Cisco CUBE multi sip trunks

[danscout01]

Hey,

I was wondering if its possible (and how) you connect a Cisco Cube to multiple sip trunks (different providers).

I have 3 sip providers one in Australia, one in the US and one in the UK, but I havent been able to find any descriptions how you do this.

I have one connected with "sip-ua" when i try to connect the second one it doesnt seem like its working.

I mostly find links to description about "Configuring Multiple Registrars on SIP Trunks" but that describes about multiple "lines" on one sip trunk.

Anyone have a good link to a description about multiple sip trunks on cube ?

Best regards

Danscout

[Ayodeji Okanlawon]

This is definitely possible. However you will need to define multiple interfaces to connect to these three providers.

You will then need to use your sip bind commands at the dial-peer level. E.g

interface loopback1----------------------------------------Interface pointing to SIP provider 1
ip address 10.10.10.1 255.255.255.0

interface loopback2-----------------------------------------Interface pointing to SIP provider 2
ip address 20.20.20.1 255.255.255.0

dial-peer voice 10 voip
description “Primary path to SIP SP-1”
destination-pattern 91[2-9]..[2-9]......
session protocol sipv2
session target ipv4:10.10.10.2
voice-class sip options-keepalive
voice-class sip bind control source-interface loopback1
voice-class sip bind media source-interface loopback1

dial-peer voice 20 voip
description “Secondary path to SIP SP-2”
destination-pattern 91[2-9]..[2-9]......
session protocol sipv2
session target ipv4:20.20.20.2
preference 2
voice-class sip options-keepalive
voice-class sip bind control source-interface loopback2
voice-class sip bind media source-interface loopback2

If your DDIs are provisioned differently on these providers, you will need to use sip profiles to authenticate before you will be allowed to place outbound calls through them.

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

[danscout01]

Hi,

Thanks for the reply i did try it, but unfortunately i dont have 3 spare ip adresses i can use on the outside router (cube).

I got stuck at the sip profiles, not sure how i would make them do the authentication to the provider, so i tried something from the "multiple registrars on sip trunks" page, use the whole night on it, but i think i got them to register to the provider.

When i look at the 2 providers status page i can see that they are registered and that the device type is "Cisco-SIPGateway/IOS-15-3-1-T",  the ip my router is at and a SIP ID of the username.

I can make calls out though the DK provider, that so far is only to free call numbers and works fine. I currently dont have a incomming number yet.

On the UK side I have a incomming number, when i try to call that nothing really happens, it rings but then time out after a few rings.

Any ideas ?

Here is the config im using, hope i remembered to include everything

voice translation-rule 10

rule 1 /1.../ /654321/

voice translation-rule 1

rule 1 /^0/ //

voice translation-rule 20

rule 1 /^123456$/ /1004/

voice translation-profile provider1

translate called 20

voice translation-profile provider2

translate calling 10

translate called 1

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

dial-peer voice 1 voip

description *** Incoming calls for primary UCM ***

preference 4

destination-pattern .T

progress_ind setup enable 3

no modem passthrough

session target ipv4:192.168.11.37

voice-class codec 1

voice-class h323 1

dtmf-relay h245-alphanumeric

fax rate disable

no vad

dial-peer voice 10 voip

description ** SIP trunk to SP-1 (DK) **

translation-profile outgoing provider2

destination-pattern 0[2-9].......$

session protocol sipv2

session target sip-server

voice-class codec 1

no voice-class sip localhost

dtmf-relay rtp-nte

no vad

authentication username 654321 password YYYYY realm provider.dk

dial-peer voice 20 voip

description ** SIP trunk to SP-2 (UK) **

translation-profile incoming provider1

destination-pattern 00044T

session protocol sipv2

session target registrar

voice-class codec 1

no voice-class sip localhost

dtmf-relay rtp-nte

no vad

authentication username 123456 password YYYYY realm provider.co.uk

sip-ua

credentials number 123456 username 123456 password XXXXX realm provider.dk

credentials number 654321 username 654321 password XXXXX realm provider.co.uk

retry register 5

registrar 1 dns:provider.co.uk expires 3600 auth-realm provider.co.uk

registrar 2 dns:provider.com expires 3600 auth-realm provider.dk

sip-server dns:provider.dk

CR0#sh sip-ua register status

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

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

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

123456                           -1         25           no

654321                           -1         242          yes

--------------------- Registrar-Index  2 ---------------------

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

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

123456                           -1         682          yes

654321                           -1         2            no

[chrysostomos1980]

Hi

I would recomended to use  sip dial peer for the communication with cucm and not h323.(Create a sip trunk to cucm)

Also  can you send the voice service voip configuration?

[danscout01]

Hi,

Thanks for the reply, just wondering, what is the benefit using sip instead of the h323 ?

