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Beginner

CME Trunk Configuration, sip-ua without credentials!

Hello all,

May someone advice me how to configure trunk without credentials and authentication section?

Topology:

CME---Mikrotik_router----trunk----ITSP(two vlan Internet+Voice)

 

For peering with IP what Configuration should I do under sip-ua in order for the CME to be authenticate with SIP server?

 

Thank you in advanced,

 

Everyone's tags (4)
1 ACCEPTED SOLUTION

Accepted Solutions
Beginner

Re: CME Trunk Configuration, sip-ua without credentials!

Hello all, 

I come back with the solution. Hope my notes can help others:

 

voice service voip
ip address trusted list
ipv4 188.x.x.1 <----- .1 and .2 are respectively SIP and RTP server of the ITSP--------
ipv4 188.x.x.2
ipv4 172.30.30.0 255.255.255.0 <--------Voice inside subnet

ipv4 10.x.x.6 <-------next hop ITSP router.
ipv4 188.x.x.3 <----.3 and .4 are respectively SIP and RTP server of the ITSP -------
ipv4 188.x.x.4
rtp-port range 10004 48198 <------------Must be same with your ITSP RTP ports----
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip handle-replaces
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server
asserted-id pai <------needed to display number to the called phone for outgoing calls
sip-profiles 1
!
voice class codec 99 <------necessary codec related to codec used by ITSP
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
codec preference 4 g729br8
!
voice translation-rule 2 <----rule for outgoing calls, strip 700 which is the prefix for outgoing calls_see below dial-peer 700T
rule 1 /^700/ //
!
voice translation-rule 10 <-----rule for incomming calls
rule 1 /\+6211/ /207/   <---translate DID to 207 extension
rule 2 /\+62../ /210/ <----all incomming calls reach 210 extension
!
!
voice translation-profile digitstr
translate called 2
!
voice translation-profile inbound
translate called 10
!
dial-peer voice 88 voip
translation-profile outgoing digitstr
destination-pattern 700T
session protocol sipv2
session target ipv4:188.x.x.1
voice-class codec 99
voice-class sip profiles 1
voice-class sip bind control source-interface GigabitEthernet0/0/0.1004
voice-class sip bind media source-interface GigabitEthernet0/0/0.1004
clid network-number +3556xxxx6211
no vad
!
dial-peer voice 10 voip
translation-profile incoming inbound
session protocol sipv2
incoming called-number +.T
voice-class codec 99
voice-class sip profiles 1
voice-class sip bind control source-interface GigabitEthernet0/0/0.1004 <----important binding with interface facing the ITSP because ITSP want the request come from the IP 10.x.x.5

 

voice-class sip bind media source-interface GigabitEthernet0/0/0.1004
!
dial-peer voice 11 voip      <-----Backup dial peer
translation-profile outgoing digitstr
preference 2  <----Because it is for backup
destination-pattern 700T
session protocol sipv2
session target ipv4:188.x.x.3
voice-class codec 99
no voice-class sip early-offer forced
voice-class sip profiles 1
voice-class sip bind control source-interface GigabitEthernet0/0/0.1004
voice-class sip bind media source-interface GigabitEthernet0/0/0.1004
clid network-number +3556xxxx6211
no vad
!
!
presence
presence call-list
!
sip-ua
registrar ipv4:188.x.x.1 expires 3600
sip-server ipv4:188.x.x.1
connection-reuse <----What bring the Trunk Up. Needed in order for CME to send the request to ITSP's SIP server on the same port it use 5060.

handle-replaces
!

Everyone's tags (4)
22 REPLIES 22
Beginner

Re: CME Trunk Configuration, sip-ua without credentials!

Alguien ya te ayudo? tengo la misma duda ya pudiste?
Beginner

Re: CME Trunk Configuration, sip-ua without credentials!

Hi Kindly follow Below Config for Sip-ua

Router# configure terminal
if you get any difficulty please revert back,
Router(config)# sip-ua

Router(config-sip-ua)# registrar ipv4:x.x.x.x
expires 3600 secondary
Router(config-sip-ua)# no redirectio
Router(config-sip-ua)# redirection



sip-ua
credentials username 100001 password 1357924680 realm sip-ua.com
authentication username 100001 password 1357924680
registrar dns:proxy.sip-ua.com expires 60
sip-server dns:proxy.sip-ua.com
retry invite 2
timers trying 150

Beginner

Re: CME Trunk Configuration, sip-ua without credentials!

hi Shazeb, 

thank you for your reply. 

 

In my case service provider does not provide username and password. Authentication is performed via IP peering.

 

I had a recommendation from cisco support that in this case nothing should be added under sip-ua:

no sip-ua

 

Regards,

Denisa

Beginner

Re: CME Trunk Configuration, sip-ua without credentials!

