SIP ITSP on CUCM 10.5.2 (No CUBE) Incoming calls fail, outgoing are fine
Hi,
I am in the process of upgrading a customer who is on 8.0.3. They have an ITSP terminating SIP Trunk directly on the CCM Server
I upgraded the system to 10.5.2. During cutover I was able to make outgoing calls but all incoming calls were failing.
After reverting back to the old system, everything is working fine again, and I dont understand what could be the possible issue that it doesnt work on 10.5.2 but it works well on 8.0.3.
I checked almost everything and dont find anything that stands out, which may be contributing to the issue.
Any idea what could be missing here?
Thanks
Thanks for all your tips.
It was turned out that, the URI was a FQDN and during the first install of the 8.0.3 (in the sandbox) I had not bothered to get the DNS Services replicated and then didnt check if the ITSP was sending the invite on URI based on FQDN or IP Address
Thanks
Similar Messages
-
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 -
Third Party Phone over SIP Trunk with CUCM 9.x
Hi all,
I have a problem where my Third Party SIP phones wont go over the SIP trunk configured in my CUCM 9.x cluster. My Cisco phones work fine and goes out the trunk. I have noticed a distinct difference in wireshark with the invite packets from Third Party SIP phones and the Cisco ones.
I have configured the SIP trunk in CUCM with the following route pattern (60.!#)and configured it with associated group and list. Heres the differense between the invite packets from Cisco and Third Party phones.
Cisco Phone: INVITE sip.60xxxx%23@ipadress
Third Party SIP Phone: INVITE sip:[email protected]
It seems the Cisco phones gets some extra configured the Third Party ones dont...
Thanks in advance for any help.
//PerThanks for the answer
Yeah i have DNS configured and i have the trunk pointed to a domain destination SRV record and like i said it works fine when calling from a Cisco phone. I tried changing the domain to an ip address but same result. I also changed the Plycom phone from being registered towards the domain of CUCM to an IP adress of CUCM and then the SIP INVITE messages in wireshark began to look kinda the same expet for the "%23" section but it still dont work.
When i look at the Real Time Data in RTMT the orig and final called from the cisco phone has stripped the 60 and forwared the rest of the number towards the correct domain for the SIP trunk.
When looking at the data from the Polycom phone the orig and final called data still contains the 60 prefix part and the called device name field is empty. The termination Cause Code is that the number requested is Unallocated/Unassigned..
In other words something is missing to get CUCM to strip 60 from the Polycom phones dialed number and send it towards the SIP trunk like it does when the Cisco phones call it.
Unfortunatley i dont have the meens to attach the trace...
Thanks again for any help/advice
With regards, Per. -
3rd Party SIP phone to CUCM via SIP Proxy
Hi all,
This is the scenario i'm currently working on :
3rd party SIP phone <--> Internet <--> SIP Proxy <--> LAN <--> CUCM
The SIP proxy basically terminates everything (REGISTER, INVITE, etc), including the RTP stream.
I can register the 3rd party SIP phone to CUCM and in CUCM and i can see SIP Proxy IP Address as the registered address of the phone.
Calls from the 3rd party SIP phone to internal Cisco or internal 3rd party SIP phone and vice versa work like charm.
The only (fatal) problem is i can only register 1 3rd party SIP phone to CUCM via this SIP proxy.
Since this SIP Proxy always use its internal IP Address and port 5060 (TCP) as its source of registration, CUCM sees multiple registrations for multiple extensions (users) come from a single IP and port, and rejects the second registration request.
It seems that CUCM binds a digest user to an IP address and port, therefore cannot accept multiple registrations from a single IP and port.
Can anyone clarify this? Or is there any way around this?
I'm using CUCM 8.6.2 and CUCM 9.X (both do not work).
Regards,
ChristianThis is most likely because of the following...
Because third-party SIP phones do not send a MAC address, they must identify themselves by using digest authentication.
The REGISTER message includes the following header:
Authorization: Digest username="xxxxxxxxxx",realm="ccmsipline",nonce="GBauADss2qoWr6k9y3hGGVDAqnLfoLk5",uri="sip:172.18.197.224",algorithm=MD5,response="126c0643a4923359ab59d4f53494552e"
The username, xxxxxxxxxxx, must match an end user that is configured in the End User Configuration window of Cisco Unified CallManager Administration. The administrator configures the SIP third-party phone with the user; for example, swhite, in the Digest User field of Phone Configuration window.
See the following document.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/5_1_3/ccmcfg/b09sip3p.html
Also Try this bug CSCef88775 -
CP-6921-C-K9 as SIP phone on CUCM 6?
Can a CP-6921-C-K9 phone be used as a third party SIP device on CUCM 6?
We will be upgrading soon to a new version of CUCM where 6921 would be supported but for now can we get them to work on CUCM 6 as SIP phones?
