Cisco SPA504G - no incoming calls
i'm not able to set up a cisco spa50x device to receive incoming calls.
the device is connected directly to the internet (no nat/fw/proxy) with a dedicated IP, outbound calls work just fine, the inbound number is mapped correctly via skype manager, all ports are correct.
any hints?
thanks
Hello thenarban,
You have a very good device that has some good reviews. The first place to start is probably at www.Cisco.com/support. I'm sure they have had this same problem in their support archives to help you with. Skype hasn't seen this perticular problem before on this device.
One thing you might want to check is that if you’re trying to set up separate lines, you’ll have to change the line appearances to correspond with the different SIP registrations, or else the phone will register all 4 lines under the first SIP registration.
The second thing is to check that the Skype Online Number is configured in the set to appear on one specific line too. That may be the problem.
The third thing to check is, are you using this behind a PBX? If so, you need to configure that line in the PBX to target that extension.
Each of these items can be checked locally by you. (If anyone is using this same set, and had this same problem getting it setup, please chime in).
If you are still having problems, you can come into our Live Chat by logging into your Skype Manager and clicking on the Chat Button in the top right of that screen. We are here and available to you 24/7 for your support.
Thank You for using Skype and the Skype Community Forums.
Regards,
Victor S.
Regards,
Victor S.
Skype Enterprise Support
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Matthew.
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
logging message-counter syslog
no aaa new-model
clock timezone GMT 0
dot11 syslog
ip source-route
ip cef
no ip dhcp use vrf connected
ip dhcp excluded-address 192.168.1.1
ip dhcp excluded-address 10.10.10.1
ip dhcp pool DATA_POOL
network 10.10.10.0 255.255.255.0
default-router 10.10.10.1
dns-server 188.92.232.50 188.92.232.100
ip dhcp pool VOICE_POOL
network 192.168.1.0 255.255.255.0
default-router 192.168.1.1
dns-server 188.92.232.50 188.92.232.100
option 150 ip 192.168.1.1
ip name-server 188.92.232.50
ip name-server 188.92.232.100
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
bind control source-interface FastEthernet0/1.20
bind media source-interface FastEthernet0/1.20
registrar server
voice class codec 1
codec preference 2 g711ulaw
codec preference 3 g711alaw
voice register global
mode cme
source-address 192.168.1.1 port 5060
max-dn 144
max-pool 42
load 7960-7940 P0S3-8-12-00
authenticate register
tftp-path flash:
create profile sync 0008072514198272
voice register dn 1
number 6999
allow watch
name SIP
label SIP
voice register pool 1
id mac 000F.902B.40E0
type 7960
number 1 dn 1
dtmf-relay sip-notify
username cisco password cisco
codec g711ulaw
voice translation-rule 1
rule 1 /^9\(.*\)/ /\1/
voice translation-rule 2
rule 1 /^6...$/ /4143*002/
voice translation-profile DiscardDigit9
translate calling 2
translate called 1
voice translation-profile IncomingSIP
translate calling 1133501788
voice-card 0
no dspfarm
username matt privilege 15 secret 5 $1$DCD0$SjWqnKgDSGVzzIKRerXh11
archive
log config
hidekeys
interface FastEthernet0/0
ip address 194.12.0.222 255.255.255.252
ip nat outside
ip virtual-reassembly
duplex auto
speed auto
interface FastEthernet0/1
no ip address
ip nat inside
ip virtual-reassembly
duplex auto
speed auto
interface FastEthernet0/1.10
description DATA
encapsulation dot1Q 10
ip address 10.10.10.1 255.255.255.0
ip nat inside
ip virtual-reassembly
interface FastEthernet0/1.20
description VOICE
encapsulation dot1Q 20
ip address 192.168.1.1 255.255.255.0
ip nat inside
ip virtual-reassembly
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 194.12.0.221
ip http server
ip http authentication local
no ip http secure-server
ip nat inside source list 1 interface FastEthernet0/0 overload
access-list 1 permit 192.168.1.0 0.0.0.255
access-list 1 permit 10.10.10.0 0.0.0.255
tftp-server flash:P003-8-12-00.bin
tftp-server flash:P003-8-12-00.sbn
tftp-server flash:P0S3-8-12-00.loads
tftp-server flash:P0S3-8-12-00.sb2
tftp-server flash:P003-8-12-00
tftp-server flash:P003-8-12-00.loads
tftp-server flash:P003-8-12-00.sb2
tftp-server flash:SIP000F902B40E0.cnf.xml
control-plane
mgcp behavior g729-variants static-pt
dial-peer cor custom
dial-peer voice 2 voip
description Outgoing Geographic
translation-profile outgoing DiscardDigit9
destination-pattern 0[7]........
