CCME, Outsid call via internet

[medzeinmaaloum]

good day,

I have a Cisco Call Manager Express on my router it works locally (it can make calls locally), But I want that my CCME be able to do an outside call over Internet, how is that done?

best regards,

[Roger Kallberg]

You need a SIP trunk with a service provider for that.

[M02@rt37]

Hello @medzeinmaaloum 

A lot of things to do!

Subscribe to a VoIP service provider that allows you to make calls over the internet. Obtain the necessary account credentials, SIP server information, and any other required details ; as @Roger Kallberg suggest. Then, define a SIP trunk to connect your CCME to the VoIP service provider. This involves configuring the SIP server address, port, and authentication details. Also, create an outgoing dial-peer to route calls through the SIP trunk. This includes specifying the destination pattern and the dial-peer for voice.

[Roger Kallberg]

Not to mention that you'll need to think about security as you'd be connecting to internet. At a minimum having the built in security for voice services property setup and an ACL assigned to the interface that faces internet that only allows the traffic from/to the service provider for the SIP trunk is imperative. Otherwise it won't be long until you have someone unwanted pocking around in your system.

[medzeinmaaloum]

good day all,

Yes, a lot of things to understand and then to applicate, I study on it.

First of all, does the name SIP trunk have any relation to the SIP Firmware? in my phones I use SCCP, should I use SIP Firmware instead of SCCP Firmware?

Best regards,

[Roger Kallberg]

No. The phones will register with CME and it will act as the relay between the service provider and the device.

[Carlo Poggiarelli]

Hi,

Here is a basic config to connect your VG to a SIP Service Provider.

allow-connections sip to sip
no supplementary-service sip refer
sip
bind control source-interface GigabitEthernet0/0 <—— This is your internal interface that manages IP Phones through Telephony Service (SCCP) and Voice Register Global (SIP)
bind media source-interface GigabitEthernet0/0
rel1xx disable
min-se 3600
session refresh
registrar server
asymmetric payload full
early-offer forced
midcall-signaling passthru
sip-profiles inbound
no call service stop

voice class codec 100
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8

voice class uri Proxy sip <—— This is a Voice Class that matches the “host” part of the SIP Invite message to be matched as inbound call leg. In this case I know the exact IP Address of my SIP Provider. Youl’ll find it later on Inbound Dialpeer
host xx.xx.xx.xx

dial-peer voice 601 voip
description *****INBOUND FROM VOIP PROVIDER*******
translation-profile incoming IN-SIP <—— You need to translate the public number to a internal Destination and you can achieve this through a Voice Translation Pattern
session protocol sipv2
session target dns:sip.proxy.com:5060 <—— Here your SIP Provider address
session transport udp
voice-class sip bind control source-interface GigabitEthernet0/1 <—— In case of dual interface (highly indicated) here you specify the outgoing interface (to the Internet)
voice-class sip bind media source-interface GigabitEthernet0/1
incoming called-number 01234566 <—— Here your public number
incoming uri to Proxy <—— With this you match the calling Host in other words, your Sip Provider IP Address
dtmf-relay rtp-nte
voice-class codec 100
no vad

dial-peer voice 500 voip
description *****OUTBOUND NATIONAL/INTERNATIONAL*******
translation-profile outgoing OUT-SIP <—— When calling out, you need to translate your internal extension to your public Number(s)
session protocol sipv2
session target dns:sip.proxy.com:5060
session transport udp
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
destination e164-pattern-map 100
voice-class codec 100
dtmf-relay rtp-nte
no vad

sip-ua
credentials username 012345 password xxxxxx realm sip.proxy.com <—— With this entry, you register to your Voip Provider to receive calls on Public number
authentication username 012345 password 7 xxxxxxx realm sip.proxy.com <—— With this entry, you authenticate yourself to make outgoing calls to your Voip Provider
nat symmetric role active
nat symmetric check-media-src
no remote-party-id
retry invite 4
retry response 3
retry bye 3
retry cancel 2
timers register 100
timers keepalive active 300
registrar 1 dns:sip.proxy.com:5060 expires 180 refresh-ratio 50 <—— Here you specify the registrar server (your Voip provider) an the frequency of the “REGISTER” message you send to keep alive your number’s registration.

Please let us know if it can help.

Regards

Carlo

[Roger Kallberg]

I would suggest these changes to your suggestion.

voice service voip
 ip address trusted list
  ipv4 <SP SIP server 1>
  ipv4 <SP SIP server 2> ! add as many as you'd need
 rtp-port range <start port> <end port>
 address-hiding
 mode border-element 
!
dial-peer voice 601 voip
 no incoming called-number 01234566 !No need to have this as the match is made on the IP
 incoming uri via Proxy !Use information in VIA header to match instead of To

And also to use tenant configuration for the registration part instead of in sip-ua as otherwise there will be issues with SIP devices to register in CME.

[medzeinmaaloum]

@Roger Kallberg I will take this security section into consideration.

THANKS.

Best regards,

[medzeinmaaloum]

Good day all,

@Carlo Poggiarelli

Thank you very much for this code, it will help me a lot and it will save me a lot of time, I created an account with voip.ms as soon as I have the necessary parameters, I will try the code that you suggested.

Best regards,

[Carlo Poggiarelli]

Hi @Roger Kallberg 

I also use Incoming called number match is case of multiple DDI to manage. It happened in the past to have the same VIA match but different Called number .

Thanks

Cheers

Carlo

[Roger Kallberg]

Hi @Carlo Poggiarelli 
Maybe I'm missing your point but why would you need that with a match of VIA header on the inbound dial peer?

[Carlo Poggiarelli]

Hi @Roger Kallberg to match both VIA and Called Number because I have the case that different incoming called number need to be threated  in different ways. I know that it could be accomplished in many ways I just mentioned one of them ..

Cheers

Carlo