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New Member

First Post....please be kind

Hi all

Am am complete novice when it comes to all things Cisco and IP based, so please bear that in mind when reading this post.

What I am trying to achieve is to set-up a SOHO PBX that utilises a Raspberry Pi as the device that runs the PBX software (FreePBX Asterisk based system) so that in the office where we are we can utilise the 2 avalable POTS lines coming in across more than 2 phones (have 4 at present, will be more eventually).

I have set-up the Pi so that all 4 phones can connect and call each other, but am having great difficulty getting this connected to the outside world.

As the connection to the outside world I have a 2621XM router that has a NM-1V and VIC 2FXO-EU.

What I am aiming to do is have calls from port 1/0/0 (extension #3521) route to FreePBX and connect to a dial-group #601 and calls on port 1/0/1 do the same, but go to dial-group #602.  For outbound calls, I am wanting any phone to be able to dial out on any available line connected to the FXO card

At present, this is what I have on my router config (has been copied for second port, so only port 1/0/0 is shown)

voice-port 1/0/0

cptone GB

input gain 10

output attenuation 10

no comfort-noise

connection plar opx 601

dial-peer voice 601 voip

destination-pattern 601

session transport udp

session target ipv4:192.168.1.10:5060

session protocol sipv2

codec g711ulaw

dtmf-relay h245-signal

sip-ua

retry invite 3

retry response

retry bye 3

retry cancel 3

timers trying 1000

sip-server ipv4:192.168.x.x

voice rtp serd-recv

In FreePBX have the following in both the Inbound and Outbound config ares for the SIP Trunk

allow=ulaw

context=from-pstn

disallow=all

dtmfmode=h245-signal

host=192.168.1.1

insecure=very

ipaddr=192.168.1.1

type=peer

At present am unable to have anything go to the Pi or have anything passed from there out.

What I am noticing is that if i dial the extension that is connected to the FXO card on port 1/0/0 then teh 'In Use'LED on port 1/0/1 comes on.  I would have expected it to come on on teh port that was receiving the call.  This is what is confusing me (amongst several other things)

If anyone can please help that would be greatly appreciated.  As I have said and as you can see, I know next to nothing on this and am on a very steep learning curve indeed.

Many thanks in advance

  • IP Telephony
5 REPLIES
Green

First Post....please be kind

Paul,

On your dial peer 601 you are using a DTMF type that is for h323/245

Can you try changing to :-

!

dial-peer voice 601 voip

destination-pattern 601

session transport udp

session target ipv4:192.168.1.10:5060

session protocol sipv2

codec g711ulaw

dtmf-relay rtp-nte

!

Regards,
Alex.
Please rate useful posts.

Regards, Alex. Please rate useful posts.
New Member

First Post....please be kind

Alex

Many thanks for the reply.

Have tried that above, but am unable to enter rtp-nte.  Only optionas available are cisco-rtp, h245-alphanumneric and h245-signal.

Missed from OP, but version of IOs running on router is 12.2(17a)

Cheers

Green

First Post....please be kind

Paul,

So the cisco 26xx does not truly support dtmf for sip.

Can you investigate using an H323 trunk intead ?

Regards,
Alex.
Please rate useful posts.

Regards, Alex. Please rate useful posts.
New Member

First Post....please be kind

Alex

Looking into that now.  Found a post on teh FreePBX forum for installing a module that is supposed to allow H323 trunks.  Working on getting that installed (also teaching myself some Linux in this project too)

Will let you know how I get on

Regards

Paul

Bronze

First Post....please be kind

Hi Paul,

For a test inbound and outbound call provide below debugs from cisco router. I am assuming you are still using SIP between router and PBX

debug vpm signal

debug voip ccapi inout

debug ccsip message

debug ccsip error

Regards,

Mohit Singh

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