voice-class codec 1
session protocol sipv2
session target dns:sip.cloudcalling.co.uk
dtmf-relay rtp-nte
no vad
dial-peer voice 1 voip
description IncomingSIP
translation-profile incoming IncomingSIP
voice-class codec 1
session protocol sipv2
session target dns:sip.cloudcalling.co.uk
incoming called-number .T
dtmf-relay sip-notify rtp-nte
no vad
sip-ua
credentials username 4143*002 password 7 password realm sip.cloudcalling.co.uk
authentication username 4143*002 password 7 password
nat symmetric role passive
nat symmetric check-media-src
calling-info sip-to-pstn number set 4143*002
no remote-party-id
retry invite 3
retry register 3
timers connect 100
registrar dns:sip.cloudcalling.co.uk expires 60
sip-server dns:sip.cloudcalling.co.uk
host-registrar
gatekeeper
shutdown
telephony-service
load 7960-7940 P0S3-8-12-00
max-ephones 24
max-dn 30
ip source-address 192.168.1.1 port 2000
max-conferences 8 gain -6
web admin system name Admin secret 5 $1$Fktw$t9GQkdDdHmoYdwptO8.or.
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
line con 0
line aux 0
line vty 0 4
login
scheduler allocate 20000 1000
ntp server 85.119.80.232
end
Router#You my friend are a star! worked straight away, many thanks. Just one more thing, when i make an outgoing call, it always appears as "blocked" on my phone, my sip trunk is set to allow CME to alter outgoing CLI's how would i program the outgoing CLI to 01133501788 also?
The new working config is below with your suggestion, which works!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
logging message-counter syslog
no aaa new-model
clock timezone GMT 0
dot11 syslog
ip source-route
ip cef
no ip dhcp use vrf connected
ip dhcp excluded-address 192.168.1.1
ip dhcp excluded-address 10.10.10.1
ip dhcp pool DATA_POOL
network 10.10.10.0 255.255.255.0
default-router 10.10.10.1
dns-server 188.92.232.50 188.92.232.100
ip dhcp pool VOICE_POOL
network 192.168.1.0 255.255.255.0
default-router 192.168.1.1
dns-server 188.92.232.50 188.92.232.100
option 150 ip 192.168.1.1
ip name-server 188.92.232.50
ip name-server 188.92.232.100
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
registrar server
voice class codec 1
codec preference 2 g711ulaw
codec preference 3 g711alaw
voice register global
mode cme
source-address 192.168.1.1 port 5060
max-dn 144
max-pool 42
load 7960-7940 P0S3-8-12-00
authenticate register
tftp-path flash:
create profile sync 0015244443466064
voice register dn 1
number 6999
allow watch
name SIP
label SIP
voice register pool 1
id mac 000F.902B.40E0
type 7960
number 1 dn 1
dtmf-relay sip-notify
username cisco password cisco
codec g711ulaw
voice translation-rule 1
rule 1 /^6...$/ /4143*002/
voice translation-rule 3
rule 1 /^01133501788$/ /6999/
rule 2 /^1133501788$/ /6999/
voice translation-profile IncomingSIP
translate called 3
voice translation-profile Translatetrunk
translate calling 1
voice-card 0
no dspfarm
