Solved: CME and analog voice gateways

Heinz Schwarzfeuer · ‎03-10-2015

Hi,

we need to connect a Cisco CME (2921 ISR G2) to the POTS using 16 analog lines.

Instead of using several FXO cards directly, the customer decided to use 2 8-port FXO gateways (Grandstream GXW4108FXO IP Analog Gateway).

Can anyone give some brief hints about how to configure these on the Cisco CME side? Are they just addressed as normal dial peers or is there something else to do?

I´m also wondering how to distribute outgoing POTS calls between this two units to utilize them both (and not only one).

Any hints would be greatly appreciated.

Many thanks

Heinz

Vivek Batra · ‎03-10-2015

Hi Heinz,

1. Grandstream gateways can be setup in either of the following ways;

With SIP accounts configured, OR

Without SIP accounts

When you configure Greandstream gateway with SIP accounts configured, gateway acting as a normal endpoint and will try to register as SIP end point with CME. You can ignore this to simplify the configuration.

Use the Grandstream gateway "Without SIP accounts" and in this case, you simply have to configure the SIP Server (CME) to perform forwarding of the SIP INVITE message with the FXO destination number to the gateways IP Address. This will be achieved using dial-peers. The Grandstream gateway will receive the digits and immediately forward them on the FXO lines to the destination PSTN.

2. Please check "Hunt Groups and Preferences" in the below document;

Hwww.cisco.com/c/en/us/td/docs/ios/12_2/voice/configuration/guide/fvvfax_c/vvfpeers.html#wp1244361

Thanks

Vivek

View solution in original post

Vivek Batra · ‎03-18-2015

Hi Heinz,

I will always prefer sending number enblock to the gateway. I tried to check with grandstream and found the following;

Seems that this parameter is default set to 2 (in your case). So when you send the number to grandstream in enblock (using 9T), it actually gives you dial tone.

Now when you changed it to destination-pattern 0, still it gives you dial tone and then caller dials the further number.

Try changing this parameter to 1 and destination-pattern to '9T'. Now when grandstream will receive the number in SIP INVITE, it should immediately off-hook the FXO port and dial out the number.

Hope it will help you.

Thanks

Vivek

View solution in original post

Vivek Batra · ‎03-10-2015

Hi Heinz,

Grandstream gateways only supports SIP so you need to create normal SIP enabled dial-peers in CME directing to Grandstream gateway.

To distribute calls between two different gateways, you can configure "preference" under dial-peers in CME.

Thanks

Vivek

Heinz Schwarzfeuer · ‎03-10-2015

Hi Vivek,

thanks. The guy who is configuring the gateways said that there is also authentication required (username/password). How can this be done under a normal dial-peer?

Regarding the distribution of calls: Could you give an example? What i´m thinking about is
of how to evenly distribute the outgoing calls to the two dial peers, so that we don´t end up in a situation where the lines of one analog gateway are all busy and the other are free.

Many thanks

Heinz

Vivek Batra · ‎03-10-2015

Hi Heinz,

1. Grandstream gateways can be setup in either of the following ways;

With SIP accounts configured, OR

Without SIP accounts

When you configure Greandstream gateway with SIP accounts configured, gateway acting as a normal endpoint and will try to register as SIP end point with CME. You can ignore this to simplify the configuration.

Use the Grandstream gateway "Without SIP accounts" and in this case, you simply have to configure the SIP Server (CME) to perform forwarding of the SIP INVITE message with the FXO destination number to the gateways IP Address. This will be achieved using dial-peers. The Grandstream gateway will receive the digits and immediately forward them on the FXO lines to the destination PSTN.

2. Please check "Hunt Groups and Preferences" in the below document;

Hwww.cisco.com/c/en/us/td/docs/ios/12_2/voice/configuration/guide/fvvfax_c/vvfpeers.html#wp1244361

Thanks

Vivek

Heinz Schwarzfeuer · ‎03-11-2015

Hi Vivek,

thanks.

