Re: Multiple authentication username in sip-ua CME

bagnolini · ‎01-16-2012

I've some problem with sip account with my CME (c2900-universalk9-mz.SPA.152-1.T.bin)

If I configure 1 single account per time everything is ok:

NUMBER1

sip-ua

credentials username 60510988031 password xxxxx realm voip2.voipvox.it

authentication username 60510988031 password xxxxx

no remote-party-id

disable-early-media 180

retry invite 2

retry response 3

retry bye 2

retry cancel 2

retry register 10

no timers hold

registrar 1 dns:voip2.voipvox.it expires 3600

sip-server ipv4:178.250.64.102:5060

!

cisco2901#show sip-ua register status

--------------------- Registrar-Index 1 ---------------------

Line peer expires(sec) registered P-Associ-URI

================================ ========== ============ ========== ============

60510988031 -1 2850 yes

NUMBER 2

sip-ua

credentials username 60516951951 password xxxxx realm voip2.voipvox.it

authentication username 60516951951 password xxxx

no remote-party-id

disable-early-media 180

retry invite 2

retry response 3

retry bye 2

retry cancel 2

retry register 10

no timers hold

registrar 1 dns:voip2.voipvox.it expires 3600

sip-server ipv4:178.250.64.102:5060

!

cisco2901#show sip-ua register status

--------------------- Registrar-Index 1 ---------------------

Line peer expires(sec) registered P-Associ-URI

================================ ========== ============ ========== ============

60516951951 -1 2872 yes

The problem is when I try to register both number (I must put the realm under at least one authentication username!!).

sip-ua

credentials username 60516951951 password xxxx realm voip2.voipvox.it

credentials username 60510988031 password yyyy realm voip2.voipvox.it

authentication username 60516951951 password xxxxx realm voip2.voipvox.it

authentication username 60510988031 password yyyyy

no remote-party-id

disable-early-media 180

retry invite 2

retry response 3

retry bye 2

retry cancel 2

retry register 10

no timers hold

registrar 1 dns:voip2.voipvox.it expires 3600

sip-server ipv4:178.250.64.102:5060

!

cisco2901#show sip-ua register status

--------------------- Registrar-Index 1 ---------------------

Line peer expires(sec) registered P-Associ-URI

================================ ========== ============ ========== ============

60510988031 -1 2736 yes

60516951951 -1 36 no

paolo bevilacqua · ‎01-16-2012

Create some voip DPs with destionation-pattern the number to register. Configure username and password in there. CME will register them using separate credentials. Then you will have a translation profile and rule to send the call to the right extension.

bagnolini · ‎01-16-2012

I created this dial-peer but the result is the same

dial-peer voice 14 voip

description Sip-0516951951

translation-profile outgoing PSTN_Outgoing2

destination-pattern 60516951951

session protocol sipv2

session target sip-server

session transport udp

dtmf-relay rtp-nte

codec g711alaw

authentication username 60516951951 password yyyyy realm voip2.voipvox.it

!

sip-ua

credentials username 60510988031 password xxxx realm voip2.voipvox.it

credentials username 60516951951 password yyyy realm voip2.voipvox.it

authentication username 60510988031 password xxxx

no remote-party-id

disable-early-media 180

retry invite 2

retry response 3

retry bye 2

retry cancel 2

retry register 10

no timers hold

registrar 1 dns:voip2.voipvox.it expires 3600

sip-server ipv4:178.250.64.102:5060

!

cisco2901#show sip register status

--------------------- Registrar-Index 1 ---------------------

Line peer expires(sec) registered P-Associ-URI

================================ ========== ============ ========== ============

60510988031 -1 862 yes

60516951951 -1 159 no

paolo bevilacqua · ‎01-16-2012

Sorry, I meant a pots dial-peer. Associate to any port, it won't be used anyway. Then you can check with "debug ccsip message" that it triggers registration.

bagnolini · ‎01-18-2012

Thank you, now I'm able to register my account and make inbound call but I've some problem to make outgoing call.

I've attached the log about incoming and outgoing call.

