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

Similar Messages

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

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

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.

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?

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

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,
.wan

Hello 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

Jabra headset and Cisco 7942 Aux port

Hi ,
Two of our employees are using Jabra headsets. They are currently working over the analog headset port, but they would like to use the digital converter cable the headset came with to improve performance. How should I turned it on to utilize the auxiliary port in behind the phone.
Head model: Jabra GN9300e
Phone model: 7942
Any help would be much appreciated.
Saima

Hi Saima,
With the Cisco 7942, you would leverage the setting under
the product specific porting of the config via device>phone
and set Wireless Headset Hookswitch Control to Enabled
http://headsetplus.com/PDF/Jabra-EHS-guide.pdf
Cheers!
Rob
"And if I should fall behind
Wait for me" - Springsteen

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

RT31P2 with SIP provider

I used this device for Nikotel and Vyke SIP accounts but it never got registered as ISP is blocking ports.
I then configured one PC with VPN to UK and shared that connection. Gave PC LAN card as gateway in RT31P2. It worked. SIP calls went fine.
My question is - how to configure VPN connection in RT31P2 to bypass ISP port blocking or how to give alternate port settings for SIP provider.

What is STUN server settings. Can this be used in anyway to bypass the ISP proxy and connect to the SIP server.
If ports are blocked is there any other way to use the device for any SIP provider. I had used the device once to make calls using Nikotel (SIP) network but the voice quality was not good. Then I found a version upgrade on the LinkSys site and upgraded the firmware. After this I was not able to get registered to Nikotel. I asked support@ Linksys for the old firmware but they could not send me as they never had in their archive.
I am not sure if the new firmware created a problem or the ISP changed anything.
I would appreciate if anyone could send me the old firmware 1.02 so that I could try it.

Cisco 877 router - Cisco IP phone won't register with SIP provider

Hi all,
I'm having a problem with a Cisco SPA504G phone not registering with the SIP carrier over the Internet. We've recently rolled out a Cisco 877 router onto a new NBN business connection and can't get the pre-configured IP phone to register.
When we tested the phone with the NBN-provided Netgear router, it worked fine, as it did with the previous Cisco 1841 router we were using on a different link.
The way it's setup is using VLANs to define the internal subnets, which are then assigned to the physical interfaces (since the 887 doesn't allow IP assignments to the interfaces directly).
VLAN 100 is the internal network and has a SBS2011 server – assigned to F0 – IP range is 192.168.1.0
VLAN 200 is the guest network and has Internet access only – assigned to F1 – IP range is 10.1.1.0
VLAN 500 is the WAN network and connects to the NBN upstream box – assigned to F3 – external IP address assigned by DHCP
I've been playing around with access lists, nat rules, basically everything in my limited Cisco knowledge to try and figure this out, but to no avail. I have even configured what I believe is unrestricted access to IP, UDP and TCP outbound and inbound to all VLANs and still can't get it to register.
Tried isolating the issue by creating a new VLAN and assigning it to the spare interface and basically allowing everything in and out, but still no luck.
The problem has to be something on the router – probably some small line of config I haven’t removed or added.
I am going to pull my hair out soon, so would really appreciate some assistance from the Cisco gurus out there.
My client has just purchased about 10 of these handsets from their provider so I need to fix this ASAP. The guy who provided them wasn't very helpful, and basically said I'm on my own once we tested using the NBN-provided Netgear router.
Happy to post my config as well.
Please help!!!!

