How to connect 3CX PBX to Cisco router gateway by IP-to-IP

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nhuonghuynh

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Dear friends,

I have Cisco gateway device It aready connected to PSTN by E1 interface, so I don't know How to configure it connect to 3CX PBX. Thank you.

I hope to reply form you as soon as posible.

Best regards.

Nhuong Huynh
 
What type of Cisco Device is it?

also provide a link to a manual that might help us.

Provide some more details like IP segments, are both devices on the same network segment?
 
Thank you for your reply.

My voip gateway is Cisco router 7200 VXR, please go to the link http://www.cisco.com/en/US/products/hw/routers/ps341/products_installation_guide_book09186a008007daa6.html to see detail. By the way, both devices are on the same network segment.

thank you

Kind regards,

Nhuong Huynh
 
Is this a SIP gateway though? A quick look at the brochure reveals nothing about it being a SIP VOIP gateway...

I have not heard about CISCO routers having a SIP interface either so most likely you can not use this device. Unless i have overlooked something

Check with the supplier whether it really can be used as a SIP VOIP Gateway and ask him to explain how.
 
Hi,

I'm sure that it's a voip gateway, and it already connected to PSTN by E1 trunking and convert PSTN to Voip signal (it's the same SPA 3000 functions). By the way, please let's me know what the codec it support for? Thanks

Kind Regards,

Nhuong Huynh
 
nhuonghuynh said:
Hi,

I'm sure that it's a voip gateway, and it already connected to PSTN by E1 trunking and convert PSTN to Voip signal (it's the same SPA 3000 functions). By the way, please let's me know what the codec it support for? Thanks

Kind Regards,

Nhuong Huynh

Hey there

Admin meant that there are different protocols involved in VoIP communications, and this device doesn't mention anywhere anything regarding SIP which is the protocol we are using. Regarding codecs, we support G711u, G711a, GSM and as of lately Speex.
 
Yes Cisco certain Cisco Routers are VOIP Gateways also. I deal with Cisco 2816 w/ 4 port FXO's all the time. I however have not had a chance to get my hands on one and try it with 3CX. Cisco VOIP gateways can perform SIP, but if I remember correctly it has to be a particular IOS. Not positive on that one, I would have to do a little research on it.
 
More on Cisco VOIP Gateways

I am several years removed from my CCNA, and to be truthful I am a little overwhelmed by the VOIP options available - and probably have reached my level of incompetence..lol

I have been looking at options for my business and am looking hard at a solution from a company that will use a Cisco 2431 as a VOIP gateway; however as I understand it THEY won't be transporting via SIP - however the router does support it.

So do we just need the SIP protocol between the 2431 and 3CX?

One of the reasons I like the use of a 2431 is the QOS available to keep voice calls at a quality level... with a prominent business, I just feel safer. My thinking, is that I can have the best of both worlds: use the 3CX as an internal PBX, but use the Cisco 2431 as the gateway (which I see described/modeled in your website).

Here is a link to the product specs.
http://www.cisco.com/en/US/products/hw/gatecont/ps887/products_data_sheet09186a008019aded.html

P.S. I am impressed with the work you guys have done both on the product and the site.
 
Cisco VOIP Gateways

To show my ignorance, they are using SIP (have to for the voice obviously). I have confirmed that I can use the 2431 in conjunction with any SIP based server. So I have a point to point at the router, which hands off to the 3CX Server.
I will attempt the negotiation between the 2431 and the 3CX and post back.
 
I have a Cisco 2621XM router and have successfully configured it so that it acts as my VoIP gateway.

Hardware configuration is as follows:

NM-HD-2V with 1 2-port VIC2-2FXO card
IOS version: c2600-ipvoicek9-mz.124-9.T5.bin

Here's the necessary configuration to get things working on the Cisco router with 3CX PBX.

The code below tells the router what kind of VoIP service to use, which is set to sip

voice service voip
sip

Next, define the codecs you wish to you in order of preference.

voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8 bytes 40
codec preference 3 g723r63 bytes 96
codec preference 4 g726r16 bytes 80

Next configure your voice ports. This example is only for the first voice port. The last line is the most important as it tell the port what extension to dial on inbound call. The opx parameter tells the router that the extension is an off premise extension.

voice-port 1/0/0
no battery-reversal
timing hookflash-out 50
connection plar opx 999

Next program your dial-peers. Pick any arbitrary number for your dial-peers but they should make some sense to you. I've chosen 20000 for my voip dial-peer, which is used for inbound calls. If you have more than one voip dial-peer, set the preference order.

dial-peer voice 20000 voip
preference 1
destination-pattern 99.
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.253.5 --> address of 3CX server
dtmf-relay rtp-nte
no vad

Next program your POTS dial-peer.

dial-peer voice 10000 pots
preference 1 --> use if you have multiple POTS dial-peers
destination-pattern .T
port 1/0/0 --> matches your dial-peer with physcial port

Next program the SIP user agent parameters. The "authentication" parameter is used to authenticate the Cisco router against the gateway you programmed in the 3CX server.

sip-ua
authentication username 10000 password 091D1E594955
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers expires 300000
registrar ipv4:192.168.253.5 expires 300
sip-server ipv4:192.168.253.5:5060

The above should work for any Cisco router that supports VoIP.
 
Hye frd thanks for your information. I want to now my gateway is already working with H323 and ISDN E1 for voice is already there if i enter these commnad there. they will affect on my previous configuration or nor ?
 
were you able to get this working? 3cx as pbx and cisco as voip gateway? you can configure cisco with both h323 and sip. just add new dial-peers accordingly.
 
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