02-04-2015 05:03 AM - edited 03-17-2019 01:50 AM
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
02-04-2015 12:37 PM
Hi,
I am assuming that you`re abbreviating the translation profile command under the voice register pool 3 cause "translate-outgoing called 66" doesn't seem to be a valide IOS command.
For me your transaltion and alias are doing the same thing. Remove the translation-profile command from the voice register pool 3 and put the alias 2 3225016599 to 3225016588 under call-manager-fallback. Or create a dial-peer which has 3225016599 as destination-pattern and use the translation-profile under the same dial-peer to send the call to 3225016588.
MK
02-05-2015 08:52 AM
I've found it, we need to add:
voice register pool 1
alias 2 3225016599 to 3225016588
translation-profile outgoing to-sip-phone
Voice translation-rule 66
rule 2 /3225016599/ /3225016588/
Voice translation-profile to-sip-phone
translate called 66
that's it, bit different than cisco website.
07-08-2015 02:40 PM
Johan,
Thanks for this post!
It's been driving me mental trying to figure out how this is supposed to be configured for SIP SRST. It works so easily and well with SCCP SRST and there isn't really clear documentation stating how this is supposed to be configured, with vague references to translation rules being required for the alias command in SIP SRST (but not required for SCCP SRST, note!).
Also, the idea that the translation is required OUTBOUND from the perspective of the gwy TO the SIP phones is also 'different'.
Thank you for putting me out of my frustration on this, finally. I deserve a cuppa tea now.
Chris
05-12-2016 08:54 AM
I'm configuring the same functionality. Can I please see your entire config?
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