This is my config:

voice service voip

no ip address trusted authenticate

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

no supplementary-service sip moved-temporarily

redirect ip2ip

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

h323

modem passthrough nse codec g711alaw redundancy

sip

bind control source-interface GigabitEthernet0/0

bind media source-interface GigabitEthernet0/0

registrar server

!

voice class uri 1 sip

pattern ^6051098xxxx

!

voice class uri 2 sip

pattern ^6051695yyyy

voice translation-rule 1

rule 2 /^32\([0,3,7-9]\)\(.*\)/ /032\1\2/

rule 3 /^33\([0,3-9]\)\(.*\)/ /033\1\2/

rule 4 /^34\([0,3,6-9]\)\(.*\)/ /034\1\2/

rule 6 /^36\([0,3,6,8]\)\(.*\)/ /036\1\2/

rule 8 /^38\([0,3,8,9]\)\(.*\)/ /038\1\2/

rule 9 /^39\([0-3]\)\(.*\)/ /039\1\2/

rule 15 /^\(\)/ /00\1/

!

voice translation-rule 2

rule 1 /^9\(.*\)/ /\1/

!

voice translation-rule 3

rule 1 /^.*/ /6051098xxxx/

!

voice translation-rule 4

rule 1 /^8\(.*\)/ /\1/

!

voice translation-rule 5

rule 1 /^.*/ /6051695yyyy/

!

voice translation-profile AddZero-Telecom

translate calling 1

!

voice translation-profile PSTN_Outgoing

translate calling 3

translate called 2

!

voice translation-profile PSTN_Outgoing2

translate calling 5

translate called 4

!

dial-peer voice 11 voip

description Chiamate in uscita verso Sip

translation-profile outgoing PSTN_Outgoing

destination-pattern 9T

session protocol sipv2

session target sip-server

session transport udp

dtmf-relay rtp-nte

codec g711alaw

!

dial-peer voice 12 voip

description Ingresso Numero 0510988xxx

translation-profile incoming AddZero-Telecom

session protocol sipv2

session target sip-server

session transport tcp

incoming uri request 1

dtmf-relay rtp-nte

codec g711alaw

!

dial-peer voice 13 voip

description Ingresso Numero 051695yyyy

translation-profile incoming AddZero-Telecom

session protocol sipv2

session target sip-server

session transport tcp

incoming uri request 2

dtmf-relay rtp-nte

codec g711alaw

!

dial-peer voice 14 voip

description Chiamate in uscita verso Sip-051695yyyy

translation-profile outgoing PSTN_Outgoing2

destination-pattern 8T

session protocol sipv2

session target sip-server

session transport tcp

no voice-class sip pass-thru headers

no voice-class sip pass-thru content unsupp

dtmf-relay rtp-nte

codec g711alaw

!

dial-peer voice 15 pots

destination-pattern 6051695xxxx

authentication username 60516951951 password xxxxx

port 0/0/0

!

dial-peer voice 16 pots

destination-pattern 6051098yyyy

authentication username 60510988031 password yyyyy

port 0/0/0

sip-ua

credentials username 6051098xxxx password xxxxxx realm voip2.voipvox.it

credentials username 6051695yyyy password yyyyyy realm voip2.voipvox.it

no remote-party-id

disable-early-media 180

retry invite 2

retry response 3

retry bye 2

retry cancel 2

retry register 10

no timers hold

registrar 1 dns:voip2.voipvox.it expires 3600

sip-server ipv4:178.250.64.102:5060

cisco2901#show sip-ua register status

--------------------- Registrar-Index 1 ---------------------

Line peer expires(sec) registered P-Associ-URI

================================ ========== ============ ========== ============

6051098xxxx 16 301 yes

6051695yyyy 15 301 yes

paolo bevilacqua · ‎01-18-2012

Take "debug ccsip message" and no other debug.

Post directly, no need to use attachments.

Please remember to rate useful posts clicking on the stars below.

bagnolini · ‎01-18-2012

OUTGOING CALL (FAIL)

Jan 18 12:24:15.497 CET: //6879/C83C76929E4E/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:3471526635@178.250.64.102:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.0.13:5060;branch=z9hG4bK97483

From: "Vem" <60510988031>;tag=38EA230-22A9

To: <3471526635>

Date: Wed, 18 Jan 2012 11:24:15 GMT

Call-ID: C9D601FE-40FD11E1-9E53D66A-796E8D54@192.168.0.13

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE: 1800

Cisco-Guid: 3359405714-1090327009-2655966826-2037288276

User-Agent: Cisco-SIPGateway/IOS-15.2(1)T,

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1326885855

Contact: <60510988031>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 319

v=0

o=CiscoSystemsSIP-GW-UserAgent 6859 411 IN IP4 192.168.0.13

s=SIP Call

c=IN IP4 192.168.0.13

t=0 0

m=audio 19762 RTP/AVP 8 100 101 19

c=IN IP4 192.168.0.13

a=rtpmap:8 PCMA/8000

a=rtpmap:100 X-NSE/8000

a=fmtp:100 192-194

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:19 CN/8000

a=ptime:20

Jan 18 12:24:15.553 CET: //6879/C83C76929E4E/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.0.13:5060;branch=z9hG4bK97483;received=192.168.0.13

