Cisco CME and Calls through SIP provider

Hello, friends.
There are Cisco (C2801-ADVENTERPRISEK9_IVS-M), Version 15.1 (4) M7.
Telephones connected to SCCP, registered SIP from the provider.
When I try to call to test number 4444 through sip in debug I see:
*Feb 10 01:51:25.317: //53363/2739DFE79696/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP XXXXXXXXXXX:5060;branch=z9hG4bK100D02077;rport=5060
From: "TEST" <sip:[email protected]>;tag=131CC60C-1D40
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.bef7
Call-ID: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="Uvf1OFL39Awnou/oMiaFQrf9jyybhFmf", qop="auth"
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
*Feb 10 01:51:25.325: //53363/2739DFE79696/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP XXXXXXXXXX:5060;branch=z9hG4bK100D02077
From: "TEST" <sip:[email protected]>;tag=131CC60C-1D40
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.bef7
Date: Sun, 09 Feb 2014 21:51:25 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Cisco при этом зарегана у провайдера SIP
DC#show sip-ua register status
Line peer expires(sec) registered P-Associ-URI
Configuration:
voice service voip
ip address trusted list
ipv4 178.16.26.122 255.255.255.255
ipv4 144.76.42.108 255.255.255.255
ipv4 176.9.145.115 255.255.255.255
ipv4 5.9.108.25 255.255.255.255
ipv4 78.46.95.118 255.255.255.255
ipv4 89.249.23.194 255.255.255.255
ipv4 178.16.26.124 255.255.255.255
ipv4 176.9.85.133 255.255.255.255
ipv4 46.4.53.86 255.255.255.255
ipv4 5.9.84.165 255.255.255.255
ipv4 78.16.26.122 255.255.255.255
ipv4 77.235.62.222 255.255.255.255
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
registrar server
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g711alaw
voice register global
max-dn 10
max-pool 10
voice register dn 1
number 150
voice register dn 2
number 151
voice translation-rule 9
rule 1 /^95/ //
voice translation-rule 1020
rule 1 /^.$/ /40232/
voice translation-profile outgoing
translate calling 1020
translate called 9
mgcp fax t38 ecm
mgcp profile default
dial-peer voice 2 voip
translation-profile outgoing outgoing
destination-pattern 95....
session protocol sipv2
session target sip-server
voice-class codec 1
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte
no vad
sip-ua
credentials username 40232 password 7 XXXXXXXXXX realm sip.zadarma.com
authentication username 40232 password 7 XXXXXXXXXXXX realm sip.zadarma.com
registrar dns:sip.zadarma.com:5060 expires 3600
sip-server dns:sip.zadarma.com:5060
connection-reuse
host-registrar
DC#show sip-ua register status
Line                             peer       expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
150                              40001      12           no
40232                            -1         550          yes
SIP provider says cisco trying to call with the internal call number, and it is necessary in order that have an SIP provider:
Wrong Remote-Party-ID: "Vankuver" <sip:61@<my ip>>;party=calling;
Should be so sip:40232@<my ip>
Please help me!

Yes, I behind nat.
*Feb 10 18:11:53.425: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport
Max-Forwards: 70
Contact:
To: "954444"
From: "150";tag=7b409f06
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 314
v=0
o=- 2 2 IN IP4 192.168.11.14
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.11.14
t=0 0
m=audio 5724 RTP/AVP 107 0 8 101
a=alt:1 2 : gNONJ/Dj BaLJhmb/ 10.200.16.55 5724
a=alt:2 1 : DQ3e8qud c1qVrWui 192.168.11.14 5724
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
*Feb 10 18:11:53.477: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK1038E7FF
From: "" >;tag=169E6BC4-1E16
To: [email protected]>
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0541864002-2442400227-2618163141-2285537806
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1392041513
Contact: outside ip cisco cme:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 262
v=0
o=CiscoSystemsSIP-GW-UserAgent 8076 2450 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18534 RTP/AVP 0 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
*Feb 10 18:11:53.481: //54340/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport
From: "150";tag=7b409f06
To: "954444"
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Feb 10 18:11:53.625: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP outside ip cisco cme:5060;branch=z9hG4bK1038E7FF;rport=5060
From: "" ;tag=169E6BC4-1E16
To: [email protected]>;tag=9fedfddccf3bcc4a1975d2cdb2a664b8.7066
Call-ID: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn", qop="auth"
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
*Feb 10 18:11:53.633: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDPoutside ip cisco cme:5060;branch=z9hG4bK1038E7FF
From: "150" [email protected]>;tag=169E6BC4-1E16
To: [email protected]>;tag=9fedfddccf3bcc4a1975d2cdb2a664b8.7066
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
*Feb 10 18:11:53.637: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDPoutside ip cisco cme:5060;branch=z9hG4bK1038F25FC
From: "" ;tag=169E6BC4-1E16
To: [email protected]>
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0541864002-2442400227-2618163141-2285537806
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1392041513
Contact: :5060>
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="40232",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="df38cd7f4af8e4a808fbbfdf5a7dd6a1",nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn",cnonce="E701683F",qop=auth,algorithm=md5,nc=00000001
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 262
v=0
o=CiscoSystemsSIP-GW-UserAgent 8076 2450 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18534 RTP/AVP 0 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
*Feb 10 18:11:53.981: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK1038F25FC;rport=5060
From: "" ;tag=169E6BC4-1E16
To: [email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Server: kamailio (4.0.3 (x86_64/linux))
Content-Length: 0
*Feb 10 18:11:54.385: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.63.X:5060;rport=5060;branch=z9hG4bK1038F25FC
Record-Route:
From: "k40232" ;tag=169E6BC4-1E16
To: [email protected]>;tag=as7e8de8e5
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Zadarma Voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 281
v=0
o=root 1942395501 1942395501 IN IP4 178.16.26.124
s=Asterisk PBX
c=IN IP4 178.16.26.124
t=0 0
m=audio 12164 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
*Feb 10 18:11:54.409: //54341/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.xxxx.xxxx:5060;branch=z9hG4bK10390E63
From: "150" [email protected]>;tag=169E6BC4-1E16
To: [email protected]>;tag=as7e8de8e5
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 102 ACK
Proxy-Authorization: Digest username="40232",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="df38cd7f4af8e4a808fbbfdf5a7dd6a1",nonce="Uvja/1L42dNbKQpCc2GzgagslkjyE1Pn",cnonce="E701683F",qop=auth,algorithm=md5,nc=00000001
Allow-Events: telephone-event
Content-Length: 0
*Feb 10 18:11:54.429: //54340/204C30429C0D/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.11.14:42294;branch=z9hG4bK-d8754z-e645887cf7416a27-1---d8754z-;rport
From: "150";tag=7b409f06
To: "954444";tag=169E6F78-88E
Date: Mon, 10 Feb 2014 14:11:53 GMT
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: :5060;transport=tcp>
Supported: replaces
Server: Cisco-SIPGateway/IOS-12.x
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 193
v=0
o=CiscoSystemsSIP-GW-UserAgent 149 3396 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 17190 RTP/AVP 8
c=IN IP4 92.63.108.115
a=rtpmap:8 PCMA/8000
a=ptime:20
*Feb 10 18:11:54.653: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 91.231.141.230:42294;branch=z9hG4bK-d8754z-95374017c126c928-1---d8754z-;rport
Max-Forwards: 70
Contact:
To: "954444";tag=169E6F78-88E
From: "150";tag=7b409f06
Call-ID: ZjUzNjkwMWMyZDAyYmY1OWU2NjgzYzQwZjYyZWM5ZGU.
CSeq: 1 ACK
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0