Current configuration : 4912 bytes
version 15.1
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router1
boot-start-marker
boot-end-marker
no aaa new-model
memory-size iomem 10
crypto pki token default removal timeout 0
no ip source-route
ip dhcp excluded-address 10.1.1.1
ip dhcp pool GUEST
network 10.1.1.0 255.255.255.0
dns-server 10.1.1.1 203.50.2.71 139.130.4.4
default-router 10.1.1.1
ip cef
no ip domain lookup
ip domain name network.local
ip name-server 192.168.1.123
ip name-server 203.23.53.12
ip name-server 197.12.32.86
ip name-server 8.8.8.8
no ipv6 cef
license udi pid CISCO887VA-K9 sn FGL171220XY
username admin privilege 15 secret 5 $1$aNsm$N1BCQYkoi8gnURyvloYEX/
controller VDSL 0
interface Ethernet0
no ip address
shutdown
interface ATM0
no ip address
no atm ilmi-keepalive
bridge-group 10
pvc 8/35
interface FastEthernet0
description NAC - Internal network
switchport access vlan 100
no ip address
interface FastEthernet1
description NAC - Guest network
switchport access vlan 200
no ip address
interface FastEthernet2
no ip address
shutdown
interface FastEthernet3
description **** WAN Port ****
switchport access vlan 500
no ip address
interface Vlan1
no ip address
bridge-group 10
hold-queue 100 out
interface Vlan100
description NAC - Internal Vlan
ip address 192.168.1.1 255.255.255.0
ip access-group IN-100 in
ip access-group OUT-100 out
ip nat inside
ip virtual-reassembly in
interface Vlan200
description NAC - Guest Vlan
ip address 10.1.1.1 255.255.255.0
ip access-group IN-200 in
ip access-group OUT-200 out
ip nat inside
ip virtual-reassembly in
interface Vlan500
description **** WAN Vlan ****
ip address dhcp
ip nat outside
no ip virtual-reassembly in
no ip forward-protocol nd
ip http server
ip http access-class 23
ip http secure-server
ip dns server
ip nat inside source list NAT-100 interface Vlan500 overload
ip nat inside source list NAT-200 interface Vlan500 overload
ip nat inside source static tcp 192.168.1.123 25 interface Vlan500 25
ip nat inside source static tcp 192.168.1.123 443 interface Vlan500 443
ip nat inside source static tcp 192.168.1.123 3389 interface Vlan500 3399
ip nat inside source static tcp 192.168.1.123 80 interface Vlan500 80
ip nat inside source static tcp 192.168.1.123 4125 interface Vlan500 4125
ip nat inside source static tcp 192.168.1.124 3389 interface Vlan500 3390
ip nat inside source static tcp 192.168.1.123 987 interface Vlan500 987
ip nat inside source static tcp 192.168.1.123 1723 interface Vlan500 1723
ip route 0.0.0.0 0.0.0.0 55.234.52.43
ip access-list extended IN-100
permit udp any any range bootps bootpc
deny ip 10.1.1.0 0.0.0.255 any
permit ip 192.168.1.0 0.0.0.255 any
ip access-list extended IN-200
permit udp any any range bootps bootpc
permit ip 10.1.1.0 0.0.0.255 any
ip access-list extended NAT-100
deny ip 192.168.0.0 0.0.255.255 192.168.0.0 0.0.255.255
permit ip 192.168.1.0 0.0.0.255 any
ip access-list extended NAT-200
deny ip 10.1.0.0 0.0.255.255 10.1.0.0 0.0.255.255
permit ip 10.1.1.0 0.0.0.255 any
ip access-list extended OUT-100
permit udp any range bootps bootpc any
deny ip 10.1.1.0 0.0.0.255 any
permit ip any 192.168.1.0 0.0.0.255
ip access-list extended OUT-200
permit udp any range bootps bootpc any
deny ip 10.1.1.0 0.0.0.255 192.168.1.0 0.0.0.255
permit ip any 10.1.1.0 0.0.0.255
access-list 23 permit 59.23.164.52
access-list 23 permit 192.168.1.0 0.0.0.255
access-list 23 permit 10.1.1.0 0.0.0.255
access-list 23 permit 120.146.0.0 0.0.255.255
access-list 23 permit 149.185.12.0 0.0.0.255
access-list 23 permit 110.44.28.0 0.0.0.255
access-list 23 permit 110.44.26.0 0.0.0.255
access-list 23 permit 103.25.212.0 0.0.0.255
access-list 23 permit any
bridge 10 protocol ieee
banner motd ^C
* Authorized personnel only! *
^C
line con 0
login local
no modem enable
line aux 0
line vty 0 4
password password01
login local
transport input all
end

Connect Cisco CallManager to external SIP provider

I need to connect my CUCM 5.1 with sip proxy on telco side.IP phones
will connect to CUCM.
The SIP server provide 90 lines with real numbers
Following is the scenario.
Cisco IP phones----------CUCM-------WAN connection to
telco---------SIP proxy server.
Can anybody explain me how this will work, what will be the
configurations and if CUCM has the capability to control the calls
between IP phones and SIP server.