username matt privilege 15 secret 5 $1$DCD0$SjWqnKgDSGVzzIKRerXh11
archive
log config
hidekeys
interface FastEthernet0/0
ip address 194.12.0.222 255.255.255.252
ip nat outside
ip virtual-reassembly
duplex auto
speed auto
interface FastEthernet0/1
no ip address
ip nat inside
ip virtual-reassembly
duplex auto
speed auto
interface FastEthernet0/1.10
description DATA
encapsulation dot1Q 10
ip address 10.10.10.1 255.255.255.0
ip nat inside
ip virtual-reassembly
interface FastEthernet0/1.20
description VOICE
encapsulation dot1Q 20
ip address 192.168.1.1 255.255.255.0
ip nat inside
ip virtual-reassembly
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 194.12.0.221
ip http server
ip http authentication local
no ip http secure-server
ip nat inside source list 1 interface FastEthernet0/0 overload
access-list 1 permit 192.168.1.0 0.0.0.255
access-list 1 permit 10.10.10.0 0.0.0.255
tftp-server flash:P003-8-12-00.bin
tftp-server flash:P003-8-12-00.sbn
tftp-server flash:P0S3-8-12-00.loads
tftp-server flash:P0S3-8-12-00.sb2
tftp-server flash:P003-8-12-00
tftp-server flash:P003-8-12-00.loads
tftp-server flash:P003-8-12-00.sb2
tftp-server flash:SIP000F902B40E0.cnf.xml
control-plane
mgcp behavior g729-variants static-pt
dial-peer cor custom
dial-peer voice 1 voip
description IncomingSIP
translation-profile incoming IncomingSIP
voice-class codec 1
session protocol sipv2
session target sip-server
incoming called-number .T
dtmf-relay sip-notify rtp-nte
no vad
dial-peer voice 2 voip
description Outgoing Geographic
translation-profile outgoing Translatetrunk
destination-pattern 0[7]........
voice-class codec 1
session protocol sipv2
session target dns:sip.cloudcalling.co.uk
dtmf-relay rtp-nte
no vad
sip-ua
credentials username 4143*002 password 7 password realm sip.cloudcalling.co.uk
authentication username 4143*002 password 7 password
nat symmetric role passive
nat symmetric check-media-src
calling-info sip-to-pstn number set 4143*002
no remote-party-id
retry invite 3
retry register 3
timers connect 100
registrar dns:sip.cloudcalling.co.uk expires 60
sip-server dns:sip.cloudcalling.co.uk
host-registrar
gatekeeper
shutdown
telephony-service
load 7960-7940 P0S3-8-12-00
max-ephones 24
max-dn 30
ip source-address 192.168.1.1 port 2000
max-conferences 8 gain -6
web admin system name Admin secret 5 $1$Fktw$t9GQkdDdHmoYdwptO8.or.
transfer-system full-consult
create cnf-files version-stamp 7960 Dec 17 2013 14:35:13
line con 0
line aux 0
line vty 0 4
login
scheduler allocate 20000 1000
ntp server 85.119.80.232
end
Router# -
SIP incoming call, won't work (CME)
Hi all,
I'm facing a weird problem and the sip-provider can't help. I suppose there is a problem with the dial-peer/translation-rule but I can't figure it out...
There is a CME (c2800nm-ipvoice-mz.124-11.XW10.bin, CME Version 4.2(0)) with a
SIP trunk.
Outgoing calls are working (DID).
Incoming calls (all DID) are ringing on the same internal number.