From: "Vem" <60510988031>;tag=38EA230-22A9

To: <3471526635>;tag=as6e65c6e9

Call-ID: C9D601FE-40FD11E1-9E53D66A-796E8D54@192.168.0.13

CSeq: 101 INVITE

Server: Switch24

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1e3d0223"

Content-Length: 0

Jan 18 12:24:15.557 CET: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:3471526635@178.250.64.102:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.0.13:5060;branch=z9hG4bK97483

From: "Vem" <60510988031>;tag=38EA230-22A9

To: <3471526635>;tag=as6e65c6e9

Date: Wed, 18 Jan 2012 11:24:15 GMT

Call-ID: C9D601FE-40FD11E1-9E53D66A-796E8D54@192.168.0.13

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

INCOMING CALL (OK)

Jan 18 12:25:42.785 CET: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:60510988031@192.168.0.13:5060 SIP/2.0

Via: SIP/2.0/UDP 178.250.64.102:5060;branch=z9hG4bK1a149b4e;rport

Max-Forwards: 70

From: "3409379407" <3409379407>;tag=as362d0249

To: <60510988031>

Contact: <3409379407>

Call-ID: 45e4a34f50c989e571915ee620508033@178.250.64.102

CSeq: 102 INVITE

User-Agent: Switch24

Date: Wed, 18 Jan 2012 11:25:40 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 298

v=0

o=root 35561604 35561604 IN IP4 178.250.64.102

s=Switch24

c=IN IP4 178.250.64.102

t=0 0

m=audio 18492 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

Jan 18 12:25:42.801 CET: //6898/FDDD06379E70/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 178.250.64.102:5060;branch=z9hG4bK1a149b4e;rport

From: "3409379407" <3409379407>;tag=as362d0249

To: <60510988031>

Date: Wed, 18 Jan 2012 11:25:42 GMT

Call-ID: 45e4a34f50c989e571915ee620508033@178.250.64.102

CSeq: 102 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.2(1)T,

Content-Length: 0

Jan 18 12:25:42.801 CET: //6898/FDDD06379E70/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 178.250.64.102:5060;branch=z9hG4bK1a149b4e;rport

From: "3409379407" <3409379407>;tag=as362d0249

To: <60510988031>;tag=38FF738-80E

Date: Wed, 18 Jan 2012 11:25:42 GMT

Call-ID: 45e4a34f50c989e571915ee620508033@178.250.64.102

CSeq: 102 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Contact: <60510988031>

Server: Cisco-SIPGateway/IOS-15.2(1)T,

Content-Length: 0

Jan 18 12:25:43.041 CET: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:60510988031@192.168.0.13:5060 SIP/2.0

Via: SIP/2.0/UDP 178.250.64.102:5060;branch=z9hG4bK1a149b4e;rport

Max-Forwards: 70

From: "3409379407" <3409379407>;tag=as362d0249

To: <60510988031>

Contact: <3409379407>

Call-ID: 45e4a34f50c989e571915ee620508033@178.250.64.102

CSeq: 102 INVITE

User-Agent: Switch24

Date: Wed, 18 Jan 2012 11:25:40 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 298

v=0

o=root 35561604 35561604 IN IP4 178.250.64.102

s=Switch24

c=IN IP4 178.250.64.102

t=0 0

m=audio 18492 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

Jan 18 12:25:43.045 CET: //6898/FDDD06379E70/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 178.250.64.102:5060;branch=z9hG4bK1a149b4e;rport

From: "3409379407" <3409379407>;tag=as362d0249

To: <60510988031>;tag=38FF738-80E

Date: Wed, 18 Jan 2012 11:25:43 GMT

Call-ID: 45e4a34f50c989e571915ee620508033@178.250.64.102

CSeq: 102 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Contact: <60510988031>

Server: Cisco-SIPGateway/IOS-15.2(1)T,

Content-Length: 0

Jan 18 12:25:43.181 CET: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:60510988031@192.168.0.13:5060 SIP/2.0

