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New Member

Cisco cube two sip trunks to same provider

Hi all,

 

have a big problem with the configuration of two sip trunks.

If I configure the second sip trunk to the provider, outgoing calls will not work anymore.

The second sip trunk has only 2 connections for fax devices.

 

Is there a solution to route outgoing calls via dial-peer to the first sip trunk and to the second sip trunk?

 

SNIP cube configuration:

sip-ua
 credentials username 0235XXXXX password 7 121F55180311070D2C realm 0235698656 >> sip trunk just for outgoing calls via phone.
 authentication username 023XXXXXpassword 7 06005F2E5D54021003

 credentials username 0235698XXXX password 7 094A1E06081F1C1B0D realm 02356XXXX << second sip trunk for fax device
 authentication username 02356XXXX password 7 110F490A0608000502 realm 023XXXXXX

 retry invite 3
 retry response 3
 retry bye 3
 retry cancel 3
 retry register 3
 retry options 10
 timers trying 1000
 timers connect 1000
 timers register 1000
 registrar ipv4:89.184.167.11 expires 3600
 sip-server ipv4:89.184.167.11
!

 

ial-peer voice 1100 voip
 description CUBE, Outgoing Enbloc to Provider
 translation-profile outgoing SIP-OUTGOING
 preference 2
 max-conn 30
 destination-pattern 0T
 session protocol sipv2
 session target sip-server
 session transport udp
 voice-class codec 100
 voice-class sip early-offer forced
 voice-class sip profiles 200
 voice-class sip bind control source-interface GigabitEthernet0/1
 voice-class sip bind media source-interface GigabitEthernet0/1
 dtmf-relay rtp-nte
 fax-relay sg3-to-g3
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 no vad
 supplementary-service pass-through

 

 

Thanks,

 

Stefan

Everyone's tags (4)
13 REPLIES

Hello Stefan, Outbound dial

Hello Stefan,

 

Outbound dial-peer matching based on the longest match rule first and then preference.

Preference 0 is the highest priority. Please modify the dial-peer configuration accordingly.

 

//Suresh

Please rate all the helpful posts

//Suresh Please rate all the useful posts.
New Member

Hi,thanks for your reply. I

Hi,

thanks for your reply.

 

I far with the preference feature.

This is not a solution. Because the second sip trunk is only configure with T.38.

So I need a other solution.

 

 

Thanks.,

Stefan

You may try dial-peer hunting

You may try dial-peer hunting feature.

http://rsccievoice.wordpress.com/2012/07/30/dial-peers/

 

also, could you please post your dial-peer configurations here?

 

// Suresh

Please rate all the helpful answers

//Suresh Please rate all the useful posts.
New Member

Hi,sorry for delay,dial-peer

Hi,

sorry for delay,

dial-peer voice 1000 voip
 description incoming from Provider
 translation-profile incoming SIP-INCOMING
 session protocol sipv2
 session target sip-server
 session transport udp
 incoming called-number .
 voice-class codec 100
 voice-class sip bind control source-interface GigabitEthernet0/1
 voice-class sip bind media source-interface GigabitEthernet0/1
 dtmf-relay rtp-nte sip-notify
 no vad
!
dial-peer voice 1010 voip
 description CUBE, Outgoing to ALU OXE - TEST
 preference 1
 destination-pattern 3020320
 session protocol sipv2
 session target ipv4:192.1.30.20
 voice-class codec 100
 voice-class sip profiles 100
 voice-class sip bind control source-interface GigabitEthernet0/2
 voice-class sip bind media source-interface GigabitEthernet0/2
 dtmf-relay rtp-nte sip-notify
 fax protocol t38 nse force version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
 no vad
!
dial-peer voice 1020 voip
 description CUBE, Outgoing to ALU OXE - TEST
 preference 1
 destination-pattern 2074009..
 session protocol sipv2
 session target ipv4:192.1.30.20
 voice-class codec 100
 voice-class sip profiles 100
 voice-class sip bind control source-interface GigabitEthernet0/2
 voice-class sip bind media source-interface GigabitEthernet0/2
 dtmf-relay rtp-nte sip-notify
 fax protocol t38 nse force version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
 no vad
!
dial-peer voice 1100 voip
 description CUBE, Outgoing Enbloc to Provider
 translation-profile outgoing SIP-OUTGOING
 preference 2
 max-conn 30
 destination-pattern 0T
 session protocol sipv2
 session target sip-server
 session transport udp
 voice-class codec 100
 voice-class sip early-offer forced
 voice-class sip profiles 200
 voice-class sip bind control source-interface GigabitEthernet0/1
 voice-class sip bind media source-interface GigabitEthernet0/1
 dtmf-relay rtp-nte
 fax-relay sg3-to-g3
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 no vad
 supplementary-service pass-through
!
dial-peer voice 1200 voip
 description CUBE, incoming from OXE
 shutdown
 session protocol sipv2
 session target sip-server
 incoming called-number 0.
 voice-class codec 100
 voice-class sip bind control source-interface GigabitEthernet0/2
 voice-class sip bind media source-interface GigabitEthernet0/2
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 1030 voip
 description CUBE, Outgoing to ALU OXE
 preference 1
 destination-pattern ^023[57]
 session protocol sipv2
 session target ipv4:192.1.30.20
 incoming called-number 0.
 voice-class codec 100
 voice-class sip profiles 100
 voice-class sip bind control source-interface GigabitEthernet0/2
 voice-class sip bind media source-interface GigabitEthernet0/2
 dtmf-relay rtp-nte sip-notify
 fax protocol t38 nse force version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
 no vad
!
dial-peer voice 1040 voip
 description CUBE, Outgoing to ALU OXE
 preference 1
 destination-pattern ^23[57]
 session protocol sipv2
 session target ipv4:192.1.30.20
 incoming called-number 0.
 voice-class codec 100
 voice-class sip profiles 100
 voice-class sip bind control source-interface GigabitEthernet0/2
 voice-class sip bind media source-interface GigabitEthernet0/2
 dtmf-relay rtp-nte sip-notify
 fax protocol t38 nse force version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
 no vad
!
dial-peer voice 1050 voip
 description CUBE, Outgoing to ALU OXE
 preference 2
 destination-pattern ^23[57]
 session protocol sipv2
 session target ipv4:10.7.2.3
 voice-class codec 100
 voice-class sip profiles 100
 voice-class sip bind control source-interface GigabitEthernet0/2
 voice-class sip bind media source-interface GigabitEthernet0/2
 dtmf-relay rtp-nte sip-notify
 fax protocol t38 nse force version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
 no vad
!