Similar Messages

Cisco CME: calls through SIP-provider again

Hello,friends!
I have already published a discussion here https://supportforums.cisco.com/discussion/12089656/cisco-cme-and-calls-through-sip-provider and you helped me, everything works well for Russian numbers.
When I tried to add the configuration for calls to Belarus, again, there was a problem. I do not understand why, although the configuration ideintichnaya.
My config:
voice service voip
ip address trusted list
ipv4 178.16.26.122 255.255.255.255
ipv4 144.76.42.108 255.255.255.255
ipv4 176.9.145.115 255.255.255.255
ipv4 5.9.108.25 255.255.255.255
ipv4 78.46.95.118 255.255.255.255
ipv4 89.249.23.194 255.255.255.255
ipv4 178.16.26.124 255.255.255.255
ipv4 176.9.85.133 255.255.255.255
ipv4 46.4.53.86 255.255.255.255
ipv4 5.9.84.165 255.255.255.255
ipv4 78.16.26.122 255.255.255.255
ipv4 77.235.62.222 255.255.255.255
ipv4 81.88.86.11 255.255.255.255
ipv4 192.168.1.50 255.255.255.255
ipv4 217.150.198.44 255.255.255.255
ipv4 178.63.96.3 255.255.255.255
ipv4 178.63.96.28 255.255.255.255
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
sip
registrar server
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g711alaw
voice class sip-profiles 20
request INVITE sip-header From modify "\"(.*)\" <sip:(.*)@(.*)>" "\"\" <sip:[email protected]>"
voice translation-rule 9
rule 1 /^98/ /7/
voice translation-rule 10
rule 1 /^9/ //
voice translation-rule 1020
rule 1 /^.*$/ /141756/
voice translation-rule 1030
rule 1 /^.*/ /141756/
voice translation-rule 1040
rule 1 /^.*$/ /21/
voice translation-profile incoming
translate called 1040
voice translation-profile outgoing
translate calling 1030
translate called 9
voice translation-profile outgoing-mezhdunarod
translate calling 1030
translate called 10
voice-card 0
dial-peer voice 2 voip
description TO-RUSSIA
translation-profile outgoing outgoing
preference 1
destination-pattern 98..........
session protocol sipv2
session target sip-server
voice-class codec 1
no voice-class sip outbound-proxy
voice-class sip profiles 20
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte sip-notify
no vad
dial-peer voice 3 voip
translation-profile incoming incoming
incoming called-number 141756
voice-class codec 1
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte
no vad
dial-peer voice 4 voip
description To-Belarus
translation-profile outgoing outgoing-mezhdunarod
destination-pattern 9375.........
session protocol sipv2
session target sip-server
voice-class codec 1
no voice-class sip outbound-proxy
voice-class sip profiles 20
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte sip-notify
no vad
sip-ua
credentials username 141756 password 7<pass> realm sip.zadarma.com
authentication username 141756 password 7 <pass>
no remote-party-id
registrar 1 dns:sip.zadarma.com expires 3600
sip-server dns:sip.zadarma.com
connection-reuse
host-registrar
DEBUG ccsip message:
Jun 17 14:23:09.033: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK36571F65
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>
Date: Tue, 17 Jun 2014 09:23:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1402996989
Contact: <sip:[email protected]:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309
v=0
o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18252 RTP/AVP 0 18 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Jun 17 14:23:09.089: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK36571F65;rport=5060
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.6d40
Call-ID: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.zadarma.com", nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1", qop="auth"
Server: kamailio (4.1.2 (x86_64/linux))
Content-Length: 0
Jun 17 14:23:09.169: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK36571F65
From: "Vankuver" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>;tag=b638310eda6e4a73cf10b7fe3c94c572.6d40
Date: Tue, 17 Jun 2014 09:23:09 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Jun 17 14:23:09.169: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK365820F2
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>
Date: Tue, 17 Jun 2014 09:23:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1402996989
Contact: <sip:[email protected]:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="9534322838cbf2e265b2004bc0aa240e",nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1",cnonce="FFF9A231",qop=auth,algorithm=md5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309
v=0
o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18252 RTP/AVP 0 18 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Jun 17 14:23:09.637: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK365820F2
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>
Date: Tue, 17 Jun 2014 09:23:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1402996989
Contact: <sip:[email protected]:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="9534322838cbf2e265b2004bc0aa240e",nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1",cnonce="FFF9A231",qop=auth,algorithm=md5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309
v=0
o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18252 RTP/AVP 0 18 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Jun 17 14:23:10.621: //14293/D2C3B137AE52/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 92.63.108.115:5060;branch=z9hG4bK365820F2
From: "" <sip:[email protected]>;tag=40FCB218-23D7
To: <sip:[email protected]>
Date: Tue, 17 Jun 2014 09:23:10 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3536040247-4114026979-2924673736-0741251102
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1402996990
Contact: <sip:[email protected]:5060>
Call-Info: <sip:92.63.108.115:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="141756",realm="sip.zadarma.com",uri="sip:[email protected]:5060",response="9534322838cbf2e265b2004bc0aa240e",nonce="U6AYAFOgFtT86kmu2Fr5tYxLYGEexIl1",cnonce="FFF9A
All possible debugging has been turned off
DC#231",qop=auth,algorithm=md5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 309
v=0
o=CiscoSystemsSIP-GW-UserAgent 6656 8059 IN IP4 92.63.108.115
s=SIP Call
c=IN IP4 92.63.108.115
t=0 0
m=audio 18252 RTP/AVP 0 18 8 101
c=IN IP4 92.63.108.115
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Debug voice ccapi inout:
Destination Pattern=9375........., Called Number=375298911396, Digit Strip=FALSE
Jun 17 15:22:13.073: //14425/13366763AF35/CCAPI/ccCallSetupRequest:
Calling Number=141756(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=375298911396(TON=Unknown, NPI=Unknown),
Redirect Number=, Display Info=Vankuver
Account Number=, Final Destination Flag=FALSE,
Guid=13366763-F540-11E3-AF35-FAC82C2E981E, Outgoing Dial-peer=4
Jun 17 15:22:13.073: //14425/13366763AF35/CCAPI/cc_api_display_ie_subfields:
ccCallSetupRequest:
cisco-username=
----- ccCallInfo IE subfields -----
cisco-ani=141756
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=0
dest=375298911396
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=0
cisco-rdnsi=0
cisco-redirectreason=0 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
Jun 17 15:22:13.073: //14425/13366763AF35/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x6968AA04, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=141756,(Calling Name=Vankuver)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=375298911396(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=RegularLine, FinalDestinationFlag=FALSE, Outgoing Dial-peer=4, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
Jun 17 15:22:13.073: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jun 17 15:22:13.073: :cc_get_feature_vsa malloc success
Jun 17 15:22:13.073: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jun 17 15:22:13.077: cc_get_feature_vsa count is 2
Jun 17 15:22:13.077: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jun 17 15:22:13.077: :FEATURE_VSA attributes are: feature_name:0,feature_time:1819298856,feature_id:3371
Jun 17 15:22:13.077: //14427/13366763AF35/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
Jun 17 15:22:13.077: //14427/13366763AF35/CCAPI/ccCallSetContext:
Context=0x6C726BF4
Jun 17 15:22:13.077: //14425/13366763AF35/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=4
Jun 17 15:22:13.085: //14427/13366763AF35/CCAPI/cc_api_call_proceeding:
Please help me... I don't know what to do!