Hi Asim
It is recommended to use CUBE (IP-IP gateway)
Look following url for configuration.
http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a00808ead0f.shtml
Regards..
Mahesh Dawar
www.cisco.com/go/pdihelpdesk

Can the Cisco 7942 phones work with Broadsoft using SIP

We have a few 7960 phones working with Broadsoft but we are having issues getting the 7942 phones to work. Is anyone using the 7942 with Broadsoft?

You can try and use the following guidelines but they natively are not supposed to support third party call control systems.
http://www.asterisk-peru.com/node/2227
http://www.888voip.com/configuring-cisco-7975-ip-phones-for-sip/

Problem connecting two trunks to sip provider using same CUBE

We need to connect two SIP trunks from service provider to Cisco CUCM 7.1 using CUBE “Cisco 2821”, SP using the following configuration:
First SIP PSTN Link Configuration(In-Out DID/DOD 218 7700 – 218 7799)
Customer IP Address =   10.196.191.158/30
SP IP Address = 10.196.191.157/30
Protocol= SIP
SIP Port = 5060
Transport Protocol=UDP
Voice Codec= G711 A-Law
DTMF = IN-Band DTMF without RFC2833
Signaling IP address = 10.201.20.49
IP Address 10.201.20.10 (Media IP) must be visible from IP PABX
Second SIP PSTN Link Configuration( Inbound Only 920009999)
Customer IP Address =   10.196.192.94/30
SP IP Address = 10.196.192.93/30
Protocol= SIP
SIP Port = 5060
Transport Protocol=UDP
Voice Codec= G711 A-Law
DTMF = IN-Band DTMF without RFC2833
SIP server IP address = 10.201.20.49
IP Address 10.201.20.10 (Media IP) must be visible from IP PABX
When we tried to configure both links on the same CUBE we faced two problems:
-          Routing issue, as we can’t route traffic using single CUBE through two different interfaces to the same destination “ i.e we have to configure static route commend (ip route 10.201.20.49 255.255.255.255 10.196.191.157 & ip route 10.201.20.49 255.255.255.255 10.196.192.93), sip traffic coming from one link can’t be sure to send it back to the same link.
-          SIP media & signaling control binding issue, as CUBE support sip binding using one interface only “one IP Address”, if we not using binding commands on the CUBE we can’t receive any calls though any link.
We have two options:
SP to send both traffic on the same trunk link
Or
Have another CUBE for the second link.
Attached network diagram.
Any solution?????
Regards,
Ahmed Rizk

I didn't mean NAT CUCM, I meant the interface towards it. But since you're using a single interface then yes that is what you NAT. You have a lot going on in that config. Probably a lot more than you need. Like I said you should work on this in two legs. CUBE to ITSP, and then CUCM to CUBE. You're trying to make the whole thing work in one shot which is going to cause you some headaches.
Install XLite free version. In the account settings set your UserID to a generic 10 digit phone number, domain to something generic, then at the bottom set the Proxy Address to the IP of your CUBE. The media ports will be negotiated dynamically between the CUBE and the ITSP. Since you said you're not registering you will also need to give the ITSP YOUR peer IP (this is how they secure the trunk) which is whatever IP you're sourcing from when you leave your network (what you're NAT'ing the CUBE to).
For testing, reduce your config to something like this:
voice service voip
allow-connections sip to sip
allow-connections h323 to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
signaling forward none
sip
dial-peer voice 10 voip
description CUBE_TO_ITSP
session protocol sipv2
session target ipv4:SIGNALING IP PROVIDED BY ITSP
destination-pattern [2-9].........
codec g711ulaw
dtmf-relay rtp-nte sip-notify
no vad
dial-peer voice 20 voip
description ITSP_TO_CUBE
destination-pattern .
session protocol sipv2
session target ipv4:Eventually your CUCM IP...for now set it to your computers IP.
codec g711ulaw
dtmf-relay rtp-nte sip-notify
no vad
Use XLite to place a phone call from your PC (if you have a mic and speakers you can have audio if the call connects). This should come pretty close to getting your outward leg established. Once you get this part working you can add in more codecs and translation profiles if you want. Let me know what happens. Include any debug or packet cap results if you can.