Regards.It looks like the SIP support is even later in the 7.1.x line. No documented operation as 3rd party SIP that I can find.
http://www.cisco.com/c/en/us/products/collateral/collaboration-endpoints/unified-ip-phone-6921/data_sheet_c78-541534.html
The phones are supported in Cisco Unified Communications Manager and Cisco Unified Communications Manager Business Edition Versions 7.1.2 and later using Skinny Client Control Protocol (SCCP) or SIP with Cisco Unified Communications Manager and Cisco Unified Communications Manager Business Edition Versions 7.1.5 and later. -
How to create multiple sip trunks between cucm and cisco unified sip proxy
Dear Expert,
Is there a way to create multiple sip trunks between CUCM and Cisco Unified SIP Proxy (CUSP)? How to achieve it without creating multiple IP interfaces on the CUSP module.
CUCM: 8.5.1.10000-9
CUSP: 8.5.2
Thank you,
.wanHello Michael,
This SIP trunk is part of UCCE solution, which used between CVP, CUSP, and CUCM.
The requirements:
1) To have different codecs for different type of calls, as the phones are at few countries
2) To pass different number of digits from CUSP to CUCM for different call treatments
.wan -
Modify calling number in SIP invite on CUCM 10.5
Hello,
I am working at a customer with CUCM 10.5 who uses MGCP gateways to access the PSTN via T1 PRI ISDN.
They use four digit DNs internally and need to prefix these with 713657 to make the outbound CLID work ok - i.e. a call to the PSTN from extension 1000 needs to send 7136571000 to the ISDN provider.
I configured this using Calling Party Transformations and this works fine e.g.
A Calling Party Transformation for 1XXX would prefix 713657.
The problem I have is that the customer has a NICE active recording system which communicates with the CUCM cluster using a SIP trunk.
The invites that CUCM sends via the SIP trunk show the full ten digits rather than the four digit extension which will not work according to company deploying the recording system.
If I remove the Calling Party Transformation then the SIP invite shows four digits and the call recording works but the outbound CLID does not work.
Can anyone suggest a way to fix this? The customer does not want to change the gateway protocol from MGCP to H323 which would be my favoured choice. Any change of calling party setting on CUCM (e.g. ticking the use external mask for calling party on route pattern) affects the SIP invite.
Ideally I need a way to modify the number in the SIP invite but I cannot find any example of how to do this.
Any suggestions are welcome.
ThanksHi, thanks for your response.
The Calling Party Transformation CSS is not applied to the SIP trunk but is applied to the T1 port of the MGCP gateway.
The Transformations are still applied to the SIP Invite messages via the trunk so I guess this is a quirk of the calling recording profile setup on CUCM.
I did try creating another Calling Party Transformation setup which stripped the unwanted digits and applied it to the SIP trunk but it had no effect. -
SIP incoming call with G722-64 codec not working
Hi, Guys.
Have setup cube sip trunk to ITSP, incoming and outgoing calls are working. Except for an incoming call with g722 codec and video h263 (just need voice call). The called number does not even ring. The caller informed that his using polycom phone.
Also, itsp provided 10 numbers for testing in which we can assigned to our phones but only the main number is working. When doing an incoming call, (dialing the other numbers except from the main number) can see always on the logs that itsp is always feeding the main number. I think it was because of the configuration under the sip-ua (register the maint number to a registrar) but itsp informed that it was also their setup for other clients and is working. Appreciate your help on these.
ThanksI have looked at your logs and here are my observations..
1. When you disabled fast start on CUCM, I asked you to enable early offer on your CUBE, however I dont see this in your logs..
This is the INVITE sent to your ITSP, as you can see, this doesnt contain any SDP, that suggest you are doing delayed offer..
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.8.134:5060;branch=z9hG4bKC862F
From: <sip:[email protected]>;tag=AE7F464-1B0F
To: <sip:[email protected]>
Date: Thu, 03 Apr 2014 07:33:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,histinfo,sdp-anat
Min-SE: 1800
Cisco-Guid: 0011462194-3037647155-0083893506-2887478836
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M4
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1396510389
Contact: <sip:[email protected]:5060>
History-Info: <sip:[email protected]:5060>;index=1,<sip:[email protected]:5060>;index=2
Expires: 300
Allow-Events: telephone-event