Here is voice service config

voice service voip

ip address trusted list

  ipv4 1.2.3.4 255.255.255.255

  ipv4 4.3.2.1 255.255.255.255

address-hiding

mode border-element

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

redirect ip2ip

signaling forward unconditional

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

h323

  no h225 timeout keepalive

  call preserve

modem passthrough nse codec g711alaw redundancy maximum-sessions 20

sip

  registrar server

  registration passthrough

[Bekzod Fakhriddinov]

Hi, that config is for outbound dial-peers, how the inbound from 2 ISP will look like ?

Thank you

[Ayodeji Okanlawon]

I agree with Chris. Use sip on the inbound leg to cucm. That means you need to configure a sip trunk from cucm to your cube.

Second you will need to send debug ccsip messaged, debug VoIP ccapi input.


Sent from Cisco Technical Support Android App

[danscout01]

Hi

Now it looks like this:

voice service voip

ip address trusted list

  ipv4 87.54.25.114 255.255.255.255

  ipv4 217.10.79.23 255.255.255.255

  ipv4 192.168.11.37 255.255.255.255

address-hiding

mode border-element

allow-connections sip to sip

redirect ip2ip

signaling forward unconditional

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

modem passthrough nse codec g711alaw redundancy maximum-sessions 20

sip

  registrar server

  registration passthrough

dial-peer voice 1 voip

description *** Incoming calls for primary UCM ***

preference 4

destination-pattern .T

progress_ind setup enable 3

no modem passthrough

session protocol sipv2

session target ipv4:192.168.11.37

voice-class codec 1 

dtmf-relay h245-alphanumeric

fax rate disable

no vad

Cube 192.168.11.253 (inside ip) 1.2.3.4 (outside ip)

Cucm 192.168.11.37

--- Debug

CR0#sh deb

CCSIP SPI: SIP Call Statistics tracing is enabled       (filter is OFF)

CCSIP SPI: SIP Call Message tracing is enabled  (filter is OFF)

013766: Feb 13 09:39:27.854 CET: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

...term no mon

013767: Feb 13 09:39:32.258 CET: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:1664819@1.2.3.4:5060 SIP/2.0

Record-Route: <217.10.79.23>

Record-Route: <172.20.40.1>

Record-Route: <217.10.79.23>

Via: SIP/2.0/UDP 217.10.79.23:5060;branch=z9hG4bKfd4c.365b3434.0

Via: SIP/2.0/UDP 172.20.40.1;branch=z9hG4bKfd4c.365b3434.0

Via: SIP/2.0/UDP 217.10.79.23:5060;received=217.10.68.222;branch=z9hG4bK0b5f848f

Via: SIP/2.0/UDP 217.116.117.9:5060;received=217.116.117.9;branch=z9hG4bK0b5f848f;rport=5060

Max-Forwards: 67

From: "004512345678" <>004512345678@sipgate.co.uk>;tag=as0f6ef5c1

To: <>00441234567890@sipgate.co.uk>

Contact: <004512345678>

Call-ID: 0df02a86702544633c24f2141becd7ca@sipgate.co.uk

CSeq: 102 INVITE

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

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 467

v=0

o=root 1963565677 1963565677 IN IP4 217.116.117.9

s=sipgate VoIP GW

c=IN IP4 217.10.77.21

t=0 0

m=audio 59720 RTP/AVP 8 0 3 97 18 112 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:97 iLBC/8000

a=fmtp:97 mode=30

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:112 G726-32/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

a=direction:active

a=nortpproxy:yes

013768: Feb 13 09:39:32.270 CET: //17026/B8E5F72AAF9C/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 217.10.79.23:5060;branch=z9hG4bKfd4c.365b3434.0,SIP/2.0/UDP 172.20.40.1;branch=z9hG4bKfd4c.365b3434.0,SIP/2.0/UDP 217.10.79.23:5060;received=217.10.68.222;branch=z9hG4bK0b5f848f,SIP/2.0/UDP 217.116.117.9:5060;received=217.116.117.9;branch=z9hG4bK0b5f848f;rport=5060

From: "004512345678" <>004512345678@sipgate.co.uk>;tag=as0f6ef5c1

To: <>00441234567890@sipgate.co.uk>

Date: Wed, 13 Feb 2013 08:39:32 GMT

Call-ID: 0df02a86702544633c24f2141becd7ca@sipgate.co.uk

CSeq: 102 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.3.1.T

Content-Length: 0

013769: Feb 13 09:39:32.270 CET: //17027/B8E5F72AAF9C/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:1004@192.168.11.37:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.11.253:5060;branch=z9hG4bK185A14CC

Remote-Party-ID: "004512345678" <004512345678>;party=calling;screen=no;privacy=off