You need to contact service provider for this , after authentication challenge your sip provider is not sending any response.
Contact them and ask whether they had received INVITE with proxy authentication details or not.

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-nte

Your 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

PMF to allow outgoing calls through SIP Trunk Without Registering

Hello,
I have an intermitant issue with one of our UC320W's running 2.3.2(6) firmware. The customers VOIP SIP trunk becomes unregistered for periods of time, stopping incoming and outgoing calls. Once unregistered it takes quite a while to rergister. Our service provider has informed us that the re-register period is the cause and we should try and shorten it, so first question is there a way to do this, also what is the re-register retry window in the first place?
I have an analogue line that can receive calls only so I have made this the fallover number with the VOIP provider, that gives a little releife for incoming calls, but not outgoing. I beleive in other phone systems a SIP trunk does not need to be registered to make an outgoing call, and it is usually an option to say only make outgoing calls if the SIP trunk is registered. I cannot find that option anywhere to deselect it, is there a PMF I could apply to allow outgoing calls without registering?
Thank you,
Tony

Hi Tony,
Please install the SIP_Trunk_Register_Timer.pmf at status->Devices->Alter PMFs in configure utility. Please remember to apply the configuration afterwards. This PMF can let user to select the re-register period. You can find the PMF at https://supportforums.cisco.com/docs/DOC-16301
Regards,
Wendy Yang

Blackberry Curve HELP with 2 things.....sending pics via text and calls/SERVICE PROVIDER IS SPRINT

Can it be done? I am having a hard time figuring that out. Can I send a picture I take using text feature? Not having to send via email or using bluetooth? You know how other phones, you just choose a picture and choose to send the picture and then you choose the contact to send to.... How do I do that with the curve? I need to send pictures and sending via email doesnt work when I need the picture to go through right away. Help!
Also, I have been having this huge problem... I get a phone call or make a phone call and the phone goes back to my screen saver and not the phone call screen where it shows how long you are on the phone, the screen where it shows you are on a phone call. It has been that sometimes the phone doesnt hang up and since the call stays on the screen saver screen I dont realize it and people on the phone are listening to my conversations. Or i accidently hit a button without realizing it and it calls someone but I dont know this because the phone doesnt show me its making a phone call and I hear voices.... not in my head, coming from the phone. Or i make a phone call and think the call didnt go through and start going through phone book and other options on my phone without realizing the person I called is already on the phone. I hope I made sense.
Message Edited by veronicazambran on 03-03-2009 07:30 PM

Welcme to the Frums!
Who is your service provider?
Nurse-Berry
Follow NurseBerry08 on Twitter

Cisco Forward and Call return to Voicemail on CUCM 8.5 question

Hi,
I have a user A that requires the call forward when busy/no answer to another extension B. If extension B is busy/no answer, A wants call to be returned to his voicemail.
Unfortunately User B has also a call forward to another extension c, so call forward from A are forwarded to C when B is busy.
Is there any means to have the calls from A return to his voicemail when B is busy/no answer.
I would be grateful if someone can help or is it a system restrictions.

Hi
That's how it works by default.
Unity looks at the 'first redirecting' number, and uses that to allocate forwarded calls to a VM box.
So if User A forwards to User B, the first forwarding number is User A. It goes in User A's box.
That applies regardless of how many times it's forwarded, unless something happens to 'lose' the forwarding number info. That wouldn't usually happen on-system, more likely if a PRI or other trunk is traversed.
Aaron