Hi Dena,
In case The Service Provider has not Given you any Authentication it means its Direct Traffic.

Is you connection SIP-SIP.??

still you need to Configure Sip-ua.

sip-ua
retry invite 5
retry bye 2
retry cancel 2
retry notify 5
retry register 10
timers notify 300
registrar ipv4:(Gateway IP ) expires 3600
sip-server ipv4:(Sip Server IP
connection-reuse
host-registrar
Beginner

Re: CME Trunk Configuration, sip-ua without credentials!

Hi again,

It's not functions. I add the configuration you proposed. Changed the topology:

 

IP phone172.30.30.207-----SW----172.30.30.1-CME -10.x.x.5---Trunk vlanVOICE--10.x.x.6- SP_Router

 

!

!
dial-peer voice 10 voip
destination-pattern 99T
session protocol sipv2
session target ipv4:188.x.x.1
voice-class codec 99
no voice-class sip early-offer forced
clid network-number 35567200xxxx
!
dial-peer voice 100 voip
description Inbound dial-peer
session protocol sipv2
session target sip-server
incoming called-number .T
!
dial-peer voice 11 voip
preference 2
destination-pattern 99T
session protocol sipv2
session target ipv4:188.x.x.3
no voice-class sip early-offer forced
codec g711alaw
clid network-number 35567200xxxx
!
!
presence
presence call-list
!
sip-ua
retry invite 5
retry bye 2
retry cancel 2
retry notify 5
retry register 10
timers notify 300
registrar ipv4:10.x.x.6 expires 3600
sip-server ipv4:188.x.x.1
presence enable
!

 

I have attached the debug logs also:

Debug ccsip messages
Debug ccsip error
Debug voice ccapi inout

 

Thank you,

Any ideas, why this happen?

 

Regards,

Denisa

Participant

Re: CME Trunk Configuration, sip-ua without credentials!

If your ITSP does not provide credentials, your sip-ua probably does not register at all. Here's an excerpt from one of our gateways for a non-authenticated ITSP (AT&T in that case). I think if you configure registration, the dial peers pointing to the sip-server target will be down if registration fails.

sip-ua
 no remote-party-id
 retry invite 2
 timers trying 250
 connection-reuse

 

Beginner

Re: CME Trunk Configuration, sip-ua without credentials!

Hi,

thank you for your reply.

Actually nothing changed with the configuration proposed by you. Can anyone explain the below messages:

 

041183: May 31 10:34:02.557: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received: ******Received correct from SP*****

OPTIONS sip:10.70.x.5:5060 SIP/2.0 *********this is CME router gateway*********

Via: SIP/2.0/UDP 188.172.x.1:5060;branch=z9hG4bKrmaae6rq71e81gravm7sr55m5;Role=3;Hpt=8e58_16;pth=0;X-HwDim=4

Call-ID: q41r7a55rrsq8e4q4l4qr1el84a68s6e@1.x.172.188

From: <sip:SBC@188.172.x.1>;tag=794s44ml **********this is the SIP server of Service Provider**********

To: <sip:10.70.x.5>

 

041184: May 31 10:34:02.558: //357737/ACC0F87689D3/SIP/Msg/ccsipDisplayMsg:

Sent: **********Sent from CME to SP********

SIP/2.0 200 OK

Via: SIP/2.0/UDP 188.172.x.1:5060;branch=z9hG4bKrmaae6rq71e81gravm7sr55m5;Role=3;Hpt=8e58_16;pth=0;X-HwDim=4

From: <sip:SBC@188.172.x.1>;tag=794s44ml *****it should be from 10.70.x.5 (CME) to 188.172.x.1 (SP sip server)******

To: <sip:10.70.x.5>;tag=231EA1DF-14A7

Date: Fri, 31 May 2019 08:34:02 GMT

Call-ID: q41r7a55rrsq8e4q4l4qr1el84a68s6e@1.x.172.188

Server: Cisco-SIPGateway/IOS-16.6.5

CSeq: 1 OPTIONS

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

Allow-Events: telephone-event

Accept: application/sdp

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

Content-Type: application/sdp

Content-Length: 364

 

Thank you in advanced,

BR,

Denisa

Highlighted
Beginner

Re: CME Trunk Configuration, sip-ua without credentials!

Hi,

thank you for your reply.