The situation:
- external call on 0815440097 is ringing on the internal nr. 296 (should be 297)
- external call on 0815440096 is ringing on the internal nr. 296
Here the config:
================================
voice service voip
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
no update-callerid
voice translation-rule 40
rule 2 /\(.*\)/ /9\1/
voice translation-rule 190
rule 1 /^0\(.*\)/ /\1/
rule 2 /^9\(.*\)/ /\1/
voice translation-rule 191
rule 2 /296/ /0815440096/
rule 3 /297/ /0815440097/
voice translation-rule 192
rule 2 /^0815440097/ /297/
rule 3 /^0815440096/ /296/
voice translation-profile TP_IN_SIP
translate calling 40
translate called 192
voice translation-profile TP_OUT_SIP
translate calling 191
translate called 190
dial-peer voice 2000 voip
description *** SIP-TRUNK (IN/OUT) ***
translation-profile incoming TP_IN_SIP
translation-profile outgoing TP_OUT_SIP
destination-pattern 0.T
b2bua
session protocol sipv2
session target dns:sip12.e-fon.ch
session transport udp
incoming called-number 0815440096
dtmf-relay rtp-nte
codec g711alaw
no vad
sip-ua
credentials username 0815440096 password 7 xxxx realm sip12.e-fon.ch
keepalive target dns:sip12.e-fon.ch
authentication username 0815440096 password 7 xxxx
calling-info pstn-to-sip from number set 0815440096
no remote-party-id
retry invite 2
retry response 2
retry bye 2
retry register 2
retry options 1
registrar dns:sip12.e-fon.ch expires 69
sip-server dns:sip12.e-fon.ch
reason-header override
connection-reuse
host-registrar
sh sip-ua register status
Line peer expires(sec) registered
================================ ========== ============ ==========
0815440096 20005 18 yes
Here the CCSIP MESSAGE debug (looks ok):
(call from 0000000000 to 0815440097)
===============================
Mar 8 21:55:10.469 METD: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:212.55.198.132;lr=on;ftag=as00cd0e7f>
Via: SIP/2.0/UDP 212.55.198.132;branch=z9hG4bK49ff.2d35e30a71291ffe3895b39164900f36.0
Via: SIP/2.0/UDP 212.55.198.134:5061;branch=z9hG4bK1cb84749;rport=5061
Max-Forwards: 69
From: "0000000000" <sip:[email protected]:5061>;tag=as00cd0e7f
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5061>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: e-fon
Date: Thu, 08 Mar 2012 20:55:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-IPCONNECT: 0815440097
X-Number: 0815440097
Content-Type: application/sdp
Content-Length: 415
v=0
o=root 770254981 770254981 IN IP4 212.55.198.134
s=Asterisk PBX 1.6.1.20
c=IN IP4 212.55.198.134
t=0 0
m=audio 11886 RTP/AVP 8 9 111 3 18 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Mar 8 21:55:10.481 METD: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.55.198.132;branch=z9hG4bK49ff.2d35e30a71291ffe3895b39164900f36.0,
Via:SIP/2.0/UDP 212.55.198.134:5061;branch=z9hG4bK1cb84749;rport=5061
From: "0000000000" <sip:[email protected]:5061>;tag=as00cd0e7f
To: <sip:[email protected]:5060>
Date: Thu, 08 Mar 2012 20:55:10 GMT
Call-ID: [email protected]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
Here is the VOICE DIAL-PEER debug (call from 0000000000 to 0815440097):
=============================================
Mar 8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=0815440096, Called Number=0815440096, Peer Info
Type=DIALPEER_INFO_SPEECH
Mar 8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=0815440096
Mar 8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
Mar 8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=20005
2: Dial-peer Tag=2000
Mar 8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=0000000000, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
Mar 8 22:00:09.498 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=2000
Mar 8 22:00:09.502 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=0000000000, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
Mar 8 22:00:09.502 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=2000
Mar 8 22:00:09.502 METD: //-1/8647979A82E1/DPM/dpAssociateIncomingPeerCore:
Calling Number=0000000000, Called Number=0815440096, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
Mar 8 22:00:09.502 METD: //-1/8647979A82E1/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=2000
Mar 8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=0815440096, Called Number=0815440096, Peer Info
Type=DIALPEER_INFO_SPEECH
Mar 8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=0815440096
Mar 8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
