CME Trunk Configuration, sip-ua without credentials!

[Imma]

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,

[LA ASCENSION]

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

[shazeb051]

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

[Imma]

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

[shazeb051]

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

[Imma]

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

[TONY SMITH]

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

[Imma]

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

[Imma]

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

[Imma]

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

[TONY SMITH]

---

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.

[Imma]

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

[TONY SMITH]

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?

[Imma]

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