Authorization: Digest username="AMM-4324-Trunk",realm="amcomvoice.ipsystems.com.au",uri="sip:[email protected]:5060",response="9555a4d29d9316d3f5d416f9a5096ee2",nonce="BroadWorksXhtjq88oeTdvuambBW",cnonce="6AFA84F5",qop=auth,algorithm=MD5,nc=00000001
Content-Length: 0
2. If you are doing DO, then your CUBE needs to send an answer to what your ITSP is offering in its ACK..but this is not happening
Here is what I see..Your CUBE sends SDP in its PRACK
Sent:
PRACK sip:[email protected]2.147.134.21:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.21.8.134:5060;branch=z9hG4bKC9232B
From: <sip:[email protected]>;tag=AE7F464-1B0F
To: <sip:[email protected]>;tag=SD7qfu599-1874793413-1396510390487
Date: Thu, 03 Apr 2014 07:33:09 GMT
Call-ID: [email protected]
CSeq: 103 PRACK
RAck: 323009643 102 INVITE
Allow-Events: telephone-event
Authorization: Digest username="AMM-4324-Trunk",realm="amcomvoice.ipsystems.com.au",uri="sip:[email protected]2.147.134.21:5060;transport=udp",response="a9d772d988ec971cdad556fd4a992bd0",nonce="BroadWorksXhtjq88oeTdvuambBW",cnonce="58621262",qop=auth,algorithm=MD5,nc=00000002
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 293
v=0
o=CiscoSystemsSIP-GW-UserAgent 1079 7198 IN IP4 172.21.8.134
s=SIP Call
c=IN IP4 172.21.8.134
t=0 0
m=audio 17082 RTP/AVP 8 96 100
c=IN IP4 172.21.8.134
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=ptime:20
###Here is your ACK to the 200 OK from ITSP###
On the ACK...Your CUBE doesnt include any SDP in its ACK, hence your ITSP disconnected the call immediately
ACK sip:[email protected]2.147.134.21:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.21.8.134:5060;branch=z9hG4bKCA1335
From: <sip:[email protected]>;tag=AE7F464-1B0F
To: <sip:[email protected]>;tag=SD7qfu599-1874793413-1396510390487
Date: Thu, 03 Apr 2014 07:33:09 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Authorization: Digest username="AMM-4324-Trunk",realm="amcomvoice.ipsystems.com.au",uri="sip:[email protected]:5060",response="9555a4d29d9316d3f5d416f9a5096ee2",nonce="BroadWorksXhtjq88oeTdvuambBW",cnonce="6AFA84F5",qop=auth,algorithm=MD5,nc=00000001
Allow-Events: telephone-event
Content-Length: 0
017151: Apr 3 07:33:11.089 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 202.147.134.21:5060;branch=z9hG4bKahsb3f108gbhq8pbn5k1sdj0gkbf0.1
From: <sip:[email protected]>;tag=SD7qfu599-1874793413-1396510390487
To: <sip:[email protected]>;tag=AE7F464-1B0F
Call-ID: [email protected]
CSeq: 323009054 BYE
Max-Forwards: 9
Content-Length: 0
I have two suggestions..
1. Downgrade or upgrade your CUBE IOS. Something is not quite right with this behaviour
2. Send your full sh run
On your inbound call issue, you need to send me the logs for a call to another of your DDI.. -
Prefixing a 9 and 91 to incoming calls from SIP provider for callback
I am wondering what would be the best options for prefixing a 9 or 91 to incoming calls over a sip connection to allow callback from missed calls and recieved calls. The setup is
callmanager 7.1.5 >>>>sip trunk>>>>>>>>>>>>CUBE>>>>>>>>>>>sip to ITSP
I am thinking voice translation rules is the only option for this? any configuration examples for this would be greatly appreciated.
would this work?
voice-translation rule 1
rule 1 // /9/
voice-translation profile prefix_9
translate calling 1
dial-peer voice 101 voip
destination-pattern ???????...$
voice-class codec 1
session protocol sipv2
session target ipv4: to callmanager
incoming called-number .
dtmf-relay rtp-nte
dial-peer voice 1001 voip
translation profile incoming prefix_9
destination-pattern T
session protocol sipv2
session target ipv4: to sip provider
incoming called-number ???????...$
dtmf-relay rtp-nteYour config should work fine, except your profile is only applied to one dial-peer, make sure you apply it to the one that is used to redirect the call to CUCM.
Also, you did not mention what country you are in, but if this is US you may want to prefix 91 to national calls as carriers don't provide 9 as part of the CLID delivery, also what about your international calls, you may would be more explicit in your first rule to match for national digit string and then have another rule for international.
HTH,
Chris -
Incoming calls issue in Third Party SIP Phone
Hi,
Yesterday I configured my third party sip phone which is yealink in this case on cucm and successfully registered it with cucm, despite of registration i have some calling issue in this phone. I am able to make outbound calls from this phone to any other phone however issue is related to inbound calls.I tried calling its DN from anywhere but call disconnect after sometime. Also didnt get any proper sip session trace in RTMT. Kindly suggest some step to sortout this issue.
ThanksDear Manish,
Call normally dicsonnected after 30-40 sec with termination code 102 in session trace. PFB SDI trace with 5030 is Thirdparty sip phone and 5033 is c7945. Looking forward for your suggestion.