 

CallFlow:

 

Alactel Phone -> ALU PBX -> SIP SBC -> SIP Trunk Provider

Could you please collect

Could you please collect 'debug ccsip message & debug voice ccapi inout' for a test call? Please provide calling & called numbers too.

//Suresh Please rate all the useful posts.
New Member

Hi,now both sip trunks to the

Hi,

now both sip trunks to the sip provider are established and outgoing calls are not possible anymore.

cube#sh sip-ua register sta
Line                             peer       expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
023569XXXX                      -1         2334         yes >
023569XXXX                      -1         1497         yes

 

see attachment

calling number - 3123

called number - 0049332831133232

 

Hello Stefan,the call is

Hello Stefan,

the call is disconnected by the provider with error "SIP/2.0 503 Service unavailable - no free line"

*Apr  2 12:34:48.833: //191936/039AB403A326/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 503 Service unavailable - no free line

Via: SIP/2.0/UDP 213.206.215.27:5060;branch=z9hG4bK1B6F29B8;rport=58603

From: <sip:3123@89.184.167.11>;tag=6613B448-7BD

To: <sip:0049332831133232@89.184.167.11>;tag=c01b7e134ccfaf6dce24f17b9ae3d18b.e900

Call-ID: 39BEC53-B99A11E3-A32CA60F-9EF8485@213.206.215.27

CSeq: 102 INVITE

Server: OpenSIPS (1.6.2-notls (x86_64/linux))

Content-Length: 0

 

 

>> what is the calling number format the provider expects from you? you are sending only 4 digit extn to the provider.

 

>> Please talk to them on why they are disconnecting the call.


//Suresh

Please rate all the helpful posts.

//Suresh Please rate all the useful posts.
New Member

Hi, but this happen only if I

Hi,

 

but this happen only if I configure the second sip trunk.

 

Thanks,

stefan

VIP Super Bronze

 

 

This is the INVITE that is sent with this call. Is the authentication credentials correct for this call to this ITSP?

Proxy-Authorization: Digest username="0235698640",realm="sipserver
Sent: 
INVITE sip:0049332831133232@89.184.167.11:5060 SIP/2.0

Via: SIP/2.0/UDP 213.206.215.27:5060;branch=z9hG4bK1B6F29B8

From: <sip:3123@89.184.167.11>;tag=6613B448-7BD

To: <sip:0049332831133232@89.184.167.11>

Date: Wed, 02 Apr 2014 12:34:48 GMT

Call-ID: 39BEC53-B99A11E3-A32CA60F-9EF8485@213.206.215.27

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

Min-SE:  1800

Cisco-Guid: 0060470275-3113882083-2737219087-0166691973

User-Agent: Cisco-SIPGateway/IOS-15.2.4.M4

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

CSeq: 102 INVITE

Timestamp: 1396442088

Contact: <sip:3123@213.206.215.27:5060>

Expires: 180

Allow-Events: telephone-event

Proxy-Authorization: Digest username="0235698640",realm="sipserver",uri="sip:0049332831133232@89.184.167.11:5060",response="a86cc29b146444174b28286a2d8597b4",nonce="533c04c300008d8a5fadc81e040afef72653e6d04dbaa66e",algorithm=md5

 

From your config, this looks like the authentication credentials for the fax devices..

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Hi Ayodeji, thanks for your

Hi Ayodeji,

 

thanks for your reply.

 

This is what I mean, If I register the second trunk to the sip provider every external call will be routed with this credentials.

So I don't know if I can configure a routing decision just for outgoing calls.

Outgoing calls should be routed trought the first sip trunk.

Only incoming fax and outgoing fax should be route to the second sip trunk.

 

Thanks,

Stefan

VIP Super Bronze

What you need to do is

What you need to do is configure different registrars.

Example below. Change the values to yours and this should work for you.

 

Router> enable

Router# configure terminal

Router(config)# sip-ua

Router(config-sip-ua)# registrar 1 dns:example1.com expires 180

Router(config-sip-ua)# registrar 2 dns:example2.com expires 360

Router(config-sip-ua)# authentication username MyUsername password MyPassword1 realm
Realm.example1.com

Router(config-sip-ua)# authentication username MyUsername password MyPassword2 realm
AnotherRealm.example1.com

Router(config-sip-ua)# authentication username MyUsername password MyPassword3 realm
Realm.example2.com

Router(config-sip-ua)# authentication username MyUsername password MyPassword4 realm
AnotherRealm.example2.com

 

 

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Hi, can you also explain me,

Hi,

 

can you also explain me, how I  bind this on the outgoing dialpeer ?!

VIP Super Bronze

Stefan,Here is a sample

Stefan,

Here is a sample config of using two different providers..You need to configure the dial-peers facing each provider with the correct authentication credentials..

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

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