3725 + CME + SIP Provider = Frustration

I am a telecom tech trying to learn about more about the Cisco world. I have been trying to get CME registered to a SIP provider (Broadvoice) for a few weeks now with no luck. Can anyone look at this and let me know if there are any blatent problems? I am including some of a DEBUG MESSAGES below as well.
*************************************3725 CONFIG****************************************************
! Last configuration change at 18:05:07 cst Thu Feb 28 2002
! NVRAM config last updated at 18:06:54 cst Thu Feb 28 2002
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname CME3725
boot-start-marker
boot-end-marker
no aaa new-model
memory-size iomem 5
clock timezone cst -6
ip cef
ip host sip.broadvoice.com 147.135.8.128
ip host proxy.nyc.broadvoice.com 147.135.20.221
multilink bundle-name authenticated
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
h323
call service stop
sip
bind control source-interface FastEthernet0/0
bind media source-interface FastEthernet0/0
registrar server expires max 3600 min 3600
   localhost dns:sip.broadvoice.com
no update-callerid
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
voice register global
mode cme
source-address 192.168.1.201 port 5060
max-dn 2
max-pool 1
authenticate register
tftp-path flash:
create profile sync 0011343535014052
voice register dn 1
number 21443XXXXX
allow watch
name cisco
shared-line
label 1005
mwi
voice register pool 1
id mac 0000.0000.0000
number 1 dn 1
dtmf-relay rtp-nte
username 1005 password 1005
codec g711alaw
voice source-group SIP-Trunks
access-list 50
voice source-group SIP_Trunks
voice translation-rule 1
rule 1 /^.*/ /21443XXXXX/
voice translation-rule 2
rule 1 /21443XXXXX/ /1005/
voice translation-rule 3
rule 1 /^214(.*)/ /\1/
rule 2 /$..........$/ /1\1/
voice translation-profile Broadvoice_IN
translate calling 3
translate called 2
voice translation-profile Broadvoice_OUT
translate calling 1
username cisco privilege 15 secret 5 $1$MB2M$RtpE/ooDpcXUIfij1GCJ0.
username 1005 password 0 1005
archive
log config
hidekeys
interface FastEthernet0/0
ip address 192.168.1.201 255.255.255.0
speed auto
half-duplex
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 192.168.1.254
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:
control-plane
dial-peer voice 1 voip
description ** Outgoing Broadvoice 10-digit **
translation-profile outgoing Bradvoice_OUT
preference 2
destination-pattern 1..........
voice-class codec 1
session protocol sipv2
session target ipv4:147.135.20.221
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 43XXXXX voip
description ** Incoming Broadvoice **
translation-profile incoming Broadvoice_IN
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number 21443XXXXX
dtmf-relay rtp-nte
codec g711ulaw
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 86 voip
description ** Outgoing Broadvoice Voice-Mail **
destination-pattern *86
voice-class codec 1
session protocol sipv2
session target ipv4:147.135.20.221
dtmf-relay rtp-nte
ip qos dscp cs5 media
no vad
sip-ua
authentication username 21443XXXXX password 7 143F21XXXXXXXXXXXXXXXXX realm BroadWorks
no remote-party-id
retry register 3
retry options 1
timers connect 100
mwi-server ipv4:147.135.20.221 expires 3600 port 5060 transport udp unsolicited
registrar ipv4:147.135.20.221 expires 3600
sip-server ipv4:147.135.20.221
host-registrar
telephony-service
load 7921 CP7921G-1.0.1/CP7921G-1.0.1.
max-ephones 5
max-dn 5
ip source-address 192.168.1.201 port 2000
max-conferences 4 gain -6
dn-webedit
transfer-system full-consult
ephone-dn 1
number 1003 no-reg primary
name The Fishers
ephone-dn 2
number 1002 no-reg primary
name Other Phones
ephone 1
device-security-mode none
mac-address 0023.5E67.74EA
type 7921
button 1:1
ephone 2
device-security-mode none
mac-address 0023.5E67.758C
type 7921
button 1:2
line con 0
stopbits 1
line aux 0
stopbits 1
line vty 0 4
login
ntp clock-period 17180118
ntp master
ntp server 129.6.15.28
end
********************************************DEBUG****************************************************
Aug 8 01:34:16.316: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:41812;branch=z9hG4bK-d87543-06266a34ed272f5b-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:41812>
To: "92145XXXXXX"<sip:[email protected]>
From: "MY NAME HERE"<sip:[email protected]>;tag=5f37a274
Call-ID: 6220fa11bb1c6c46ODhkYmEwYzRlMmFmNzY0NDdkZjQzZDFlMzEzMzFhM2Q.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1002tx stamp 29712
Content-Length: 485
v=0
o=- 5 2 IN IP4 192.168.1.200
s=<CounterPath eyeBeam 1.5>
c=IN IP4 192.168.1.200
t=0 0
m=audio 26344 RTP/AVP 107 119 0 98 8 3 101
a=alt:1 3 : orcMzWYQ jqWa9BMB 192.168.1.200 26344
a=alt:2 2 : S9KWsCq2 awpCGnJ0 192.168.1.76 26344
a=alt:3 1 : rMS6WAXp CvmP73Zj 192.168.1.100 26344
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:A8F366E8CB8B472F8215DFD332367F73
Aug 8 01:34:16.444: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.200:41812;branch=z9hG4bK-d87543-06266a34ed272f5b-1--d87543-;rport
From: "MY NAME HERE"<sip:[email protected]>;tag=5f37a274
To: "92145XXXXXX"<sip:[email protected]>
Date: Sun, 08 Aug 2010 01:34:16 GMT
Call-ID: 6220fa11bb1c6c46ODhkYmEwYzRlMmFmNzY0NDdkZjQzZDFlMzEzMzFhM2Q.
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 INVITE
Allow-Events: telephone-event
Content-Length: 0
Aug 8 01:34:16.592: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK1A91C
From: "MY NAME HERE" <sip:[email protected]>;tag=2E67CC-894
To: <sip:[email protected]>
Date: Sun, 08 Aug 2010 01:34:16 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 3828225533-2713915871-2151408495-2897475455
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1281231256
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 250
v=0
o=CiscoSystemsSIP-GW-UserAgent 3473 6602 IN IP4 192.168.1.201
s=SIP Call
c=IN IP4 192.168.1.201
t=0 0
m=audio 16398 RTP/AVP 8 101
c=IN IP4 192.168.1.201
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Aug 8 01:34:16.752: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Call-ID: [email protected]
CSeq: 101 INVITE
From: "MY NAME HERE" <sip:[email protected]>;tag=2E67CC-894
To: <sip:[email protected]>
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK1A91C
Content-Length:    0
Aug 8 01:34:16.792: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Forbidden
Call-ID: [email protected]
CSeq: 101 INVITE
From: "MY NAME HERE" <sip:[email protected]>;tag=2E67CC-894
To: <sip:[email protected]>;tag=vwxy
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK1A91C
Allow-Events: telephone-event
User-Agent: Cisco-SIPGateway/IOS-12.x
Content-Length: 187
Content-Type: application/sdp
v=0
o=1664745546 3473 6602 IN IP4 99.53.0.78
s=-
c=IN IP4 99.53.0.78
t=0 0
m=audio 16398 RTP/AVP 8 101
c=IN IP4 99.53.0.78
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
Aug 8 01:34:16.900: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK1A91C
From: "MY NAME HERE" <sip:[email protected]>;tag=2E67CC-894
To: <sip:[email protected]>;tag=vwxy
Date: Sun, 08 Aug 2010 01:34:16 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Aug 8 01:34:16.912: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.200:41812;branch=z9hG4bK-d87543-06266a34ed272f5b-1--d87543-;rport
From: "MY NAME HERE"<sip:[email protected]>;tag=5f37a274
To: "92145XXXXXX"<sip:[email protected]>;tag=2E6920-1C05
Date: Sun, 08 Aug 2010 01:34:16 GMT
Call-ID: 6220fa11bb1c6c46ODhkYmEwYzRlMmFmNzY0NDdkZjQzZDFlMzEzMzFhM2Q.
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=57
Content-Length: 0
Aug 8 01:34:16.984: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:41812;branch=z9hG4bK-d87543-06266a34ed272f5b-1--d87543-;rport
To: "92145XXXXXX"<sip:[email protected]>;tag=2E6920-1C05
From: "MY NAME HERE"<sip:[email protected]>;tag=5f37a274
Call-ID: 6220fa11bb1c6c46ODhkYmEwYzRlMmFmNzY0NDdkZjQzZDFlMzEzMzFhM2Q.
CSeq: 1 ACK
Content-Length: 0
************************************SIP REG STATUS************************************************
CME3725#SHO SIP REG STATUS
Line          peer           expires(sec) registered
============ ============= ============ ===========
CME3725#

Two things appear to be occurring:
a) You don't have a registration with your provider. Maybe they don't require that. But if they do, no numbers are trying to be registered.
b) The inbound call is not matching an internal extension, and as a result is matching a pattern and routing back out to your ITSP.
You can take care of both of these with:
ephone-dn 1
number 1003 secondary no-reg primary
name The Fishers
Now, make a call to that number you used for the secondary number. Assuming a phone is assigned to DN 1 and registered, it will ring that phone.
-Steve

Unable to perform call transfer or call park for an outbound call via SIP Trunk (SKYPE)

We have configured the SIP Trunk & SIP profile and successfull make outbound call through SIP Trunk (SKYPE). However, we are not able to perform call transfer or call park when the call is connected.
The scenario is:
A call to an phone number via SIP trunk, when call established, A perform call-transfer to B. After the call-transfer, the call Drop and Phone B show error code "Temp Fail"
When i select "enable MTP" in SIP trunk, we are able to call transfer and call park. But it limit the number of call session to 1.

You are probably running into some sort of Codec issue. IE, your phone is G.711 and the trunk is G.729. You will need to transcode the call at somepoint.