CallingPartyNumber=5033
|DialingPartition=
|DialingPattern=5030
|FullyQualifiedCalledPartyNumber=5030
|DialingPatternRegularExpression=(5030)
|DialingWhere=
|PatternType=Enterprise
|PotentialMatches=NoPotentialMatchesExist
|DialingSdlProcessId=(0,0,0)
|PretransformDigitString=5030
|PretransformTagsList=SUBSCRIBER
|PretransformPositionalMatchList=5030
|CollectedDigits=5030
|UnconsumedDigits=
|TagsList=SUBSCRIBER
|PositionalMatchList=5030
|VoiceMailbox=
|VoiceMailCallingSearchSpace=PT-LHR-LOCAL:PT-Local:Unityvmpt:PT-F6-Local:PT-ISL-LOCAL:PT-KHI-LOCAL:PT_Operator_LHR:PT_Operator_KHI:PT_Operator_ISL
|VoiceMailPilotNumber=7103
|RouteBlockFlag=RouteThisPattern
|RouteBlockCause=0
|AlertingName=Syed Ahmer
|UnicodeDisplayName=Syed Ahmer
|DisplayNameLocale=1
|OverlapSendingFlagEnabled=0
12:17:38.028 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 172.16.200.21:[5062]:
[23928282,NET]
INVITE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.11:5060;branch=z9hG4bK1ca0cc6e317649
From: "Syed Ahmer" ;tag=8787406~039e2a80-8561-4586-8954-d01ed2aa12c8-246211918
To:
Date: Thu, 30 Jan 2014 07:17:38 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Send-Info: conference, x-cisco-conference
Alert-Info:
Contact:
Remote-Party-ID: "Syed Ahmer" ;party=calling;screen=yes;privacy=off
Max-Forwards: 70
Content-Length: 0
|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
12:17:38.028 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xaf299320|*^*^*
12:17:38.028 |EnvProcessUdpPort::fireSignal - SEND, destination = 172.16.200.21:5062|*^*^*
12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 850, 172.16.200.21:5062)|*^*^* -
I am trying to configure a CUCM with a SIP trunk to a 2811 and a voice GW to my SIP trunk provider.
CUCM8.6 <SIP>2811<SIP> Callcentric.
I am able to make outgoing calls but am failing miserably with incoming.
I suspect it is my incoming dial peer.
The incoming calls hit my 2811 but do not seem to go to my CUCM.
I have attached an output from my "debug ccsip calls"
Anything help would be greatly appreciated.Robert,
surely that is not all the sip debug information, I am missing the INVITES, TRYING etc SIP messages, can you re-attach and maybe also debug your dial peers to see what gets hit (if anything at all ) when making an inbound call.
Cheers
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Cisco 2911 Voice Gateway SIP PSTN Calls Fail
Hello All,
I am having trouble with outboud SIP PSTN calls through a Cisco 2911 Voice Gateway. 2911 VG terminates PSTN SIP Traffic and connects to Avaya CS1000M via QSIG PRI Trunks. When calls are attempted outbound fron the PBX the caller gets a fast busy. Debug ISDN q931 shows the call hitting the 2911 properly, debug voip ccapi inout shows the call matching the correct dial peers and debug ccsip shows the invite to the PSTN Provider SBC, however within the invite the "from" address incorrectly shows the calling number with the provider SBC address (see below). does anyone have any insight on how to correct this? Attached are VG config and Debug isdn q931, voip ccapi inout, ccsip messages and ccsip call. Thanks in advance for any help!!
From: <sip:[email protected]>:tag=6166CDC4-882
To: <sip:[email protected]>
Shawn C. Smithi have same problem my cucm ip is 192.168.200.53
my Voice Gateway is SIP by ip 192.168.200.86 for internal
and 172.29.7.94
and my SIP Server is 10.208.9.69
if its oky can yuo take a look at my problem please
this is the syslog from debug
May 30 20:19:34.284: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Call-Info: <sip:192.168.200.53:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 3047462016-0000065536-0000004549-0902342848
Session-Expires: 1800
P-Asserted-Identity: "Aysar Mohamed" <sip:[email protected]>
Remote-Party-ID: "Aysar Mohamed" <sip:[email protected]>;party=calling;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Content-Length: 0
May 30 20:19:34.284: //-1/B5A494800000/CCAPI/cc_api_display_ie_subfields:
cc_api_call_setup_ind_common:
cisco-username=2217156
----- ccCallInfo IE subfields -----
cisco-ani=2217156
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=90555769123
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
May 30 20:19:34.288: //-1/B5A494800000/CCAPI/cc_api_call_setup_ind_common:
Interface=0x30CF41D4, Call Info(
Calling Number=2217156,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=90555769123(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
Incoming Dial-peer=0, Progress Indication=NULL(0), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=465
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :cc_get_feature_vsa malloc success
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: cc_get_feature_vsa count is 1
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :FEATURE_VSA attributes are: feature_name:0,feature_time:832953048,feature_id:85
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_api_call_setup_ind_common:
Set Up Event Sent;
Call Info(Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=90555769123(TON=Unknown, NPI=Unknown))
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_process_call_setup_ind:
Event=0x2B82D890
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
Try with the demoted called number 90555769123
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetContext:
Context=0x2ABC2E44
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_process_call_setup_ind:
>>>>CCAPI handed cid 465 with tag 0 to app "_ManagedAppProcess_Default"
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallProceeding:
Progress Indication=NULL(0)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
Destination=, Calling IE Present=TRUE, Mode=0,
Outgoing Dial-peer=802, Params=0x2ABC19D4, Progress Indication=NULL(0)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCheckClipClir:
In: Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCheckClipClir:
Out: Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
Destination Pattern=9T, Called Number=0555769123, Digit Strip=FALSE
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=0555769123(TON=Unknown, NPI=Unknown),
Redirect Number=, Display Info=Aysar Mohamed
Account Number=2217156, Final Destination Flag=TRUE,
Guid=B5A49480-0001-0000-0000-11C535C8A8C0, Outgoing Dial-peer=802
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_api_display_ie_subfields:
ccCallSetupRequest:
cisco-username=2217156
----- ccCallInfo IE subfields -----
cisco-ani=2217156
