq: 101 INVITE
Server: Cisco-CP7861/10.2.1
Contact: <sip:0@10.10.10.9:5060>
Diversion: "" <sip:0@192.168.240.18>;reason=unconditional;privacy=off;screen=yes
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "" <sip:0@10.10.10.9>;party=called;id-type=subscriber;privacy=off;screen=yes
Allow-Events: kpml,dialog
Content-Length: 0
*Feb 3 20:39:42.419: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:3225016599@192.168.240.18:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.9:5060;branch=z9hG4bK13268A
From: <sip:3225016588@10.10.10.9>;tag=679F4C-1218
To: <sip:3225016599@192.168.240.18>;tag=bc671c316fb0003a1a13ca06-61077078
Date: Tue, 03 Feb 2015 20:39:42 GMT
Call-ID: 9D8EA3B2-AB1B11E4-80BA9ED8-D977F7B0@10.10.10.9
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
----------------------------------------
what do I need to change so the 7861 accepts this call to 6599 coming from isdn/pstn ???
kind regards,
johan C
