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Cisco CUCME and Asterisk(Elastix) Integration Question

Unanswered Question
Dec 1st, 2011
User Badges:

Hello almighty community!!


I´ve have the following scenario: CUCME (192.168.1.251) ======== SIP TRUNK ========= ELASTIX  (192.168.1.3)


I found the following config which worked:


voice service voip

address-hiding

allow-connections sip to sip

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

fax protocol cisco

sip



dial-peer voice 3000 voip

destination-pattern 3...

session protocol sipv2

session target ipv4:192.168.1.3

dtmf-relay rtp-nte

codec g711ulaw


!

dial-peer voice 6000 voip

description **Llamadas entrantes Troncal SIP**

translation-profile incoming AsteriskTrunk

destination-pattern 6000

session protocol sipv2

session target ipv4:192.168.1.3

codec g711ulaw


sip-ua

keepalive target ipv4:192.168.1.3 

credentials username 5000 password wxyz realm 192.168.1.3

authentication username 5000 password wxyz

retry invite 4

retry response 3

retry bye 2

retry cancel 2

retry register 5

timers register 250

registrar ipv4:192.168.1.3 expires 3600

sip-server ipv4:192.168.1.3:5060



But the problem is that only CUCME is registering against Asterisk server, but not viceversa. The person whos managing the Asterisk server provide me this info


host=192.168.1.251

username=6000 

secret=wxyz


Register String

6000:[email protected]


, so i´m suppose to make something in CUCME to allow it to register, but as you may have notice by now, i´m not an expert, this is the first time i´m trying to acomplish this. How can i do this?. The result  by the moment is that an IP phone connected to Asterisk can call a Cisco IP phone but when i try to call back from the Cisco IP phone to the Asterisk phone, it doesn´t work. The Asterisk PBX sees that a call is coming but it doesn´t establish it. Also, de Caller-ID

that Asterisk sees is 5000. I need that CID to be calling party number. For this last thing i was thinking to do this:


!

voice translation-rule 1

  rule 1 /^3\(...\)/ /3\1/

!

!

voice translation-profile AsteriskCID

translate called 1

!

!


dial-peer voice 3000 voip

translation-profile outgoing AsteriskCID


But im not sure if that should work. Any help would be appreciated. Thanks

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Andres Franco Sat, 12/03/2011 - 19:47
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Hello! everyone who could give any idea about this subject? Thanks in advance.

rivasque Mon, 12/05/2011 - 08:52
User Badges:

Hello Andres..


Do you need to get the CME to become as a register server for the asterisk server?


With this config:


sip-ua

keepalive target ipv4:192.168.1.3 

credentials username 5000 password wxyz realm 192.168.1.3

authentication username 5000 password wxyz

retry invite 4

retry response 3

retry bye 2

retry cancel 2

retry register 5

timers register 250

registrar ipv4:192.168.1.3 expires 3600

sip-server ipv4:192.168.1.3:5060


The CME will send sip registrations to 192.168.1.3


In order to convert the CME to a register server you need todo:


-ena

-config t

-voice service voip

-sip

-registrar server


With this the CME will accept SIP register messages coming in.


Hope it helps

Andres Franco Mon, 12/05/2011 - 10:05
User Badges:

Hola Richard!


Actually, i want to test both scenarios. The first one worked for me but then i try it using ip addressess on the data VLAN and stop working. Your answer is helpfull because clarifies me some doudbts but, i´m supposed to register the asterisk box on CME being like a register server using authentication, because the asterisk box is also sending me some credentials (

6000:[email protected]). How could i perform this?. Thanks again for your help, and if you could tell me about this last issue, it will be great.


AF.

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