Via: SIP/2.0/UDP 178.250.64.102:5060;branch=z9hG4bK1a149b4e;rport

Max-Forwards: 70

From: "3409379407" <3409379407>;tag=as362d0249

To: <60510988031>

Contact: <3409379407>

Call-ID: 45e4a34f50c989e571915ee620508033@178.250.64.102

CSeq: 102 INVITE

User-Agent: Switch24

Date: Wed, 18 Jan 2012 11:25:40 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 298

v=0

o=root 35561604 35561604 IN IP4 178.250.64.102

s=Switch24

c=IN IP4 178.250.64.102

t=0 0

m=audio 18492 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

Jan 18 12:25:43.181 CET: //6898/FDDD06379E70/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 178.250.64.102:5060;branch=z9hG4bK1a149b4e;rport

From: "3409379407" <3409379407>;tag=as362d0249

To: <60510988031>;tag=38FF738-80E

Date: Wed, 18 Jan 2012 11:25:43 GMT

Call-ID: 45e4a34f50c989e571915ee620508033@178.250.64.102

CSeq: 102 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Contact: <60510988031>

Server: Cisco-SIPGateway/IOS-15.2(1)T,

Content-Length: 0

Jan 18 12:25:44.557 CET: //6898/FDDD06379E70/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 178.250.64.102:5060;branch=z9hG4bK1a149b4e;rport

From: "3409379407" <3409379407>;tag=as362d0249

To: <60510988031>;tag=38FF738-80E

Date: Wed, 18 Jan 2012 11:25:43 GMT

Call-ID: 45e4a34f50c989e571915ee620508033@178.250.64.102

CSeq: 102 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Contact: <60510988031>

Supported: replaces

Supported: sdp-anat

Server: Cisco-SIPGateway/IOS-15.2(1)T,

Supported: timer

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 247

v=0

o=CiscoSystemsSIP-GW-UserAgent 9947 3995 IN IP4 192.168.0.13

s=SIP Call

c=IN IP4 192.168.0.13

t=0 0

m=audio 28252 RTP/AVP 8 101

c=IN IP4 192.168.0.13

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

Jan 18 12:25:44.609 CET: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:60510988031@192.168.0.13:5060 SIP/2.0

Via: SIP/2.0/UDP 178.250.64.102:5060;branch=z9hG4bK18f631cc;rport

Max-Forwards: 70

From: "3409379407" <3409379407>;tag=as362d0249

To: <60510988031>;tag=38FF738-80E

Contact: <3409379407>

Call-ID: 45e4a34f50c989e571915ee620508033@178.250.64.102

CSeq: 102 ACK

User-Agent: Switch24

Content-Length: 0

Jan 18 12:25:45.965 CET: //6898/FDDD06379E70/SIP/Msg/ccsipDisplayMsg:

Sent:

BYE sip:3409379407@178.250.64.102:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.0.13:5060;branch=z9hG4bK98D06

From: <60510988031>;tag=38FF738-80E

To: "3409379407" <3409379407>;tag=as362d0249

Date: Wed, 18 Jan 2012 11:25:44 GMT

Call-ID: 45e4a34f50c989e571915ee620508033@178.250.64.102

User-Agent: Cisco-SIPGateway/IOS-15.2(1)T,

Max-Forwards: 70

Timestamp: 1326885945

CSeq: 101 BYE

Reason: Q.850;cause=16

P-RTP-Stat: PS=54,OS=8640,PR=65,OR=10400,PL=0,JI=0,LA=0,DU=1

Content-Length: 0

paolo bevilacqua · ‎01-18-2012

You do not have configured authentication under sip-ua.

Please try to be more fair when rating, because I've 100% solved yuor problem above, but was rated only '4'.

bagnolini · ‎01-18-2012

I can configure only one authentication per-time under sip-ua, it works only one number!

If I try to configure the second number, it takes the place of first number.

sip-ua

credentials username 6051098xxx password xxxx realm voip2.voipvox.it

credentials username 6051695yyy password xxxx realm voip2.voipvox.it

authentication username 6051098xxx password xxxx

no remote-party-id

disable-early-media 180

retry invite 2

retry response 3

retry bye 2

retry cancel 2

retry register 10

no timers hold

registrar 1 dns:voip2.voipvox.it expires 600

sip-server ipv4:178.250.64.102:5060

!

paolo bevilacqua · ‎01-18-2012

Correct. Already years ago I asked Cisco if there was any plan to support multiple authentication for outgoig calls, but their answer was not satisfying at all.

Eugene Zuevski · ‎04-15-2015

Have you resolved your issue with outgoing calls with multiple authentication?