The CME configuration is as follows at the moment:

voice translation-rule 11
rule 3 /^0\(.*\)/ /\1/
!
!
voice translation-profile OUTGOING_TRANSLATION
translate called 11
!
dial-peer voice 102 voip
description Grandstream GXW4108FXO AVG01
translation-profile outgoing OUTGOING_TRANSLATION
destination-pattern 0T
session protocol sipv2
session target ipv4:10.0.2.12
no voice-class sip localhost
no voice-class sip outbound-proxy
dtmf-relay rtp-nte
codec g711ulaw
!

The translation rule is for stripping off the leading "0" we use as a prefix to external numbers.

However, it´s not working. The SIP calls are directed to the Grandstream device, but from there on they are not going to the outside POTS.

Has anyone a working configuration example for this combination?

Thanks

Heinz

Vivek Batra · ‎03-11-2015

Hi Heinz,

What is the SIP error cause getting from Grandstream? We can get it in CME using "debug ccsip messages" command.

Thansk

Vivek

Heinz Schwarzfeuer · ‎03-12-2015

Hi Vivek,

it´s now working. The problem was the "T" in the destination pattern of the dial peer:

destination-pattern 0T

So the correct configuration is:

dial-peer voice 102 voip
description Grandstream GXW4108FXO AVG01
translation-profile outgoing OUTGOING_TRANSLATION
destination-pattern 0
session protocol sipv2
session target ipv4:10.0.2.12
no voice-class sip localhost
no voice-class sip outbound-proxy
dtmf-relay rtp-nte
codec g711ulaw

Many thanks for your support!

Heinz

Vivek Batra · ‎03-12-2015

Hi Heinz,

I'm curious to know if this was the reason of call not getting pass through grandstream gateway. I believe that with destination-pattern configured now ("0"), CME won't be able to match variable length number and may send incorrect number outside.

I may be wrong also as I'm running with older version of CME and newer version may have some improvements.

Can you please let me know with earlier configuration of destination-pattern ("0T"), whether call was hitting grandstream or not and if yes, what was the called number received by grandstream?

Indeed I don't want you to put in trouble once issue is already been fixed :)

Thanks

Vivek

Heinz Schwarzfeuer · ‎03-13-2015

Hi Vivek,

it just didn´t work with the "0T".

With "0" only you are immediately connected to the Grandstream after dialing 0 and you hear the DTMF tones while entering the subsequent numbers; after a timeout of about 10 seconds the Grandstream does the POTS connection to the number you have dialed.

However, this is all a bit strange. I would like to send out the dialed number at once (like I do with all the other VOIP dialpeers, i.e. 0.........) to the Grandstream, but it´s not working.

Does anyone know how the Grandstream needs to be configured to work with the CME and can maybe post a configuration example?

Thanks

Heinz

Vivek Batra · ‎03-18-2015

Hi Heinz,

I will always prefer sending number enblock to the gateway. I tried to check with grandstream and found the following;

Seems that this parameter is default set to 2 (in your case). So when you send the number to grandstream in enblock (using 9T), it actually gives you dial tone.

Now when you changed it to destination-pattern 0, still it gives you dial tone and then caller dials the further number.

Try changing this parameter to 1 and destination-pattern to '9T'. Now when grandstream will receive the number in SIP INVITE, it should immediately off-hook the FXO port and dial out the number.

Hope it will help you.

Thanks

Vivek

Heinz Schwarzfeuer · ‎03-18-2015

Hi Vivek,

thanks, that´s exactly the setting i was looking for. It´s working now!

Best regards

Heinz

Vivek Batra · ‎03-18-2015

Hi Heinz,

Gr8 we could fixed it.

Thanks

Vivek

CiscoAzs · ‎05-23-2015

Hi Heinz Schwarzfeuer

Сould you show GXW4108 settings, I have the same problem?