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=0555769123
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x30CF41D4, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=2217156,(Calling Name=Aysar Mohamed)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=0555769123(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=802, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :cc_get_feature_vsa malloc success
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: cc_get_feature_vsa count is 2
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :FEATURE_VSA attributes are: feature_name:0,feature_time:832952824,feature_id:86
May 30 20:19:34.292: //466/B5A494800000/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
May 30 20:19:34.292: //466/B5A494800000/CCAPI/ccCallSetContext:
Context=0x2ABC1984
May 30 20:19:34.292: //465/B5A494800000/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=802
May 30 20:19:34.292: //466/B5A494800000/CCAPI/cc_api_call_proceeding:
Interface=0x30CF41D4, Progress Indication=NULL(0)
May 30 20:19:34.292: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
May 30 20:19:34.292: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Remote-Party-ID: "Aysar Mohamed" <sip:[email protected]>;party=calling;screen=yes;privacy=off
From: "Aysar Mohamed" <sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3047462016-0000065536-0000004549-0902342848
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1401481174
Contact: <sip:[email protected]:5060>
Call-Info: <sip:172.29.7.94:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: kpml, telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Length: 0
May 30 20:19:34.300: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>
CSeq: 101 INVITE
Content-Length: 0
May 30 20:19:34.612: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Record-Route: <sip:10.208.9.69:5060;transport=udp;lr>
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
CSeq: 101 INVITE
Contact: <sip:[email protected]:5060;user=phone>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Content-Length: 328
Content-Type: application/sdp
v=0
o=- 17192647 17192647 IN IP4 10.208.9.69
s=SBC call
c=IN IP4 10.208.9.69
t=0 0
m=audio 39910 RTP/AVP 8 0 102 102 18 116
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 AMR/8000
a=rtpmap:102 AMR/8000
a=rtpmap:18 G729/8000
a=rtpmap:116 telephone-event/8000
a=ptime:5
a=fmtp:116 0-15
a=fmtp:18 annexb=yes
May 30 20:19:34.612: %SIP-3-UNSUPPORTED: Unsupported ptime value
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_caps_ind:
Destination Interface=0x0, Destination Call Id=-1, Source Call Id=466,
Caps(Codec=0x2, Fax Rate=0x2, Vad=0x1,
Modem=0x0, Codec Bytes=160, Signal Type=2)
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
May 30 20:19:34.612: //465/B5A494800000/CCAPI/cc_api_caps_ack:
Destination Interface=0x0, Destination Call Id=-1, Source Call Id=465,
Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=3882)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/cc_api_caps_ack:
Destination Interface=0x0, Destination Call Id=-1, Source Call Id=465,
Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=3882)
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
Event=170, Call Id=466
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
Event=98, Call Id=466
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_call_cut_progress:
Interface=0x30CF41D4, Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1),
Cause Value=0
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_call_cut_progress:
Call Entry(Responsed=TRUE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccCallCutProgress:
Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1), Cause Value=0
Voice Call Send Alert=FALSE, Call Entry(Alert Sent=FALSE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccCallCutProgress:
Call Entry(Responsed=TRUE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccConferenceCreate:
(confID=0x30C11410, callID1=0x1D1, gcid=8C9E3127-E76E11E3-8274BE8C-EC3B12A0, tag=0x0)
May 30 20:19:34.616: //466/B5A494800000/CCAPI/ccConferenceCreate:
(confID=0x30C11410, callID2=0x1D2, gcid=8C9E3127-E76E11E3-8274BE8C-EC3B12A0, tag=0x0)
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
Conference Id=0x30C11410, Call Id1=465, Call Id2=466, Tag=0x0
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
May 30 20:19:34.616: ccConferenceCreate: ret1=0, codecMask1=2, bytes1=160, negot1=0, dtmf1=0
ret2=0, codecMask2=2, bytes2=160, negot2=1, dtmf2=6,
tx_dynamic_pt1=0, rx_dynamic_pt1=0, codec_mode1=0, params_bitmap1 =0
tx_dynamic_pt2=8, rx_dynamic_pt2=8, codec_mode2=0, params_bitmap2 =0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
delay media to slow start case, codec negotation is not done
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_api_bridge_done:
Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=465,
Destination Call Id=466, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/B5A494800000/CCAPI/cc_api_bridge_done:
Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_generic_bridge_done:
Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
Call Entry(Conference Id=0x16, Destination Call Id=466)
May 30 20:19:34.616: //466/B5A494800000/CCAPI/ccConferenceCreate:
Call Entry(Conference Id=0x16, Destination Call Id=465)
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_process_notify_bridge_done:
Conference Id=0x16, Call Id1=465, Call Id2=466
May 30 20:19:34.616: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:[email protected]>;party=called;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 233
v=0
o=CiscoSystemsSIP-GW-UserAgent 2639 5276 IN IP4 192.168.200.86
s=SIP Call
c=IN IP4 192.168.200.86
t=0 0
m=audio 18288 RTP/AVP 8 0 19
c=IN IP4 192.168.200.86
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:19 CN/8000
May 30 20:19:34.680: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Record-Route: <sip:10.208.9.69:5060;transport=udp;lr>
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
CSeq: 101 INVITE
Reason: Q.850;cause=127;text="interworking unspecified"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 0
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_call_disconnected:
Cause Value=41, Interface=0x30CF41D4, Call Id=466
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=TRUE, Cause Value=41, Retry Count=0)
May 30 20:19:34.680: //465/B5A494800000/CCAPI/ccCallReleaseResources:
release reserved xcoding resource.
