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,

[TONY SMITH]

What does it show with the "timeout" option on the end, that's telling the command to act as if the interdigit timeout has expired.  For example "sh dialplan number 991234 timeout"

Can you also try a "debug voip dialpeer inout" and see what that shows when you try an outbound call.   

Also the output from "sh dial-peer voice sum"

[Imma]

Hi Tony,

i have attached the information you requested. Also some other trace.

Thanks,

Denisa

[TONY SMITH]

Just had another thought.  You're relying on inter-digit timeout, but what do you have that set to and are you sure you're waiting for that time?   For example you've dialled 990672001258, but the CME will be sitting waiting to see if any further digits are going to be dialled.  Default might be quite long, for example it's 15 seconds for CUCM.

[Imma]

Yes you are right, but in that case would experience just 15 sec default timeout delay and 1 sec for call routing. I have experienced that delay with E1 dial peer (and resolved isdn sending-complete under serial interface).

I wait more than 16 sec and nothing happen.

Regards,

Denisa

[TONY SMITH]

Are you using SIP phones?  If so then enable just these two debugs "debug ccsip mess" and "debug voip dial inout".

Try both an outbound and an inbound call while capturing all the debug output.   I'm not quite sure whether this is a CME or a SIP issue, I suspect the former as it doesn't seem to be even trying an outbound call.

[TONY SMITH]

Those Options pings and replies all look correct.  What happens when you try an outbound call, it's not quite clear to me which dial peer you expect to be hitting.

[Imma · Accepted solution]

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
!

[vishnu5555]

Thank you for the details.