From: "004512345678" <004512345678>;tag=11780480-1DAE

To: <1004>

Date: Wed, 13 Feb 2013 08:39:32 GMT

Call-ID: B8E72F7A-74EF11E2-AFA2E149-3AD11FA7@192.168.11.253

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

Min-SE:  1800

Cisco-Guid: 3102078762-1961824738-2946294089-0986783655

User-Agent: Cisco-SIPGateway/IOS-15.3.1.T

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

CSeq: 101 INVITE

Timestamp: 1360744772

Contact: <004512345678>

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 66

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 208

v=0

o=CiscoSystemsSIP-GW-UserAgent 5346 483 IN IP4 192.168.11.253

s=SIP Call

c=IN IP4 192.168.11.253

t=0 0

m=audio 30796 RTP/AVP 8 0

c=IN IP4 192.168.11.253

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

013770: Feb 13 09:39:32.274 CET: //17027/B8E5F72AAF9C/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.11.253:5060;branch=z9hG4bK185A14CC

From: "004512345678" <004512345678>;tag=11780480-1DAE

To: <1004>

Date: Wed, 13 Feb 2013 08:39:32 GMT

Call-ID: B8E72F7A-74EF11E2-AFA2E149-3AD11FA7@192.168.11.253

CSeq: 101 INVITE

Allow-Events: presence

Content-Length: 0

013771: Feb 13 09:39:32.278 CET: //17027/B8E5F72AAF9C/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.11.253:5060;branch=z9hG4bK185A14CC

From: "004512345678" <004512345678>;tag=11780480-1DAE

To: <1004>;tag=2215~51ae08a6-d9b1-40ed-8048-61635a68d171-22773051

Date: Wed, 13 Feb 2013 08:39:32 GMT

Call-ID: B8E72F7A-74EF11E2-AFA2E149-3AD11FA7@192.168.11.253

CSeq: 101 INVITE

Allow-Events: presence

Reason: Q.850;cause=1

Content-Length: 0

013772: Feb 13 09:39:32.278 CET: //17027/B8E5F72AAF9C/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x0x328AD688

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           : 004512345678

Called Number            : 1004

Source IP Address (Sig  ): 192.168.11.253

Destn SIP Req Addr:Port  : 192.168.11.37:5060

Destn SIP Resp Addr:Port : 192.168.11.37:5060

Destination Name         : 192.168.11.37

013773: Feb 13 09:39:32.278 CET: //17027/B8E5F72AAF9C/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : No Codec  

Negotiated Codec Bytes   : 0

Nego. Codec payload      : 255 (tx), 255 (rx)

Negotiated Dtmf-relay    : 0

Dtmf-relay Payload       : 0 (tx), 0 (rx)

Source IP Address (Media): 192.168.11.253

Source IP Port    (Media): 30796

Destn  IP Address (Media):  -

Destn  IP Port    (Media): 0

Orig Destn IP Address:Port (Media): [ - ]:0

013774: Feb 13 09:39:32.278 CET: //17027/B8E5F72AAF9C/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 1

Disconnect Cause (SIP)   : 404

013775: Feb 13 09:39:32.282 CET: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:1004@192.168.11.37:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.11.253:5060;branch=z9hG4bK185A14CC

From: "004512345678" <004512345678>;tag=11780480-1DAE

To: <1004>;tag=2215~51ae08a6-d9b1-40ed-8048-61635a68d171-22773051

Date: Wed, 13 Feb 2013 08:39:32 GMT

Call-ID: B8E72F7A-74EF11E2-AFA2E149-3AD11FA7@192.168.11.253

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

013776: Feb 13 09:39:32.282 CET: //17026/B8E5F72AAF9C/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 217.10.79.23:5060;branch=z9hG4bKfd4c.365b3434.0,SIP/2.0/UDP 172.20.40.1;branch=z9hG4bKfd4c.365b3434.0,SIP/2.0/UDP 217.10.79.23:5060;received=217.10.68.222;branch=z9hG4bK0b5f848f,SIP/2.0/UDP 217.116.117.9:5060;received=217.116.117.9;branch=z9hG4bK0b5f848f;rport=5060

From: "004512345678" <>004512345678@sipgate.co.uk>;tag=as0f6ef5c1

To: <>00441234567890@sipgate.co.uk>;tag=11780488-1571

Date: Wed, 13 Feb 2013 08:39:32 GMT

Call-ID: 0df02a86702544633c24f2141becd7ca@sipgate.co.uk

CSeq: 102 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.3.1.T

Reason: Q.850;cause=1

Content-Length: 0

013777: Feb 13 09:39:32.310 CET: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:1664819@1.2.3.4:5060 SIP/2.0