Cisco ISE and SecurID Integration Questions

I'm looking for some clarity trying to understand something conceptually. I want to integrate Cisco ISE with RSA SecurID, the idea being that if the user authenticates with RSA SecurID they end up on one VLAN, however, if they don't authenticate with (or don't use, or don't have) SecurID they'll end up on another VLAN. Note that I'm not using SecurID for wireless access...all PCs are wired to Ethernet.
We have been using RSA SecurID for a while and are currently on version 8.0. Our users are authenticating via the RSA Agent typically on Windows 8.1. Instead of the usual Windows login prompt, the RSA Agent first prompts for the username and passcode (they use an app on their smartphones to get the passcode), then after a moment or two, it prompts for their Windows domain password.
We have recently installed Cisco ISE version 1.3. With the help of a local Cisco engineer and going through the "Cisco Identity Services Engine User Guide", I have it set up and running along with a few 'test' ports on our Cisco 6809 switch, it basically works...as a test it's simply set up that if they authenticate they're on one VLAN, if not, they end up on another (this is currently without using RSA...just out-of-the-box Windows authentication).
The Cisco engineer was unable to help me with RSA SecurID, so pressing on without him, out of the same user guide I have followed the directions for "RSA Identity Sources" under the "Managing Users and External Identity Sources", and that went well as far as ISE is concerned; I am now ready to get serious about getting ISE and SecurID working together.
My mistake in this design so far was assuming that the RSA agent on the Windows client PCs would communicate with Cisco ISE...there doesn't seem to be a way to have them point to a non-RSA SecurID server for authentication. The concept I'm missing is what, or how, the end-user machine is supposed to authenticate taking advantage of both ISE and SecurID.
I have dug deeper into the Cisco ISE documentation but it seems heavily biased towards Wi-Fi and BYOD implementations and it's not clear to me what applies to wired vs wireless. Perhaps it's a case that I'm not seeing the forest for the trees, but I'm not understanding what the end-user authentication looks like. It apears that as I learn more about ISE, it should become the primary SSO source, that SecurID becomes just an identity source and the PC clients would no-longer directly communicate with the SecurID servers. That being the case, do I need to replace the SecurID client on the PCs and something else Cisco-ish fills this role? An agent for ISE? How do they continue to use their passcode without the RSA agent?
Thanks!

The external db not operation indicates that there is no communication between ACS and RSA. Did you fetch the package.cab file to analyse the auth.log file?
Have you already gone through the below listed link?
http://www.security-solutions.co.za/cisco-CSACS-1113-SE-4.2-RSA-Authentication-Manager-Integration-Configuration-Example.html
Regards,
Jatin Katyal
- Do rate helpful posts -

Cisco Phone 7960 and SIP provider

Hi,
i have an account with a Sip provider.
I have all information for make a connection with xlite sip client but if i try to configure a Cisco Phone with SIP Firmware (7.5), phone not work.
My provider is messagenet.it.
Can you help me?
Thanks

Hello,
have a look at the configuration guide "Getting Started with Your Cisco SIP IP Phone" at
http://www.cisco.com/en/US/products/sw/voicesw/ps2156/products_administration_guide_chapter09186a0080080edf.html
This should pretty much answer your questions and allow you to succeed with your task.
Hope this helps! Please rate all posts.
Regards, Martin

Problem with sip trunk between CCM and Huawei through Cisco ASA5520

Hello,
I have a next problem
During SIP conversation ASA is changing the ip address of CCM to corresponding name in ASA configuration inside the SIP packet:
To: <sip:443230282@Server_CCM1;user=phone>
ASA name configuration:
name x.y.z.h Server_CCM1
But it should be without any changes like that: To: <sip:[email protected];user=phone>. Because of that session cant be established. Remote SIP peer gives an error "Bad Request - 'Malformed/Missing URL"
When name was deleted in ASA "no name x.y.z.h Server_CCM1" we have no any problem with SIP initialization and call proccesing.
We are going to upgrade ASA from 8.2 to 8.3 and it seems that we will have the same problem because object will be created automaticly in new version (we are using a NAT) and we will not be able to delete an object like we did in version 8.2.
What configuration in ASA version 8.3 should be done to avoid this issue.
P.S Detailed debug from Huawei in attachment.
Thank you.

Hi.
depending on your config, you might be hitting CSCta16361, this is fixed in 8.2(4)
if you can confirm it's still happening in latest 8.2 release, then a TAC case needs to be opened so investigation is done and a new bug is opened.
if you've tested 8.2(4) already and it's still doing the same, then a TAC Service Request should be opened for more investigation and possibly opening a new defect.
Best regards,
Fadi.
does this answer your question? if yes please mark it resolved.

