cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
1475
Views
25
Helpful
14
Replies

VoIP connection to PABX

Nukesquad
Level 1
Level 1

Essentially, I'm trying to extend a analog line from a Nortel PABX across a data network to a slightly out of reach phone.

I have 2 routers (2801) equipped with FXO and FXS cards each. I want to extend a analog line (RJ11) from the PABX to the router with the FXO card and from that router to another router with a FXS card, and from there, to a POTS phone. The PABX would provide an extension to the phone, and would route calls to that extension to the connected port on the router. On the other hand, the router with the FXS card would capture whatever number the phone dials and just pass it straight alot to the PABX.

Is this possible? I've been reading reference material all week trying to figure it out.

1 Accepted Solution

Accepted Solutions

Can you post the rest of your config? What POTS destination pattern are you using on the other router. Is that the same as your PLAR target?

One suggestion would be to make your VOIP destination pattern non-overlapping with the POTS. As an example if you used 3888 on the other router change your pattern to 38..

Dave

View solution in original post

14 Replies 14

dgahm
Level 8
Level 8

Laurence,

This is a very common setup. This document emphasizes passing hookflash, but also includes the basic config required for an OPX.

http://www.cisco.com/en/US/tech/tk1077/technologies_configuration_example09186a008009431b.shtml

Please rate helpful posts.

Dave

Looks like exactly what I wanted. Man... how did I miss this. Thanks!

EDIT: Just one question. On the configuration, there's a line that goes "incoming called-number ."

Is this a description? Or an actual command?

Ah.. The previous question is kinda stupid but I have another one.

Question: Is hookflash really nessesary if I'm not using anything like call forwarding? The basic configuration for dialpeers from the example is pretty similar to what I have before, which isn't working.

I just posted this when I saw your post:

I have implemented the following:

http://www.cisco.com/en/US/tech/tk1077/technologies_configuration_example09186a008009431b.shtml

inter-branch calls work 100%. Problem is that when remote site phones out (PSTN breakout at HQ) the audio is extremely soft. Any ideas?

If you don't mind, could you share the specifics of the configurations?

What I have at the moment is 2 routers back to back, connected via fastethernet instead of serial as in the picture.

No routing protocol is setup at the moment, but it will once I solve the voice issue.

I've pretty much followed it to the letter when it comes to the voice port config and dial-peer. Only that my extension is set to 2888, as that's the line number for the PABX.

dial-peer voice 1 pots

destination-pattern 2888

port 1/0

dial-peer voice 2 voip

incoming called-number .

destination-pattern ....

session target ipv4:10.10.3.2

dtmf-relay h245-signal

And my destination pattern for voip routing is set to all wildcards as any number coming in from that line would to be routed to the remote router.

On the other side, my phone is connected to port 1/0.

dial-peer voice 1 pots

destination-pattern 2888

port 1/0

dial-peer voice 2 voip

incoming called-number .

destination-pattern ....

session target ipv4:10.10.3.3

dtmf-relay h245-signal

Routing all incoming calls to the next router and all outgoing call matching 2888 to port 1/0.

Have set the hookflash as well, and I got nothing. A call from the PABX would ring, but the phone remains silent. The phone, does not even have a dialtone, just a busy signal.

Hi,

for the HQ router with 4 FXO's i have:

!

voice-port 0/3/0

input gain 2

output attenuation -4

timing hookflash-out 500

connection plar opx 200

impedance complex1

!

voice-port 0/3/1

input gain 2

output attenuation -4

timing hookflash-out 500

connection plar opx 201

impedance complex1

!

voice-port 0/3/2

input gain 2

output attenuation -4

timing hookflash-out 500

connection plar opx 202

impedance complex1

!

voice-port 0/3/3

input gain 2

output attenuation -4

cptone ZA

timing hookflash-out 500

connection plar opx 203

impedance complex2

!

dial-peer voice 5572 pots

destination-pattern 5572

port 0/3/0

!

dial-peer voice 5852 pots

destination-pattern 5852

port 0/3/1

!

dial-peer voice 5815 pots

destination-pattern 5815

port 0/3/2

!

dial-peer voice 5853 pots

destination-pattern 5853

port 0/3/3

!

dial-peer voice 200 voip

destination-pattern 20.

session target ipv4:10.61.156.1

incoming called-number .

dtmf-relay h245-signal

no vad

!

and for the remote site with 4 FXS i have:

!

voice-port 0/2/0

input gain 2

output attenuation -4

cptone ZA

connection plar 5572

!

voice-port 0/2/1

input gain 2

output attenuation -4

cptone ZA

connection plar 5852

!

voice-port 0/3/0

input gain 2

output attenuation -4

cptone ZA

connection plar 5815

!

voice-port 0/3/1

input gain 2

output attenuation -4

cptone ZA

connection plar 5853

!