Mar 8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=20006
2: Dial-peer Tag=2000
Mar 8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=0815440096, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
Mar 8 22:00:09.510 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ANSWER; Incoming Dial-peer=2000
Mar 8 22:00:09.514 METD: //-1/8647979A82E1/DPM/dpMatchPeersCore:
Calling Number=, Called Number=0815440096, Peer Info
Type=DIALPEER_INFO_SPEECH
Mar 8 22:00:09.514 METD: //-1/8647979A82E1/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=0815440096
Mar 8 22:00:09.514 METD: //-1/8647979A82E1/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
Mar 8 22:00:09.514 METD: //-1/8647979A82E1/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=20006
2: Dial-peer Tag=2000
show 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
555 voip up up 555 0 syst loopback:rtp
20001 pots up up 296$ 0 50/0/1
20002 pots up up 297$ 0 50/0/2
2000 voip up up 0.T 0 syst dns:sip12.e-fon.ch
20005 pots up up 0815440096$ 0 50/0/150
20006 pots up up 0815440097$ 9 50/0/2
voip translation debugging (call from 0794142975 to 0815440097):
=========================================
Mar 8 22:35:26.145 METD: //-1/73E51DB2834F/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x46FBFCA0; count=1
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x46FBFCA0; count=0
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: number=0794142975 type=unknown plan=unknown numbertype=calling
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_match_internal: Matched with rule 2 in ruleset 40
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_match_internal: Matched with rule 2 in ruleset 40
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/sed_subst: Successful substitution; pattern=0794142975 matchPattern=(.*) replacePattern=9\1 replaced pattern=90794142975
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_subst_num_type: Match Type = none, Replace Type = none Input Type = unknown
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_subst_num_plan: Match Plan = none, Replace Plan = none Input Plan = unknown
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: xlt_number=90794142975 xlt_type=unknown xlt_plan=unknown
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: number= type=unknown plan=unknown numbertype=redirect-called
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_get_RegXrule: Invalid translation ruleset tag=0
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_match_internal: Error: ruleset for redirect-called number not found
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: No match: number= type=unknown plan=unknown
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: number=0815440096 type=unknown plan=unknown numbertype=called
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_match: No match; number=0815440096 rule precedence=2
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_match: No match; number=0815440096 rule precedence=3
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_match_internal: No match found
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_profile_translate_internal: No match: number=0815440096 type=unknown plan=unknown
Mar 8 22:35:26.149 METD: //-1/73E51DB2834F/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x46FBFCA0; count=1
Mar 8 22:35:26.153 METD: //-1/73E51DB2834F/RXRULE/regxrule_dp_translate: No profile found in peer 20005 for outgoing direction
Mar 8 22:35:26.153 METD: //-1/73E51DB2834F/RXRULE/regxrule_dp_translate: calling_number=90794142975 calling_octet=0x0
called_number=0815440096 called_octet=0x0
redirect_number= redirect_type=0 redirect_plan=0 redirect_PI=-1 redirect_SI=-1
Mar 8 22:35:26.181 METD: //-1/73E51DB2834F/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x46FBFCA0; count=2
Thanks,
NorbertHi Alex,
Thank you for the reply.
After changing the "incoming called-number" I got the same output.
The weird think is, why the dial-peer debug shows the 0815440096 number, despite the right "to: number" in the SIP-Message.
Is there a problem with the "voice service voip" or "sip-ua"?
on the voice translation debug I see:
Match Rule=DP_MATCH_TO_URI; URI=sip:0815440097
Match Rule=DP_MATCH_FROM_URI; URI=sip:0819262424
But I guess the translation rule is maching this one:
Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=0815440096
So how can the voice translation rule be set to map the entry DP_MATCH_TO_URI; URI=sip:0815440097
Thanks for the help.