May 30 20:19:34.680: //466/B5A494800000/CCAPI/ccCallSetAAA_Accounting:
Accounting=0, Call Id=466
May 30 20:19:34.680: //465/B5A494800000/CCAPI/ccConferenceDestroy:
Conference Id=0x16, Tag=0x0
May 30 20:19:34.680: //465/B5A494800000/CCAPI/cc_api_bridge_drop_done:
Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=465,
Destination Call Id=466, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_bridge_drop_done:
Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //465/B5A494800000/CCAPI/cc_generic_bridge_done:
Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
From: "Aysar Mohamed" <sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: kpml, telephone-event
Content-Length: 0
May 30 20:19:34.684: //466/B5A494800000/CCAPI/ccCallDisconnect:
Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=41)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/ccCallDisconnect:
Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x30CF41D4, Tag=0x0, Call Id=466,
Call Entry(Disconnect Cause=41, Voice Class Cause Code=0, Retry Count=0)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.684: :cc_free_feature_vsa freeing 31A5D9F0
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.684: vsacount in free is 1
May 30 20:19:34.684: //465/B5A494800000/CCAPI/ccCallDisconnect:
Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
May 30 20:19:34.684: //465/B5A494800000/CCAPI/ccCallDisconnect:
Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)
May 30 20:19:34.684: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=41
Content-Length: 0
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0
May 30 20:19:34.688: //465/B5A494800000/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x30CF41D4, Tag=0x0, Call Id=465,
Call Entry(Disconnect Cause=41, Voice Class Cause Code=0, Retry Count=0)
May 30 20:19:34.688: //465/B5A494800000/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
May 30 20:19:34.688: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.688: :cc_free_feature_vsa freeing 31A5DAD0
May 30 20:19:34.688: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.688: vsacount in free is 0
May 30 20:19:36.044: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172.29.7.94:5060 SIP/2.0
Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKmisco3ykfiooegpygsphkocp1T20326
Call-ID: isbcfemyk1p1mkteets1tcmi53eeehfhikcp@SoftX3000
From: <sip:172.29.7.94:5060>;tag=sbc0803k1pyk51o
To: <sip:172.29.7.94>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0
May 30 20:19:36.048: //467/8DAABF6C8278/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKmisco3ykfiooegpygsphkocp1T20326
From: <sip:172.29.7.94:5060>;tag=sbc0803k1pyk51o
To: <sip:172.29.7.94>;tag=739BBC-1CE2
Date: Fri, 30 May 2014 20:19:36 GMT
Call-ID: isbcfemyk1p1mkteets1tcmi53eeehfhikcp@SoftX3000
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 446
v=0
o=CiscoSystemsSIP-GW-UserAgent 3496 1601 IN IP4 172.29.7.94
s=SIP Call
c=IN IP4 172.29.7.94
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 172.29.7.94
m=image 0 udptl t38
c=IN IP4 172.29.7.94
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy
My SIP GW internal ip address is 192.168.200.86
and the Public IP is : 172.29.7.94
My CUCM is 192.168.200.53
my GW Config is :
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
codec preference 4 g729br8
voice translation-rule 3
rule 1 /^9\(\)/ /\1/
voice translation-rule 4
rule 4 /^22217/ /7/
rule 5 /^2217/ /7/
rule 6 /^022217/ /7/
rule 7 /^0122217/ /7/
voice translation-rule 5
rule 1 /^5/ /905/
rule 2 /^1/ /901/
rule 3 /^2/ /902/
rule 4 /^3/ /903/
rule 5 /^4/ /904/
rule 6 /^6/ /906/
rule 7 /^7/ /907/
rule 8 /^8/ /908/
rule 10 /^00/ /900/
rule 11 /'+'/ /900/
voice translation-profile OUT
translate called 3
voice translation-profile REDIAL
translate calling 5
voice translation-profile SIP-NEW
translate called 4
application
service mva http://192.168.200.53:8080/ccmivr/pages/IVRMainpage.vxml
service ccm http://192.168.200.53:8080/ccmivr/pages/IVRMainpage.vxml
license udi pid CISCO2921/K9 sn FCZ164960G0
hw-module pvdm 0/0
hw-module pvdm 0/1
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
ip address 192.168.200.86 255.255.255.0
duplex auto
speed auto
interface GigabitEthernet0/1
ip address 172.29.7.94 255.255.255.252
duplex auto
speed auto
ip http server
ip http access-class 23
ip http authentication local
no ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip route 0.0.0.0 0.0.0.0 192.168.200.1
ip route 10.208.9.0 255.255.255.0 172.29.7.93
access-list 23 permit 10.10.10.0 0.0.0.7
control-plane
mgcp profile default
sccp local GigabitEthernet0/0
sccp ccm 192.168.200.53 identifier 1 priority 1 version 7.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 2 register NAGHI-MTP
dspfarm profile 2 mtp
codec g711alaw
maximum sessions hardware 25
associate application SCCP
dial-peer voice 802 voip
description ** SIP TO STC **
translation-profile outgoing OUT
destination-pattern 9T
session protocol sipv2
session target ipv4:10.208.9.69:5060
session transport udp
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay sip-notify rtp-nte sip-kpml
no vad
dial-peer voice 811 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 022217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 812 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 22217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 813 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 2217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 814 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 022217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 815 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 22217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 816 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 2217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 817 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 0122217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 818 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 0122217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
Please i need ur help ASAP -
CUCME Not Incoming Calls, Outgoing calls ok
Hello everybody,
i am configuring a CUCME with SIP trunk, i can make calls to outside but i can´t recieve any from outside, this is my second time a configure with SIP
i´ve used the command debug voice dialpeer all to check was going on, but i can´t find the problem.