Actually nothing changed with the configuration proposed by you. Can anyone explain the below messages:

 

041183: May 31 10:34:02.557: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received: ******Received correct from SP*****

OPTIONS sip:10.70.x.5:5060 SIP/2.0 *********this is CME router gateway*********

Via: SIP/2.0/UDP 188.172.x.1:5060;branch=z9hG4bKrmaae6rq71e81gravm7sr55m5;Role=3;Hpt=8e58_16;pth=0;X-HwDim=4

Call-ID: q41r7a55rrsq8e4q4l4qr1el84a68s6e@1.x.172.188

From: <sip:SBC@188.172.x.1>;tag=794s44ml **********this is the SIP server of Service Provider**********

To: <sip:10.70.x.5>

 

041184: May 31 10:34:02.558: //357737/ACC0F87689D3/SIP/Msg/ccsipDisplayMsg:

Sent: **********Sent from CME to SP********

SIP/2.0 200 OK

Via: SIP/2.0/UDP 188.172.x.1:5060;branch=z9hG4bKrmaae6rq71e81gravm7sr55m5;Role=3;Hpt=8e58_16;pth=0;X-HwDim=4

From: <sip:SBC@188.172.x.1>;tag=794s44ml *****it should be from 10.70.x.5 (CME) to 188.172.x.1 (SP sip server)******

To: <sip:10.70.x.5>;tag=231EA1DF-14A7

Date: Fri, 31 May 2019 08:34:02 GMT

Call-ID: q41r7a55rrsq8e4q4l4qr1el84a68s6e@1.x.172.188

Server: Cisco-SIPGateway/IOS-16.6.5

CSeq: 1 OPTIONS

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

Allow-Events: telephone-event

Accept: application/sdp

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

Content-Type: application/sdp

Content-Length: 364

 

Thank you in advanced,

BR,

Denisa

Everyone's tags (4)
Beginner

Re: CME Trunk Configuration, sip-ua without credentials!

 
Beginner

Re: CME Trunk Configuration, sip-ua without credentials!

Hi,

Thank you for your reply. 

When I try to make a call nothing happen. I type the number and nothing, I even do not hear any dial-tone at all. When I try inbound calls I receive "network failure". I have configured two dial-peer for outbound calls (dial-peer voice 10 and dial-peer voice 11) the second one is for backup. Because the ITSP has provided two SIP servers IP addresses 188.172.x.1 and 188.172.x.3).

dial-peer voice 100 is for incoming calls. I thought that "session target sip-server" is same as write "session target ipv4:188.x.x.1" as long as I have configured  "sip-server ipv4:188.x.x.1" under sip-ua. Isn't this true?

Could you please advise me how should I configure the dial -peer?

 

Thank you,

Regards,

Denisa

Everyone's tags (4)
Participant

Re: CME Trunk Configuration, sip-ua without credentials!


When I try to make a call nothing happen. I type the number and nothing, I even do not hear any dial-tone at all. When I try inbound calls I receive "network failure". I have configured two dial-peer for outbound calls (dial-peer voice 10 and dial-peer voice 11) the second one is for backup. Because the ITSP has provided two SIP servers IP addresses 188.172.x.1 and 188.172.x.3).

dial-peer voice 100 is for incoming calls. I thought that "session target sip-server" is same as write "session target ipv4:188.x.x.1" as long as I have configured  "sip-server ipv4:188.x.x.1" under sip-ua. Isn't this true?

Could you please advise me how should I configure the dial -peer?

 

Thank you,

Regards,

Denisa


It sounds like the outbound call attempt isn't matching a dial peer.   Currently your outbound dial peers expect the dialled number to start 99 and to pass the whole dialled number to the provider.   Difficult to be more specific with out knowing your country's numbering plan, and the format the provider expects.

 

You can use the command "show dialplan number xxx time" to show which dial peers, if any, are matched by a particular dialled number.  Obviously xxx should be your dialled number.

 

Edit .. you say you don't even get dial tone.  Can your phones call each other, I mean is the CME all working correctly other than this SIP trunk?   If you mean secondary dial tone, to be played after you've dialled the PSTN access code then that needs to be specifically configured.

Beginner

Re: CME Trunk Configuration, sip-ua without credentials!

Hello Tony,

I have specified the clid network-number 35567200xxxx under dial-peer just for testing purposes. And destination-pattern 99T for outbound calls. So I type 99067200xxxx when I make the test. 

show dialplan number 35567200xxxx - does not display any output. 

 

Regards,

Denisa

Participant

Re: CME Trunk Configuration, sip-ua without credentials!

The dial peer command "clid" sets the calling number presentation, nothing to do with the dialled number or the destination.  I'm not even sure wildcards are supported.

 

Your dial peer destination pattern is 99T, meaning any number starting with 99 followed by any further digits until inter-digit timeout.  You don't have anything matching any other dialled number, so I'd expect that "show dialplan" command would return "No match, result=-1".   

 

Are internal calls from one handset to another working, phones properly registered?

Beginner

Re: CME Trunk Configuration, sip-ua without credentials!

Hello again Tony,

internal calls work properly, yes the ip-phones are all registered. 

 

yes the show dialplan return the below result:

No match, result=1(MORE_DIGITS_NEEDED)

 

thanks,

Denisa

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