Regards,
Norbert
voip translation debugging (call from 0819262424 to 0815440097):
===================================================
Mar 9 07:45:16.371 METD: //-1/439ABF97847F/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x46FBFAFC; count=1
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x46FBFAFC; count=0
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: number=0819262424 type=unknown plan=unknown numbertype=calling
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_match_internal: Matched with rule 2 in ruleset 40
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_match_internal: Matched with rule 2 in ruleset 40
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/sed_subst: Successful substitution; pattern=0819262424 matchPattern=(.*) replacePattern=9\1 replaced pattern=90819262424
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_subst_num_type: Match Type = none, Replace Type = none Input Type = unknown
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_subst_num_plan: Match Plan = none, Replace Plan = none Input Plan = unknown
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: xlt_number=90819262424 xlt_type=unknown xlt_plan=unknown
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: number= type=unknown plan=unknown numbertype=redirect-called
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_get_RegXrule: Invalid translation ruleset tag=0
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_match_internal: Error: ruleset for redirect-called number not found
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: No match: number= type=unknown plan=unknown
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: number=0815440096 type=unknown plan=unknown numbertype=called
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_match: No match; number=0815440096 rule precedence=2
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_match_internal: Matched with rule 3 in ruleset 192
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_match: No match; number=0815440096 rule precedence=2
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_match_internal: Matched with rule 3 in ruleset 192
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_match: No match; number=0815440096 rule precedence=2
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/sed_subst: Successful substitution; pattern=0815440096 matchPattern=^0815440096 replacePattern=296 replaced pattern=296
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_subst_num_type: Match Type = none, Replace Type = none Input Type = unknown
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_subst_num_plan: Match Plan = none, Replace Plan = none Input Plan = unknown
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_profile_translate_internal: xlt_number=296 xlt_type=unknown xlt_plan=unknown
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x46FBFAFC; count=1
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_dp_translate: No profile found in peer 20001 for outgoing direction
Mar 9 07:45:16.375 METD: //-1/439ABF97847F/RXRULE/regxrule_dp_translate: calling_number=90819262424 calling_octet=0x0
called_number=296 called_octet=0x0
redirect_number= redirect_type=0 redirect_plan=0 redirect_PI=-1 redirect_SI=-1
Mar 9 07:45:16.379 METD: //-1/439ABF97847F/RXRULE/regxrule_vp_translate: No profile found in voice port or trunk group for outgoing direction
Mar 9 07:45:16.379 METD: //-1/439ABF97847F/RXRULE/regxrule_vp_translate: calling_number=90819262424 calling_octet=0x0
called_number=296 called_octet=0x0
redirect_number= redirect_type=0 redirect_plan=0
Mar 9 07:45:18.195 METD: //-1/439ABF97847F/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x46FBFAFC; count=2
debug voice dialpeer detail
=====================
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=0815440096, Expanded String=0815440096, Calling Number=0815440096T
Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Outgoing Dial-peer=2000 Is Matched
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Outgoing Dial-peer=20005 Is Matched
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ANSWER; Calling Number=0819262424
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=0819262424T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ORIGINATE; Calling Number=0819262424
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=0819262424T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.572 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Incoming Dial-peer=2000 Is Matched
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ANSWER; Calling Number=0819262424
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=0819262424T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ORIGINATE; Calling Number=0819262424
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=0819262424T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Incoming Dial-peer=2000 Is Matched
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_REQUEST_URI; URI=sip:[email protected]:5060
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_TO_URI; URI=sip:[email protected]:5060
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_FROM_URI; URI=sip:[email protected]:5061
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=0815440096
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Dial String=0815440096, Expanded String=0815440096, Calling Number=
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ANSWER; Calling Number=0819262424
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=0819262424T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ORIGINATE; Calling Number=0819262424
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=0819262424T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.576 METD: //-1/D8245087848D/DPM/MatchNextPeer:
Result=Success(0); Incoming Dial-peer=2000 Is Matched
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=296, Expanded String=296, Calling Number=296T
Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Outgoing Dial-peer=20001 Is Matched
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ANSWER; Calling Number=296
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=296T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ORIGINATE; Calling Number=296
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=296T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.584 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Incoming Dial-peer=20001 Is Matched
Mar 9 07:49:25.584 METD: //-1/D8245087848D/DPM/dpMatchCore:
Dial String=296, Expanded String=296, Calling Number=
Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.584 METD: //-1/D8245087848D/DPM/MatchNextPeer:
Result=Success(0); Outgoing Dial-peer=20001 Is Matched
Mar 9 07:49:25.588 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=90819262424, Expanded String=90819262424, Calling Number=
Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.588 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=0819262424, Expanded String=0819262424, Calling Number=
Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Outgoing Dial-peer=2000 Is Matched
Mar 9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ANSWER; Calling Number=296
Mar 9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=296T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.592 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ORIGINATE; Calling Number=296
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=296T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Incoming Dial-peer=20001 Is Matched
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=296, Expanded String=296, Calling Number=
Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Outgoing Dial-peer=20001 Is Matched
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=90819262424, Expanded String=90819262424, Calling Number=
Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=90819262424, Expanded String=90819262424, Calling Number=
Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result=-1
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=296, Expanded String=296, Calling Number=
Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Mar 9 07:49:25.596 METD: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Outgoing Dial-peer=20001 Is Matched -
ISDN E1 PRI incoming calls busy out after 2nd b channel picks up
I have an ISDN E1 PRI line coming in from the Telco. D Channel comes up nicely and B channels accept calls, but only the first two calls. Once two callers are online, any further incoming calls get a busy tone. ISDN Debug data reports:
**ERROR**: call_incoming: Remote Protocol Error: No b channel assigned by switch, call id 0xCE
Hardware I'm using? Cisco 2650, 12.3(7T) and NM-HDV, VWIC-2MFT-E1-DI
Anybody please help?Problem sorted - believe it or not, it was the telco. They changed their switch setting for the line to preferrable B channel - and voila!