How can i transfer a call from SIP 9971 to PBX system on CME router

hello everybody,
I have a critical problem about interaction of transfering feature between CME router and pbx panasonic system in some status. let me explain more detail about this issue..i have a SIP 9971(CP-9971) registered on CME at the one site and a voice gateway that is connect with PBX system through a E1 pri trunk connection at the other site. totally the integration between CME and PBX is ok and there is no problem in two direction, i mean i can call pbx system from cp-9971 and vise versa but when i call from a phone which is registered on PBX site to SIP 9971 which is registered on cisco CME call is connected,then when i try to transfer that call to another phone at PBX site, the session is open between two panasonic phones but no audio transmited in two direction. in addition every thing works fine about SCCP phones(transfer feature works fine). here is my configuration file. i hope someone could help me because i've searched a lot but no result help help help plz....
cme router 3845 configuration
VOIP-3845#show running-config
Building configuration...
Current configuration : 12657 bytes
! Last configuration change at 11:44:01 UTC Mon Oct 31 2011 by admin
! NVRAM config last updated at 11:44:02 UTC Mon Oct 31 2011 by admin
! NVRAM config last updated at 11:44:02 UTC Mon Oct 31 2011 by admin
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname VOIP-3845
boot-start-marker
boot-end-marker
no aaa new-model
clock calendar-valid
dot11 syslog
ip source-route
ip cef
no ipv6 cef
multilink bundle-name authenticated
voice-card 0
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
sip
bind control source-interface Loopback10
bind media source-interface Loopback10
registrar server
voice register global
mode cme
source-address 192.168.2.1 port 5060
max-dn 720
max-pool 262
load 9971 sip9971.9-1-1SR1.loads
authenticate register
authenticate realm cisco.com
tftp-path flash:
file text
create profile sync 0063544528862458
camera
video
voice register dn 1
number 500
voice register dn 2
number 600
voice register dn 3
number 700
name test
voice register template 1
softkeys idle Newcall Redial Cfwdall
softkeys connected Confrn Endcall Hold Trnsfer
voice register pool 1
id mac B8BE.BF23.5242
type 9971
number 1 dn 1
template 1
username test password test
camera
video
blf-speed-dial 4 600 label "test"
voice register pool 2
id mac B8BE.BF9C.5476
type 9971
number 1 dn 2
template 1
username bank password bank
camera
video
voice register pool 3
id mac B8BE.BF9C.51D4
type 9971
number 1 dn 3
template 1
username test1 password test1
camera
video
voice register pool 4
id mac B8BE.BF9C.4FA2
number 1 dn 1
camera
video
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-1576175886
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-1576175886
revocation-check none
rsakeypair TP-self-signed-1576175886
crypto pki certificate chain TP-self-signed-1576175886
certificate self-signed 01
30820241 308201AA A0030201 02020101 300D0609 2A864886 F70D0101 04050030
31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
69666963 6174652D 31353736 31373538 3836301E 170D3131 31303038 30393034
34365A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D31 35373631
37353838 3630819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
8100D6EC 47BCDC3C 82F43FF3 23522678 2616868D 9910DCD2 E36016B3 D7B40DA7
53A6E339 4978D451 21F051BE B21F8AD5 86B952DC 1ECCE371 3E094B54 26A41E14
A3055C06 AE860756 425E5C50 E62B3287 631B1E87 9BAC2E39 2810E120 DA3BF823
947EA591 81CA5489 1B868239 E835EC7C 0AA7651A 22D6E47F 545EBEF3 A172C9A3
5A0D0203 010001A3 69306730 0F060355 1D130101 FF040530 030101FF 30140603
551D1104 0D300B82 09564F49 502D3338 3435301F 0603551D 23041830 1680146C
934AD072 99DDC600 ECD6F389 8F71E0C2 18EC2E30 1D060355 1D0E0416 04146C93
4AD07299 DDC600EC D6F3898F 71E0C218 EC2E300D 06092A86 4886F70D 01010405
00038181 000E82F6 5FBB847C 49226955 6F7DECE7 0B093513 D57C35D5 4CD22FA7
8144A080 B0D56C8D 86AF8156 0152443A A3FBE59F B1AEFFBC BEB43E09 35757BAD
4C06FC4A 0F3695E0 B00FBD30 4E8F36CE 7748F39C F9602650 7A1D2D48 DBC31237
AE3D63CE 593D31F5 62E4916F D20E30E8 30DC55C0 120FBD26 D2768DBC A67DDC34
5BDB66B1 E3
quit
license udi pid CISCO3845-MB sn FOC14421Q1Y
archive
log config
hidekeys
username admin privilege 15 secret 5 $1$Zf7j$P93opukmmEBIioVpjmHB3.
redundancy
interface Loopback10
ip address 192.168.2.1 255.255.255.0
interface Tunnel1
ip address 172.25.10.1 255.255.255.0
no ip redirects
ip nhrp map multicast dynamic
ip nhrp network-id 10
tunnel source GigabitEthernet0/1.1
tunnel mode gre multipoint
tunnel key 100
interface Tunnel2
ip address 172.25.11.1 255.255.255.0
no ip redirects
ip nhrp map multicast dynamic
ip nhrp network-id 20
tunnel source GigabitEthernet0/1.2
tunnel mode gre multipoint
interface Tunnel14
ip address 192.168.13.129 255.255.255.252
tunnel source GigabitEthernet0/1.1
tunnel destination 10.2.68.25
interface Tunnel18
ip address 192.168.13.137 255.255.255.252
tunnel source GigabitEthernet0/1.1
tunnel destination 10.9.160.236
interface GigabitEthernet0/0
no ip address
shutdown
duplex auto
speed auto
media-type rj45
interface GigabitEthernet0/1
no ip address
duplex auto
speed auto
media-type rj45
interface GigabitEthernet0/1.1
encapsulation dot1Q 10
ip address 10.9.160.25 255.255.255.0
interface GigabitEthernet0/1.2
encapsulation dot1Q 50
ip address 10.10.9.25 255.255.255.0
router eigrp 202
network 172.25.11.0 0.0.0.255
network 192.168.2.0 0.0.0.15
redistribute static route-map MYMAP1
router eigrp 201
network 172.25.10.0 0.0.0.255
network 192.168.2.0 0.0.0.15
redistribute static route-map MYMAP1
ip forward-protocol nd
ip http server
ip http secure-server
ip http path flash:/gui
ip route 10.2.68.0 255.255.255.0 10.9.160.1
ip route 10.10.0.0 255.255.0.0 10.10.9.1
ip route 10.64.164.30 255.255.255.255 10.9.160.1
ip route 192.168.14.0 255.255.255.0 192.168.13.130
ip route 192.168.17.0 255.255.255.0 Tunnel18
ip access-list standard REDIS1
permit 192.168.14.0
permit 192.168.17.0
route-map MYMAP1 permit 10
match ip address REDIS1
snmp-server community test RO
tftp-server flash:term11.default.loads
tftp-server flash:dkern9971.100609R2-9-0-3.sebn
tftp-server flash:kern9971.9-0-3.sebn
tftp-server flash:rootfs9971.9-0-3.sebn
tftp-server flash:sboot9971.111909R1-9-0-3.sebn
tftp-server flash:sip9971.9-0-3.loads
tftp-server flash:skern9971.022809R2-9-0-3.sebn
tftp-server flash:sccp11.9-0-2sr1s
tftp-server flash:SCCP11.9-1-1SR1S.loads
tftp-server flash:apps11.9-1-1TH1-16.sbn
tftp-server flash:cnu11.9-1-1TH1-16.sbn
tftp-server flash:cvm11sccp.9-1-1TH1-16.sbn
tftp-server flash:dsp11.9-1-1TH1-16.sbn
tftp-server flash:jar11sccp.9-1-1TH1-16.sbn
tftp-server flash:term06.default.loads
tftp-server flash:sip9971.9-1-1SR1.loads
tftp-server system:cme/sipphone
tftp-server flash:Desktops/320x212x12/NantucketFlowers.png
tftp-server flash:Desktops/320x212x12/TN-CampusNight.png
tftp-server flash:Desktops/320x212x12/TN-CiscoFountain.png
tftp-server flash:Desktops/320x212x12/TN-Fountain.png
tftp-server flash:Desktops/320x212x12/TN-MorroRock.png
tftp-server flash:Desktops/320x212x12/TN-NantucketFlowers.png
tftp-server flash:Desktops/320x212x12/Fountain.png
tftp-server flash:Desktops/320x212x12/CiscoLogo.png
tftp-server flash:Desktops/320x212x12/TN-CiscoLogo.png
tftp-server flash:Desktops/320x212x12/List.xml
tftp-server flash:Desktops/320x216x16/List.xml
tftp-server flash:Desktops/320x212x16/List.xml
tftp-server flash:gui/admin_user.html
tftp-server flash:gui/admin_user.js
tftp-server flash:gui/CiscoLogo.gif
tftp-server flash:gui/Delete.gif
tftp-server flash:gui/dom.js
tftp-server flash:gui/downarrow.gif
tftp-server flash:gui/ephone_admin.html
tftp-server flash:gui/logohome.gif
tftp-server flash:gui/normal_user.html
tftp-server flash:gui/normal_user.js
tftp-server flash:gui/Plus.gif
tftp-server flash:gui/sxiconad.gif
tftp-server flash:gui/Tab.gif
tftp-server flash:gui/telephony_service.html