this is my config:
ip host sip-server A.B.C.D
voice service voip
ip address trusted list
ipv4 A.B.C.D 255.255.255.252
voice translation-rule 1
rule 1 /325277\(\)/ /1\1/
voice translation-profile IN
translate called 1
dial-peer voice 1 voip
description **Incoming Call from SIP Trunk**
translation-profile incoming IN
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
ephone-dn 1
number 100
description RECEPTION
ephone 2
mac-address AAAA.BBBB.CCCC
ephone-template 1
type 7942
keep-conference
button 1:1
NOTE: IP Address are hidden, just for security
These are the output of my debug/tests:
#test voice translation-rule 1 32527700
Matched with rule 1
Original number: 32527700 Translated number: 100
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=32527700, Called Number=32527700, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=32527700
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=32527700, Expanded String=32527700, Calling Number=32527700T
Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result=-1
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring=32527700, saf_enabled=1, saf_dndb_lookup=1, dp_result=-1
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=NO_MATCH(-1)
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=59513212, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ANSWER; Calling Number=59513212
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=59513212T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result=-1
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:exit@6076
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ORIGINATE; Calling Number=59513212
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=59513212T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result=-1
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:exit@6076
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=NO_MATCH(-1) After All Match Rules Attempt
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-1
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeer:exit@6704
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
Calling Number=59513212, Called Number=32527700, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_VIA_URI; URI=sip:A.B.C.D:5060
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Result=-1
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:exit@6076
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_REQUEST_URI; URI=sip:[email protected]:5060;user=phone
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Result=-1
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:exit@6076
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_TO_URI; URI=sip:[email protected];user=phone
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Result=-1
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:exit@6076
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_FROM_URI; URI=sip:[email protected];user=phone
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Result=-1
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:exit@6076
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=32527700
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Dial String=32527700, Expanded String=32527700, Calling Number=
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/MatchNextPeer:
Result=Success(0); Incoming Dial-peer=1 Is Matched
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:exit@6076
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchSafModulePlugin:
dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=0
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerSPI:exit@6655
Can Anyone help me???
Thanks in Advance!!!Thanks, these are the output
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:32527700@(WAN):5060;user=phone SIP/2.0
Via: SIP/2.0/UDP (SIP_SERVER):5060;branch=z9hG4bK776928550196f0d843ca0b092
Call-ID: SBC9722bb53005161bf8cca630444260574@SoftX3000
From: ;tag=6e8b9968-CC-25
To:
CSeq: 1 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Max-Forwards: 70
Supported: 100rel,timer
User-Agent: Huawei SoftX3000 V300R601
Session-Expires: 300
Min-SE: 90
Contact:
Content-Length: 376
Content-Type: application/sdp
v=0
o=HuaweiSoftX3000 4507886 4507886 IN IP4 (SIP_SERVER)
s=Sip Call
c=IN IP4 (SIP_SERVER)
t=0 0
m=audio 11554 RTP/AVP 8 0 18 4 2 98 98 98
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:98 G726-40/8000
a=rtpmap:98 G726-32/8000
a=rtpmap:98 G726-24/8000
a=ptime:20
a=fmtp:18 annexb=no
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=32527700, Called Number=32527700, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=32527700
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring=32527700, saf_enabled=1, saf_dndb_lookup=1, dp_result=-1
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=NO_MATCH(-1)
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=59513212, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=NO_MATCH(-1) After All Match Rules Attempt
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-1
*Jan 29 16:53:19: //-1/FB88A7CE80F0/DPM/dpAssociateIncomingPeerCore:
Calling Number=59513212, Called Number=32527700, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:53:19: //-1/FB88A7CE80F0/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1