Thanks,
Brian -
Matching B-channel for dialpeer assigment (Incoming call)
Is it possible to manipulate a specific B Channel to use a dialpeer? For example on a T1 with 24 channels, I need the last 4 B-channels to choose an specific dialpeer, same concepts as matching incoming DID's but on a B-Channel slot.
If a call arrives on B-channel 20, I need that call to be sent to a specific DN at the CallManager.
I am using H.323 on the gateway side.
thanks in advanced.
OscarHi!
I'm stuck in a similar situation at the moment, and hope someone has solved this issue.
I have a "back to back connection" between two PBX'es that communicate using Q.931. I have had to replace my old hardware running a legacy CCS solution due to its incapability of understanding overlap signalling. I have replaced it with two Cisco routers running E1 (Q931). Between the units I have a high latency low bandwidth network.
The issue is that the PBX'es are configured in a way that requires calls between the PBX'es to use the same time slot at both ends. I'm running a VoIP network with CUCM between the two sites.
=============================================
controller E1 0/2/0
framing NO-CRC4
pri-group timeslots 1-21
trunk-group TS01 timeslots 1
trunk-group TS02 timeslots 2
trunk-group TS03 timeslots 3
trunk-group TS04 timeslots 4
trunk-group TS05 timeslots 5
trunk-group TS06 timeslots 6
trunk-group TS07 timeslots 7
trunk-group TS08 timeslots 8
trunk-group TS09 timeslots 9
trunk-group TS10 timeslots 10
trunk-group TS11 timeslots 11
trunk-group TS12 timeslots 12
trunk-group TS13 timeslots 13
trunk-group TS14 timeslots 14
trunk-group TS15 timeslots 15
trunk-group TS17 timeslots 17
trunk-group TS18 timeslots 18
trunk-group TS19 timeslots 19
trunk-group TS20 timeslots 20
trunk-group TS21 timeslots 21
dial-peer voice 81030001 pots
trunkgroup TS01
description ** PBX TS01 **
translation-profile incoming 61031401
translation-profile outgoing 25
destination-pattern 81030001T
progress_ind alert enable 8
progress_ind progress enable 2
incoming called-number .
no digit-strip
dial-peer voice 81030002 pots
trunkgroup TS02
description ** PBX TS02 **
translation-profile incoming 61031402
translation-profile outgoing 25
destination-pattern 81030002T
progress_ind alert enable 8
progress_ind progress enable 2
incoming called-number .
no digit-strip
voice translation-rule 25
rule 1 /^810300../ //
voice translation-rule 61031401
rule 1 // /61031401\1/
voice translation-rule 61031402
rule 1 // /61031402\1/
voice translation-profile 25
translate called 25
voice translation-profile 61031401
translate called 61031401
voice translation-profile 61031402
translate called 61031402
=====================================
The idea is that I do not know (or care) what numbers are used as SOURCE or DESTINATION of the original call. My network should be transparent to the PBX number plan. I need to add a prefix, and it should be based on the timeslot the call comes in on. I route the traffic between the routers using the prefix.
The configuration excerpt above should add 61031401 prefix to all calls entering on TS01, and 61031402 to all calls entering on TS02 etc. Calls from the remote should have corresponding prefixes 81030001 for TS01 and 81030002 for TS02 etc.
The outbound (from voip to pots) routing of the above configuration works.
However I have a challenge with the incoming prefixing.
All calls inbound end up using "dial-peer 81030001 pots".