tftp-server flash:gui/uparrow.gif
tftp-server flash:gui/xml-test.html
tftp-server flash:gui/xml.template
tftp-server flash:ringtones/Analog1.raw
tftp-server flash:ringtones/Analog2.raw
tftp-server flash:ringtones/AreYouThere.raw
tftp-server flash:ringtones/AreYouThereF.raw
tftp-server flash:ringtones/Bass.raw
tftp-server flash:ringtones/CallBack.raw
tftp-server flash:ringtones/Chime.raw
tftp-server flash:ringtones/Classic1.raw
tftp-server flash:ringtones/Classic2.raw
tftp-server flash:ringtones/ClockShop.raw
tftp-server flash:ringtones/DistinctiveRingList.xml
tftp-server flash:ringtones/Drums1.raw
tftp-server flash:ringtones/Drums2.raw
tftp-server flash:ringtones/FilmScore.raw
tftp-server flash:ringtones/HarpSynth.raw
tftp-server flash:ringtones/Jamaica.raw
tftp-server flash:ringtones/KotoEffect.raw
tftp-server flash:ringtones/MusicBox.raw
tftp-server flash:ringtones/Piano1.raw
tftp-server flash:ringtones/Piano2.raw
tftp-server flash:ringtones/Pop.raw
tftp-server flash:ringtones/Pulse1.raw
tftp-server flash:ringtones/Ring1.raw
tftp-server flash:ringtones/Ring2.raw
tftp-server flash:ringtones/Ring3.raw
tftp-server flash:ringtones/Ring4.raw
tftp-server flash:ringtones/Ring5.raw
tftp-server flash:ringtones/Ring6.raw
tftp-server flash:ringtones/Ring7.raw
tftp-server flash:ringtones/RingList.xml
tftp-server flash:ringtones/Sax1.raw
tftp-server flash:ringtones/Sax2.raw
tftp-server flash:ringtones/Vibe.raw
tftp-server flash:APPS-1.2.1.SBN
tftp-server flash:SYS-1.2.1.SBN
tftp-server flash:GUI-1.2.1.SBN
tftp-server flash:CP7921G-1.2.1.LOADS
tftp-server flash:TNUX-1.2.1.SBN
tftp-server flash:TNUXR-1.2.1.SBN
tftp-server flash:WLAN-1.2.1.SBN
tftp-server flash:apps37sccp.1-2-1-0.bin
tftp-server flash:APPSH-1.3.1.SBN
tftp-server flash:GUIH-1.3.1.SBN
tftp-server flash:CP7925G-1.3.1.LOADS
tftp-server flash:SYSH-1.3.1.SBN
tftp-server flash:TNUXH-1.3.1.SBN
tftp-server flash:WLANH-1.3.1.SBN
tftp-server flash:SCCP11.9-2-1S.loads
tftp-server flash:Desktops/320x212x12/CampusNight.png
tftp-server flash:Desktops/320x212x12/CiscoFountain.png
tftp-server flash:Desktops/320x212x12/MorroRock.png
tftp-server flash:skern9971.022809R2-9-2-1.sebn
tftp-server flash:sip9971.9-2-1.loads
tftp-server flash:sboot9971.031610R1-9-2-1.sebn
tftp-server flash:rootfs9971.9-2-1.sebn
tftp-server flash:dkern9971.100609R2-9-2-1.sebn
tftp-server flash:kern9971.9-2-1.sebn
tftp-server flash:United_States/g4-tones.xml
tftp-server flash:English_United_States/gd-sip.jar
tftp-server flash:sboot9971.031610R1-9-1-1SR1.sebn alias sboot9971.031610R1-9-1-1SR1.sebn
tftp-server flash:rootfs9971.9-1-1SR1.sebn alias rootfs9971.9-1-1SR1.sebn
tftp-server flash:kern9971.9-1-1SR1.sebn alias kern9971.9-1-1SR1.sebn
tftp-server flash:dkern9971.100609R2-9-1-1SR1.sebn alias dkern9971.100609R2-9-1-1SR1.sebn
tftp-server flash:skern9971.022809R2-9-1-1SR1.sebn alias skern9971.022809R2-9-1-1SR1.sebn
control-plane
mgcp profile default
dial-peer voice 1 voip
description connection-trough-PBX
destination-pattern 0....
session target ipv4:192.168.13.130
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 100 voip
description K
destination-pattern 9T
session target ipv4:192.168.13.130
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 5 voip
shutdown
destination-pattern *3709
session protocol sipv2
session target ipv4:192.168.13.130
session transport tcp
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
dial-peer voice 2 pots
incoming called-number .
dial-peer voice 10 voip
gatekeeper
shutdown
telephony-service
em logout 0:0 0:0 0:0
max-ephones 262
max-dn 400
ip source-address 192.168.2.1 port 2000
load 7911 SCCP11.9-2-1S
max-conferences 12 gain -6
web admin system name admin secret 5 $1$IKnn$tyKyuBcGqXFl6nhxCSu.z0
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp 7960 Oct 29 2011 12:39:25
ephone-template 1
softkeys connected Confrn Endcall Trnsfer Hold
keep-conference endcall
ephone-dn 1 dual-line
number 200
label test
name test
ephone-dn 2 dual-line
number 300
label Sepahbod
name Sepahbod
ephone-dn 4 dual-line
number 666
ephone-dn 5 dual-line
number 660
ephone-dn 6 dual-line
number 670
ephone-dn 7 dual-line
number 770
ephone-dn 8 dual-line
number 770
ephone-dn 9 dual-line
number 999
ephone 1
device-security-mode none
mac-address 18EF.639F.BCB0
keep-conference endcall
button 1:1
ephone 2
device-security-mode none
mac-address 0025.8418.B017
ephone-template 1
keep-conference endcall
button 1:2
ephone 3
device-security-mode none
mac-address F04D.A243.3154
keep-conference endcall
button 1:4
ephone 4
device-security-mode none
mac-address 6CF0.496A.69E9
button 1:4
ephone 5
device-security-mode none
mac-address 0015.E987.345F
keep-conference endcall
button 1:5
ephone 6
device-security-mode none
mac-address 0024.1DEA.614A
keep-conference endcall
button 1:6
ephone 9
device-security-mode none
mac-address 001D.7D4D.4DCB
button 1:9
line con 0
line aux 0
line vty 0 4
login local
transport input telnet
scheduler allocate 20000 1000
end
and Voice Gateway connected two PBX system configuration
Current configuration : 3486 bytes
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Voice-GW
boot-start-marker
boot-end-marker
card type e1 0 2
no aaa new-model
network-clock-participate wic 2
dot11 syslog
ip source-route
ip cef
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type primary-net5
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
voice-card 0
crypto pki token default removal timeout 0
license udi pid CISCO2811 sn FHK1352F0E9
username admin privilege 15 secret 5 $1$O6AN$1kvvqiLdIl3/ZTHoyYRy0/
redundancy
controller E1 0/2/0
framing NO-CRC4
pri-group timeslots 1-31
controller E1 0/2/1
interface Tunnel14
ip address 192.168.13.130 255.255.255.252
tunnel source FastEthernet0/1
tunnel destination 10.9.160.25
interface Tunnel17
ip address 192.168.13.134 255.255.255.252
tunnel source FastEthernet0/1
tunnel destination 10.9.160.25
interface FastEthernet0/0
ip address 192.168.14.252 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
ip address 10.2.68.25 255.255.255.0
duplex auto
speed auto
interface Serial0/2/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn overlap-receiving
isdn incoming-voice voice
no cdp enable
router eigrp 201
network 172.25.10.0 0.0.0.255
network 192.168.14.0
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 10.9.160.0 255.255.255.0 10.2.68.1
ip route 10.128.0.69 255.255.255.255 Tunnel14
ip route 192.168.2.1 255.255.255.255 192.168.13.129
ip route 192.168.17.0 255.255.255.0 Tunnel14
tftp-server flash:SCCP11.9-2-1S.loads
tftp-server flash:jar11sccp.9-2-1TH1-13.sbn
tftp-server flash:dsp11.9-2-1TH1-13.sbn
tftp-server flash:cvm11sccp.9-2-1TH1-13.sbn
tftp-server flash:cnu11.9-2-1TH1-13.sbn
tftp-server flash:apps11.9-2-1TH1-13.sbn
control-plane
voice-port 0/0/0
caller-id enable
voice-port 0/0/1
voice-port 0/0/2
supervisory disconnect dualtone mid-call
dial-type pulse
disc_pi_off
output attenuation 1
echo-cancel coverage 32
timeouts call-disconnect 5
timeouts wait-release 1
timing hookflash-out 50
timing sup-disconnect 50
connection plar 600
caller-id enable
voice-port 0/0/3
caller-id enable
voice-port 0/2/0:15
mgcp profile default
dial-peer voice 1 pots
description connection-to-PBX
destination-pattern 0....
direct-inward-dial
port 0/2/0:15
forward-digits 4
dial-peer voice 10 voip
destination-pattern ...
session target ipv4:192.168.13.129
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 20 pots
description FXO-K
destination-pattern 9T
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
direct-inward-dial
port 0/0/2
prefix 9
dial-peer voice 30 pots
description FXO-K2
destination-pattern 9T
direct-inward-dial
port 0/0/1
prefix 9
telephony-service
max-ephones 20
max-dn 100
ip source-address 192.168.14.252 port 2000
cnf-file location flash:
load 7911 term11.default.loads
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1
number 770
line con 0
line aux 0
line 1/0 1/15
line vty 0 4
login local
transport input telnet
scheduler allocate 20000 1000
end

