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New Member

Route FXO Call to Asterisk via SIP Trunk

I am having an issue with trying to route all call that come in on one of the FXO lines over a SIP Trunk to my Asterisk box. Here is what I have working and not working:

WORKING:

Exchange 2010 UM (Voice Mail)

CISCO 2621XM with VIC2-FXO ports, connected as Gateway in CUCM

FXO Ports configured on CUCM 6.1 - these are working ona VIC2-FXO card

I have IP Phones configured and connected to CUCM.

I Have a SIP Trunk configured for Asterisk connections.

I have a ROUTE PLAN that says any 5XXXX Number route over the SIP Trunk

I can call the Asterisk Box just fine when I use the IP phones that are connected to CUCM such as EXT 1000  on CUCM can call any extension on the Asterisk Box that are 5XXX.

I can call from the outside and reach an extension on the Asterisk Box as long as the phone is login.

NOT WORKING

If I setup the FXO port for a DN say 5000 and I dial in and the phone is not login I get a BUSY as I have all of the extensions setup to route to Exchange 2010 voice mail.

I need to figure out how to hard code a CALLER ID number from the CISCO FXO Port as when it tries to dial the Exchange Voice Mail, this is where I am getting a BUSY. These FXO lines from the PSTN do not have CALLER-ID and I beleive this is what is causing this issue with BUSY.

So what I need is advice on how to setup the FXO PORT with a CALLER-ID Number or some type of FIXED Number that is sent to Exchange.

Thanks,

Cliff

Any help would be appreciated.

3 ACCEPTED SOLUTIONS

Accepted Solutions
Hall of Fame Super Gold

Re: Route FXO Call to Asterisk via SIP Trunk

voice-port ...

station number ...

Hall of Fame Super Gold

Re: Route FXO Call to Asterisk via SIP Trunk

Are you using MGCP? If so, it doesn't support caller-id, and you must use H.323 instead.

Cisco Employee

Re: Route FXO Call to Asterisk via SIP Trunk

first remove mgcp from router: "no mgcp"

then you need 2 things:

1. Configure PLAR for fxo so when incoming call hits port it goes to plar configured, then...

2. Configure dial-peer for the previous plar

i.e.

voice-port x/y


connection plar 5000

dial-peer voice 1 voip

destination-pattern 5000

session target ipv4:x.x.x.x   ----->call manager's ip address

codec ?(depending on region)

note that this affects all your dialplan so if outgoing calls is to be passsed through this it will also need to be changed. Here are some exapmles:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/pbx/interop/notes/295164fx.pdf

does it help ?

7 REPLIES
Hall of Fame Super Gold

Re: Route FXO Call to Asterisk via SIP Trunk

voice-port ...

station number ...

New Member

Re: Route FXO Call to Asterisk via SIP Trunk

I tried your suggestion and still no number.

When I dial from the outside and connect to a phone IP Phone that is registered to CUCM it displays unknown@mydomainname.com

It doesn't seem to sending the station ID.

I have both ports configured on the VIC2-FXO card: station-number 2000 and station-number 2020

I also tried:

caller-id enable

sip-ua & remote-party-id

Is there any thing else I could try?

Thanks,

Cliff

Here is my partial config:

voice-port 1/0/0

signal groundStart

timing hookflash-out 50

station-id name Outside

station-id number 2000

!

voice-port 1/0/1

signal groundStart

timing hookflash-out 50

station-id name Outside

station-id number 2020

Hall of Fame Super Gold

Re: Route FXO Call to Asterisk via SIP Trunk

Are you using MGCP? If so, it doesn't support caller-id, and you must use H.323 instead.

New Member

Re: Route FXO Call to Asterisk via SIP Trunk

Yes I am using MGCP and within CUCM I setup the gateway.

How can I change to H.323?

I am very new to the CISCO Platform.

Thanks,

Cliff

Cisco Employee

Re: Route FXO Call to Asterisk via SIP Trunk

first remove mgcp from router: "no mgcp"

then you need 2 things:

1. Configure PLAR for fxo so when incoming call hits port it goes to plar configured, then...

2. Configure dial-peer for the previous plar

i.e.

voice-port x/y


connection plar 5000

dial-peer voice 1 voip

destination-pattern 5000

session target ipv4:x.x.x.x   ----->call manager's ip address

codec ?(depending on region)

note that this affects all your dialplan so if outgoing calls is to be passsed through this it will also need to be changed. Here are some exapmles:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/pbx/interop/notes/295164fx.pdf

does it help ?

New Member

Re: Route FXO Call to Asterisk via SIP Trunk

I was able to solve my issue and I did not have to remove the MGCP.  Under the SIP Trunk that is used to connect there is an option for OUTBOUND, under OUTBOUND I set the following:

Caller ID DN: 9999

Caller Name: CISCO

Now everything that leaves the CISCO 2621XM Gateway over the SIP trunk passes this number.

Now everything above is working, Exchange VM, Exchange UC, CISCO Gateway, CISCO CUCM with MCGP enabled.

I didn't have to remove anything and now Caller ID is spoofed since these FXO lines were not setup with Caller ID. It also passes this info to all phones either on the Asterisk side, Excahnge Side or the CISCO side which is OK for now until Caller-ID is enabled on the PSTN Lines.

I am going to assume that since it is passing this info as Caller ID and Caller Name, it will pass true Caller ID info over the FXO lines. :-)

Thanks for everyone's help. In the end just a little digging into the system and all is great!!!!!!!!

Hall of Fame Super Gold

Re: Route FXO Call to Asterisk via SIP Trunk

I am going to assume that since it is passing this info as Caller ID and  Caller Name, it will pass true Caller ID info over the FXO lines. :-)

No, it will not, as mentioned above.

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