I believe the reason this dial-peer "takes" all of the calls inbound from pots is due to the line "incoming called-number ."
Removing this makes no inbound pots call work as the "destination-pattern 8103001T" is never matched.
Removing "destination-pattern 8103001T" from the dial-peer is not working as it kills the voip to pots routing of inbound calls from the remote router.
Anyone got a good idea for me? -
000049: Oct 26 14:51:04.181: %VOICE_IEC-3-GW: H323: Internal Error (H323 Interwo
rking Error): IEC=1.1.127.5.21.0 on callID 261 GUID=80316352D0FE011E02002302A5CC
B8F9
000050: Oct 26 15:24:50.315: %CALL_CONTROL-6-CALL_LOOP: The incoming call has a
global identfier already present in the list of currently handled calls. It is b
eing refused.
000051: Oct 26 15:24:50.315: %VOICE_IEC-3-GW: CCAPI: Internal Error (Incoming lo
op): IEC=1.1.180.1.28.0 on callID 0
000052: Oct 26 15:24:50.315: %VOICE_IEC-3-GW: H323: Internal Error (H323 Interwo
rking Error): IEC=1.1.127.5.21.0 on callID 363
According to Cisco it says:
It means that the voice gateway has detected a loop in the call route
What debugs can I do to track this problem down.
All dial peers are like this. They are all voip dial peers with different destination-patterns and different session target IP addresses. No PSTN dial peers at all.
dial-peer voice 30000 voip
huntstop
destination-pattern 3[01]...
session target ipv4:192.168.99.4
dtmf-relay h245-signal
ip qos dscp cs5 mediaQ: How long has this system been in place - is it a new system, or is this something that just started on a system that has been in place for awhile?
A: My customer says to me that this system has been in place for nearly a year and the problem started only a couple of days ago.
Q: If so were there any changes to the infrastructure, or the dialplan in particular?
A: Again this is the first question I asked and he said “NO CHANGES”. But that is what most customer say – right?
Q: Have you seen this occur in the past in the logs? If so how often do you see it?
A: Started on 26th Oct 2010. Since the problem started the router has been reloaded and the problem is still there. It is very intermittent and I have to do the debug when I get the problem.
Q: Have you tried to establish what is causing these: H323: Internal Error (H323 Interworking Error) ?
A: Not sure how to troubleshoot this. The explanation of this message on CCO is not that great. -
AC - change incoming call alert wav files
Hi,
With CCM4.1(3)sr3a I need to change the audible alerts that AC plays when an incoming call arrives.
I've found the source files on each attendants PC (C:\Program Files\Cisco\Call Manager Attendant Console\audio), copied in a replacement & renamed it to Cisco's original file name.
However, on startup AC is copying the original files back from CCM.
If I changed the files on CCM then this would solve my problem but I can't find them.
How can I change the audio file source ?
Any help appreciated,
Paul.The current capabilities and features of MoH include:
MoH multicast and unicast streaming service
Music streaming service for "user" hold and "network" hold
51 sources per media convergence server (MCS)
Fifty continuously looping .wav file sources
One real-time streaming source
Each source configurable as either unicast or multi-cast stream
Support for audio streaming to selected devices
Gateways (multicast only):
DT-24+No
6608Not until Seaview 3.3
VG200 (H.323)Yes with Cisco IOS. Software Release 12.2(11)T and later, enable the ccm-manager music-on-hold command
VG200 (MGCP)Yes, same as VG200 (H.323)
VG248Yes
Gateways (unicast only)AT-2/-4/-8, AS-2/-4/-8, and all other Cisco IOS. VoIP gateways, including Cisco 1750, 2600/3600, 5300, 58xx, and 72xx
Cisco IP phones (unicast, multicast)7910, 7940, and 7960
Cisco IP phones (unicast only)Cisco 7935, 12 SP+, and 30 VIP phones
Cisco IP SoftPhone (unicast only)
Maximum 250 simultaneous on-hold streaming sessions per server
Multiple server instances for application scalability
Multiple server instances for server load balancing and redundancy
G.711, G.729A, and wide-band audio codec support
Off-line audio translation utility
This URL should help you:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_qanda_item09186a0080094766.shtml
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