Having looked at your spreadsheet I see you're failing H323 transfers back to your ISDN system, but only under certain circumstances. Quite why, I'm not sure, possibly because you haven't codec defined on your H323 dial peers. or it could be something else
I think you may be able to work around the problem by adding
" supplementary-service h450.12 " under voice service voip on your CME router as a quick fix.
reference
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmetrans.html#wpxref44614
worth a try
Adam

Changing external Caller ID over a SIP Trunk to SIP Provider

I am working with a client and when they place calls out to any external user they have the wrong name showing on the external caller ID.
I have spoken with the SIP provider and apparently they want us to pass the CNAM, or rather they have it setup for us to do this.
I opened a case with Cisco and the TAC engineer said the provider has to do this because it cannot be done from CUCM or the gateway.
For example, it says right now "location A" for external calls and I want to change this to say "location B" .
Is this even possible?

what is the call flow? did you check the caller name in SIP trunk configuration?

Cisco 7942 + SIP Provider

Hello!
Can the Cisco 7942 with SIP Firmware used as standalone SIP device?
I mean can it works with SIP provider through NAT, like it can Cisco SPA-303?

There has been a discussion on this before.
https://supportforums.cisco.com/discussion/11955621/register-cisco-phone-7942-external-voip-provider
However, there was no conclusion to it.
This discussion here talked about registering 7942 with Asterisk.
http://www.experts-exchange.com/Networking/Telecommunications/IP_Telephony/Q_26895490.html
Since Asterisk is a 3rd party PBX, this shows that the phone CAN register with SIP firmware with a Provider. However, you will have to work extensively with the provider to get this done.
For instance, you need to create a custom cnf.xml file for the phone to download. To do this you'll need to copy the configuration from the CUCM, and then modify it as per your needs. Apart from this, the firmware files should also be located on the TFTP server that you're pointing to on the phone.
Also, you need to make sure that the provider doesn't have any mechanism on their side to block messages going out from the phone to their end. Packet captures would help you here.
There isn't a guarantee that this would work, but you can definitely try it.
Thanks

Calls from Sip Trunk to UC540 and then to CUE returned ** Service Unavailable**

Hi to all
i have something strange here and i need your assistance
Call Flow:
Sip trunk-->UC540--> CUE
When calls coming to UC540 from outside and then going to cue then we send back service unavailable.I made a translation and i sent directly the incoming calls to CUE
The same behavior is also if i send the calls to dummy number and then from there set forward all to voice mail.
Incoming voicemail is working fine
Incoming calls to phones also ok
Uc540: 8.6
CUE: 8.6.5
A number: 99999999
B number: 22777777
Voice Mail Number:111
Attached is the trace
i see that we hit the correct dial peers .
I have enable only trancoder since MTP is not register ( don't know why , but i don't think also that is necessary..
voice service voip
ip address trusted list
ipv4 172.16.80.0 255.255.255.0
ipv4 172.16.81.0 255.255.255.0
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
supplementary-service media-renegotiate
sip
no update-callerid
dial-peer voice 1000 voip
description **SIP TRUNK**
translation-profile incoming SIP-INCOMING
translation-profile outgoing SIP-OUTGOING
destination-pattern 9T
modem passthrough nse codec g711alaw
session protocol sipv2
session target sip-server
incoming called-number .T
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
fax-relay ecm disable
no fax-relay sg3-to-g3
fax rate 9600
fax protocol pass-through g711alaw
no vad
dial-peer voice 2001 voip
description ** cue voicemail pilot number **
destination-pattern 111
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number 111
no voice-class sip outbound-proxy
dtmf-relay sip-notify
codec g711ulaw
no vad
Regards
chrysostomos

Hi
Interface                  IP-Address      OK? Method Status                Protocol
FastEthernet0/0            unassigned      YES NVRAM up                    up
FastEthernet0/0.10         192.168.0.10    YES DHCP   up                    up   ----> For internet
FastEthernet0/0.20         10.151.5.130    YES NVRAM up                    up ------> For sip trunk
In0/0                      10.1.10.2       YES unset up                    up    --------> default gw for cue
Vlan1                      unassigned      YES unset up                    up
Vlan100                    unassigned      YES unset up                    up
Vlan200                    unassigned      YES unset up                    down
Vlan300                    unassigned      YES unset up                    down
NVI0                       10.1.10.2       YES unset up                    up
BVI1                       192.168.20.1    YES NVRAM up                    up
BVI100                     10.1.1.1        YES NVRAM up                    up   ---------> ip for cme
Loopback0                  10.1.10.2       YES NVRAM up                    up   ------> default gw for cue
dial-peer voice 2001 voip
description ** cue voicemail pilot number **
destination-pattern 111
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
incoming called-number 111
no voice-class sip outbound-proxy
voice-class sip bind control source-interface BVI100
voice-class sip bind media source-interface BVI100
dtmf-relay sip-notify
codec g711ulaw
no vad
interface FastEthernet0/0.10
description **FOR INTERNET**
encapsulation dot1Q 10
ip address dhcp
ip access-group 105 in
ip nat outside
ip inspect SDM_LOW out
ip virtual-reassembly in
interface FastEthernet0/0.20
description **FOR SIP TRUNK WITH ISP**
encapsulation dot1Q 20
ip address 10.151.5.130 255.255.255.240
ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
ping 10.1.10.1 source bvi100
Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 10.1.10.1, timeout is 2 seconds:
Packet sent with a source address of 10.1.1.1
Success rate is 100 percent (5/5), round-trip min/avg/max = 1/1/1 ms
I have bind the interface of cme ( 10.1.1.1) but the call fails again
Attached is the trace
Anything to advice?

Cisco CME and Calls through SIP provider

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