Cisco PIX SIP Issue

Hello All,
I am having an issue with running SIP through my Cisco Pix. A VOIP solution has just been installed, and softphones from the outside are trying to call in using SIP and are failing. The configuration is below. and the code is 6.3 (5). You'll see below that I have the no fixup protocol for sip, as the fixup wasn't working either. Is there something that needs to be configured that I'm missing or could this be a bug in the code? Any other show commands or debug commands I can provide if needed. The call manager server in the below config is 1.2.3.4. Thanks in advance for all your help, you guys are always so helpful.
XXXt# show ver
Cisco PIX Firewall Version 6.3(5)
Cisco PIX Device Manager Version 3.0(4)
Compiled on Thu 04-Aug-05 21:40 by morlee
XXX up 1 hour 45 mins
Hardware:   PIX-506E, 32 MB RAM, CPU Pentium II 300 MHz
Flash E28F640J3 @ 0x300, 8MB
BIOS Flash AM29F400B @ 0xfffd8000, 32KB
0: ethernet0: address is 001c.582b.3c65, irq 10
1: ethernet1: address is 001c.582b.3c66, irq 11
Licensed Features:
Failover:                    Disabled
VPN-DES:                     Enabled
VPN-3DES-AES:                Enabled
Maximum Physical Interfaces: 2
Maximum Interfaces:          4
Cut-through Proxy:           Enabled
Guards:                      Enabled
URL-filtering:               Enabled
Inside Hosts:                Unlimited
Throughput:                  Unlimited
IKE peers:                   Unlimited
This PIX has a Restricted (R) license.
XXXt# show run
: Saved
PIX Version 6.3(5)
interface ethernet0 auto
interface ethernet1 auto
nameif ethernet0 outside security0
nameif ethernet1 inside security100
enable password vQ0/erypfvYyzFoc encrypted
passwd vQ0/erypfvYyzFoc encrypted
hostname DTPIX35thst
domain-name digitaltransitions.com
fixup protocol dns maximum-length 512
fixup protocol ftp 21
fixup protocol h323 h225 1720
fixup protocol h323 ras 1718-1719
fixup protocol http 80
fixup protocol rsh 514
fixup protocol rtsp 554
no fixup protocol sip 5060
no fixup protocol sip udp 5060
fixup protocol skinny 2000
fixup protocol smtp 25
fixup protocol sqlnet 1521
fixup protocol tftp 69
names
access-list out_in permit udp any host 1.2.3.4 eq 5060
access-list out_in permit tcp any host 1.2.3.43 eq 5060
pager lines 24
logging on
logging buffered informational
logging trap informational
logging queue 2048
mtu outside 1500
mtu inside 1500
ip address outside 4.34.119.130 255.255.255.248
ip address inside 192.168.1.1 255.255.255.0
ip audit info action alarm
ip audit attack action alarm
ip local pool vpn_pool 192.168.100.50-192.168.100.75
pdm location 192.168.1.250 255.255.255.255 inside
pdm location 192.168.1.252 255.255.255.255 inside
pdm location 65.215.8.100 255.255.255.255 inside
pdm location 192.168.100.0 255.255.255.0 outside
pdm logging informational 100
pdm history enable
arp timeout 14400
global (outside) 1 interface
nat (inside) 0 access-list nonat
nat (inside) 1 0.0.0.0 0.0.0.0 0 0
static (inside,outside) 1.2.3.4 172.20.1.2 netmask 255.255.255.255 0 0
access-group out_in in interface outside
timeout xlate 0:05:00
timeout conn 1:00:00 half-closed 0:10:00 udp 0:02:00 rpc 0:10:00 h225 1:00:00
timeout h323 0:05:00 mgcp 0:05:00 sip 0:00:00 sip_media 0:00:00
timeout sip-disconnect 0:02:00 sip-invite 0:03:00
timeout uauth 0:05:00 absolute
aaa-server TACACS+ protocol tacacs+
aaa-server TACACS+ max-failed-attempts 3
aaa-server TACACS+ deadtime 10
aaa-server RADIUS protocol radius
aaa-server RADIUS max-failed-attempts 3
aaa-server RADIUS deadtime 10
aaa-server LOCAL protocol local
aaa authentication ssh console LOCAL
http server enable
http 199.96.104.108 255.255.255.255 outside
http 192.168.1.0 255.255.255.0 inside
no snmp-server location
no snmp-server contact
snmp-server community public
no snmp-server enable traps
floodguard enable

Hi Jumora,
No need to troubleshoot this direct issue anymore. The client will be upgrading to an ASA 5505. Is there anything you may know of before I configure the ASA that I need to do to allow SIP through with no issues? Thanks again Jumora

Cisco 7942 + SIP Provider

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