Via: SIP/2.0/UDP 217.10.79.23:5060;branch=z9hG4bKfd4c.365b3434.0

Via: SIP/2.0/UDP 172.20.40.1;branch=z9hG4bKfd4c.365b3434.0

From: "004512345678" <>004512345678@sipgate.co.uk>;tag=as0f6ef5c1

Call-ID: 0df02a86702544633c24f2141becd7ca@sipgate.co.uk

To: <>00441234567890@sipgate.co.uk>;tag=11780488-1571

CSeq: 102 ACK

Max-Forwards: 69

Content-Length: 0

X-hint: rr-enforced

013778: Feb 13 09:39:32.310 CET: //17026/B8E5F72AAF9C/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x0x328A7468

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           : 004512345678

Called Number            : 1664819

Source IP Address (Sig  ): 1.2.3.4

Destn SIP Req Addr:Port  : 217.10.79.23:5060

Destn SIP Resp Addr:Port : 217.10.79.23:5060

Destination Name         : 217.10.79.23

013779: Feb 13 09:39:32.310 CET: //17026/B8E5F72AAF9C/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : g711alaw

Negotiated Codec Bytes   : 160

Nego. Codec payload      : 8 (tx), 8 (rx)

Negotiated Dtmf-relay    : 6

Dtmf-relay Payload       : 101 (tx), 101 (rx)

Source IP Address (Media): 1.2.3.4

Source IP Port    (Media): 30794

Destn  IP Address (Media): 217.10.77.21

Destn  IP Port    (Media): 59720

Orig Destn IP Address:Port (Media): [ - ]:0

013780: Feb 13 09:39:32.310 CET: //17026/B8E5F72AAF9C/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 1

Disconnect Cause (SIP)   : 404

[Ayodeji Okanlawon · Accepted solution]

Hi,

From the logs..

The called number :00441234567890 is translated to extension 1004.

Now CUCM is saying that it cant find the extension. Do you have extension 1004 configured?

013776: Feb 13 09:39:32.282 CET: //17026/B8E5F72AAF9C/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.11.253:5060;branch=z9hG4bK185A14CC

From: "004512345678" <004512345678>;tag=11780480-1DAE

To: <1004>;tag=2215~51ae08a6-d9b1-40ed-8048-61635a68d171-22773051

Date: Wed, 13 Feb 2013 08:39:32 GMT

Call-ID: B8E72F7A-74EF11E2-AFA2E149-3AD11FA7@192.168.11.253

CSeq: 101 INVITE

Allow-Events: presence

Reason: Q.850;cause=1

Content-Length: 0

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

[danscout01]

Hi

Yes ext. 1004 is definately there, I just tried to call out, some of the output is here:

015091: Feb 13 12:27:33.060 CET: //17412/57B355800000/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x0x328B9AC8

State of The Call        : STATE_ACTIVE

TCP Sockets Used         : YES

Calling Number           : 1004

Called Number            : 080102030

Source IP Address (Sig  ): 192.168.11.253

Destn SIP Req Addr:Port  : 192.168.11.37:5060

Destn SIP Resp Addr:Port : 192.168.11.37:49490

Destination Name         : 192.168.11.37

015092: Feb 13 12:27:33.060 CET: //17412/57B355800000/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : g711alaw

Negotiated Codec Bytes   : 160

Nego. Codec payload      : 8 (tx), 8 (rx)

Negotiated Dtmf-relay    : 0

Dtmf-relay Payload       : 0 (tx), 0 (rx)

Source IP Address (Media): 192.168.11.253

Source IP Port    (Media): 30836

Destn  IP Address (Media): 192.168.1.29

Destn  IP Port    (Media): 46378

Orig Destn IP Address:Port (Media): [ - ]:0

So Maybe its in the CUCM that the prob is then.

Im still a bit new to CUCM, so got the outgoing working fine, but dialin seems to be a prob

[Ayodeji Okanlawon]

Check the CSS assigned to the SIP trunk under "inbound calls" (you need to scroll down to see this)..Make sure the CSS has access to the partion of the ip phone 1004

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

[danscout01]

I checked and checked and then realized that the CSS was wrong and that made the incomming call go though.

I still need to test the rest of the setup, but it seems like you are able to have multiple sip trunks on the cube

Thanks everyone for the support.

[chrysostomos1980]

Gald to hear that

Finally the communication with cucm stayed with h323 or is sip now

Please rate all useful posts

[danscout01]

I changed it to sip, so all h323 was removed also from the voice service voip.

all in all it seems like its working.

Pretty good since i have only worked with cucm since november and have been at cipt1 and 2

But those courses doesnt really cover the cube in detail.

but thanks to you guys i finally managed to get incomming calls to work. (i have once worked a bit with asterisk, but got stuck with incomming there was well).

So i have come a long way and now hopefully will get to implement som need features as well.

[Ayodeji Okanlawon]

Dan,

glad its looking good. Dont forget to rate useful/correct post. So others know  in the future how the issue was resolved

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"