*Jan 29 16:53:19: //-1/FB88A7CE80F0/DPM/dpMatchSafModulePlugin:
dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=0
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 422 Session Timer too small
Via: SIP/2.0/UDP (SIP_SERVER):5060;branch=z9hG4bK776928550196f0d843ca0b092
From: ;tag=6e8b9968-CC-25
To: ;tag=4CD1E84-2094
Date: Wed, 29 Jan 2014 22:53:19 GMT
Call-ID: SBC9722bb53005161bf8cca630444260574@SoftX3000
CSeq: 1 INVITE
Allow-Events: telephone-event
Min-SE: 1800
Server: Cisco-SIPGateway/IOS-15.2.4.M3
Content-Length: 0
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:32527700@(WAN):5060;user=phone SIP/2.0
Via: SIP/2.0/UDP (SIP_SERVER):5060;branch=z9hG4bK776928550196f0d843ca0b092
Call-ID: SBC9722bb53005161bf8cca630444260574@SoftX3000
From: ;tag=6e8b9968-CC-25
To: ;tag=4CD1E84-2094
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
*Jan 29 16:53:31: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:(SIP_SERVER):5060 SIP/2.0
Via: SIP/2.0/UDP (WAN):5060;branch=z9hG4bK3B11F0F
From: ;tag=4CD4D7C-1634
To:
Date: Wed, 29 Jan 2014 22:53:31 GMT
Call-ID: 3017EE62-885411E3-80B4FEFC-CAA82B4A
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M3
Max-Forwards: 70
Timestamp: 1391036011
CSeq: 66 REGISTER
Contact:
Expires: 3600
Supported: path
Content-Length: 0
*Jan 29 16:53:31: //973/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP (WAN):5060;branch=z9hG4bK3B11F0F
Call-ID: 3017EE62-885411E3-80B4FEFC-CAA82B4A
From: ;tag=4CD4D7C-1634
To: ;tag=f2056e8e
CSeq: 66 REGISTER
Content-Length: 0
I´ve replaced the IP Adress for (SIP_SERVER) / (WAN) / SIP_SERVER_INTERNAL
Thank you -
Install of CUCM 8.6.1.10000-43 on VMWare fails on installing progmeter
Hi, Everyone
First time post here, and first time installing CUCM (any version).
I built a VM using a downloaded OVA template for CUCM 6.5.1.10000-43, and the install fails with a critical stop. Once I figured out how to dump logs to the virtual serial port, and looked at the install.log, I found the following information that looks like where the install failed (these are the only LVL::Error and LVL::Critical messages in the log).
It looks like install failed to install the progmeter application component.
Does anyone know what this app does? Any ideas on why it might have failed to install and possible workaround?
I may already have the answer to the question, as I'm attempting install on a VM running on VMWare vServer (rather than ESX/ESXi/vSphere). It is possible that simply being on an unsupported configuration is my problem. I'm trying to work with the equipment immediately available to me, but fully understand that I may have set myself up for failure by even attempting this.
Any help is appreciated greatly.
Regards,
Sean C
INSTALL LOG SNIPPET
08/17/2011 17:51:51 component_install|File:/opt/cisco/install/bin/component_install:743, Function: exec_progmeter(), /opt/cisco/install/bin/progmeter failed (1)|<LVL::Error>
08/17/2011 17:51:51 appmanager.sh|Internal Error, File:/usr/local/bin/base_scripts/appmanager.sh:155, Function: install(), failed to install application components|<LVL::Critical>
08/17/2011 17:51:51 post_install|File:/opt/cisco/install/bin/post_install:869, Function: install_applications(), /usr/local/bin/base_scripts/appmanager.sh -install failed (1)|<LVL::Error>
08/17/2011 17:51:51 post_install|Exiting with result 1|<LVL::Info>
08/17/2011 17:51:51 post_install|INSTALL_TYPE="Basic Install"|<LVL::Debug>
08/17/2011 17:51:51 post_install|File:/opt/cisco/install/bin/post_install:570, Function: check_for_critical_error(), check_for_critical_error, found /common/log/install/critical.log, exiting|<LVL::Error>
08/17/2011 17:51:52 post_install|(CAPTURE) Mail notification cancelled - smtp server address for email not found! [/usr/local/platform/conf/platformConfig.xml]|<LVL::Debug>
08/17/2011 17:51:52 display_screen|Arguments: "Critical Error" "The installation has encountered a unrecoverable internal error. For further assistance report the following information to your support provider.
"/opt/cisco/install/callmanager/scripts/cm_msa_post.sh install PostInstall 8.6.1.10000-43 8.6.1.10000-43 /usr/local/cm/ /usr/local/cm/ /common/log/install/capture.txt " terminated. Exceeded max time (360)
The system will now halt.
Continuing will allow you to dump diagnostic information before halting." "Continue"|<LVL::Debug>Hi,
Has anyone found a fix for this?
I am trying to run call manager 8.6.1 on the following enviroment
VMware ESXI 4.1
4GB of RAM
80 GB HDD Space
I have allocated the VM one CPU
In regards to vsergeyey reponse I have checked the network adapter and this is flexible.
I have run the OVA template firstly before mounting the image
I have tried installing numerous times.
Each and every time near the "Installing Database Component" I get the following error; -
Hi Guys,
I have a SIP trunk setup with a 2811 running CME version 7. I can make outbound calls ok but having issues getting the incoming calls working, i have 1 number on my SIP trunk and that is 01133501788 and i want that to ring my Cisco 7960 which is running SIP firmware not SCCP. I have included by config for anyone who can help me, i just want the incoming call to work.
Many Thanks.
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#
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