!

dial-peer voice 200 pots

destination-pattern 200

port 0/2/0

!

dial-peer voice 201 pots

destination-pattern 201

port 0/2/1

!

dial-peer voice 202 pots

destination-pattern 202

port 0/3/0

!

dial-peer voice 203 pots

destination-pattern 203

port 0/3/1

!

dial-peer voice 100 voip

max-conn 4

destination-pattern 5...

session target ipv4:10.59.1.70

incoming called-number .

dtmf-relay h245-signal

no vad

!

Hope this helps

Can you post the rest of your config? What POTS destination pattern are you using on the other router. Is that the same as your PLAR target?

One suggestion would be to make your VOIP destination pattern non-overlapping with the POTS. As an example if you used 3888 on the other router change your pattern to 38..

Dave

Silly me.

My PLAR connection did not match. However, fixing that, and setting PLAR connection on the remote router, whenever I pick up the phone on the remote site, it immediately starts dialing/ringing. Does not wait for dialing.

Calling to the remote site however, works fine now.

Hi,

usally the plar command is used only in the router connected to PBX, so that when a call comes in, it rings the remote phone.

In the other router, you will need a dial-peer voip with a destination-pattern that covers all the destinations the phone is allowed to dial.

I suspected as much, but the diagram and the config example from the site was so promising.

My problem is that there's going to be multiple phone connected on the remote site, each with a corresponding line to the PBX. I'm having problem creating a POTS dial-peer on the main router that will put a call coming in from a specific port to its corresponding port.

For example, a dialed called on voice port 0/0 on the remote site needs to be connected to the voice port 0/0 on the router connected to the PBX, and 0/1 to 0/1 etc. I'm using wildcards on that dial-peer, which means that while the setup works with 1 phone, I don't think it will work with 2, as the POTS dial-peer would overlap (pretty much the same actually).

Ok, I see. Multiple lines requires 1 to 1 mapping, else there could be confusion about the identy of the calling line.

There is a way to handle the above even without connection plar, but I won't mention that here.

The key is to use different numbers and destination patterns:

--- ROUTER PBX ---

voice-port 0/0

connection plar opx 100

voice-port 0/1

connection plar opx 101

dial-peer voice 100 pots

destination-pattern 200

port 0/0

destination-pattern 201

port 0/1

dial-peer voice 101 voip

session-target ipv4:10.10.10.2

destination-pattern 1..

--- ROUTER REMOTE ---

voice-port 0/0

connection plar 200

voice-port 0/1

connection plar 201

dial-peer voice 100 pots

destination-pattern 100

port 0/0

destination-pattern 101

port 0/1

dial-peer voice 101 voip

session-target ipv4:10.10.10.1

destination-pattern 2..

You will see that I used opx keyword at PBX router only, so that unless the remote phone actually takes the call, the PBX will see no answer (for billing, CDR, whatever).

There are variations to PLAR like tie-line, etc but to not make thing too complicated are we won't discussed here.

Hmm...

I actually did this.

The problem I had is that while the call from PBX to phone works well. When I pick up a phone from the remote site to call an extension, all I get is an immediate ringing tone (unsure of what number its dialing).

I though that PLAR would allow me to get a dial tone from the PBX, and then transfer whatever number I dial over. Doesn't seem to be the case for me.

The phone didn't wait for me to dial, it just started ringing once off hook.

I would like to dial a 4 digit extension at the remote site, and get it passed on to the PABX. The issue is, there should be no limit on what the extension is, on any of the ports (phones).

Yes, with PLAR, it is the PBX that gives you dialtone and collect digits, not the router.

Now, I'm not sure what could be happening. You say that you hear a ringback tone, can you find out on the PBX which number is being called? Perhaps someone will answer if you let it ring long enough.

To the same effect, on the PBX router, please enable "term mon" and "debug vpm signal", then pick-up a phone remotely, so we can see what is going on.

For sanity check, please post you config too.

Ah, I switch to a fresh port and reconfigured everything and it work!

Proceeded to wipe out the previous configurations and dial-peers and started over with a fresh slate. I had a strange moment when the dial tone from the PBX seems to be busy for 30 seconds before I get a dial time but that soon passed and didn't crop up again.

Thanks guys for all the assist!

Getting Started

Find answers to your questions by entering keywords or phrases in the Search bar above. New here? Use these resources to familiarize yourself with the community: