Redirect PSTN Calls CallManager 5.x to Ericsson MD110 and viceversa

Answered Question

Hello everybody

I have to implement the following scenario:


The Calls from PSTN received on h323 gateway with 1E1 ISDN PRI must be redirect to 2E1 ISDN PRI connected to PBX Ericsson to the extensions that belong to Ericsson, and viceversa when a call is made must go first to 2E1 ISDN PRI on h323 gateway and then redirect to 1E1 ISDN PRI to PSTN.


The 97% of the extensions belong to the PBX Ericsson and the rest of extension belong to the CallManager 5.x

The Equipment is:

CallManager 5.X with 2 IP PHONES

h323 gateway with two E1 ISDN PRI

PBX Ericsson with 50 extensiones.


has anybody ever worked on simular scenario?

I know that I must work with voice translations but I dont have it clear sending calles between the two E1 ISDN PRI.


Any help will be apreciated.

Regards

Walter

Correct Answer by pcameron about 9 years 3 months ago

Looks like the PBX is sending overlap -


*Feb 22 01:17:34.956: ISDN Se0/0/1:15 Q931: RX <- SETUP pd = 8 callref = 0x1777

Bearer Capability i = 0x8090A3

Standard = CCITT

Transfer Capability = Speech

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0xA18396

Preferred, Channel 22

Progress Ind i = 0x8183 - Origination address is non-ISDN

Calling Party Number i = 0x0181, '1025'

Plan:ISDN, Type:Unknown

Called Party Number i = 0x81, '64' ********

Plan:ISDN, Type:Unknown



There is only the access code of '64' and the sending complete flag is not set in the setup message.


Add the command 'isdn overlap-recieving' under interface serial 0/0/1:15 to allow the gateway to buffer the recieved digits.



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pcameron Wed, 02/20/2008 - 16:25
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This is called tandem switching and it works very well on IOS voice gateways.


You connect the PSTN line to one E1 interface of the router and the PBX goes to another E1 interface. By using dial peers to match on particular called numbers, you can switch calls between the E1 voice ports, or send calls to the VOIP legs.


The best way to do this is seperate the dial plans on the PBX and the call manager as this makes the dial peer configuration far easier to summarise.


E1 clocking is very important in this scenario so you must take a clock reference from the PSTN and use this to drive the PBX - this ensures a consistent clock reference on all devices. The PBX connects to the router via a E1 cross over cable (diagram below) and you set the D channel to be ISDN network side.


1 ---- 4

2 ---- 5

4 ---- 1

5 ---- 2


Following is a configuration that shows the concepts - we will assume the number ranges are as follows :


PSTN sends numbers in range 5551000 - 5553999

5551000 through to 5552999 are for existing PBX

5553000 through to 5553999 are sent to CallManager


Users on PBX and CCM dial 0 for an outside line. The PBX and the CCM are configure to leave this leading 0 on the called number (so it can be used as a 'steering' code for outwards calls through the gateway)


Using the mix of POTS and VOIP dial peers, you can switch calls based on the called numbers between the PBX and PSTN trunks, or towards the CallManager/VOIP.


Keep in mind that this is only an example and you may need to use other translations to handle your particular number ranges.



!

network-clock-participate wic 0

network-clock-select 1 e1 0/0/0

!


isdn switch-type primary-net5


!

controller e1 0/0/0

description - connection to PSTN

pri-group timeslots 1-31

!

controller e1 0/0/1

description - connection to PBX

clock source internal

pri-group timeslots 1-31

!


!

interface serial 0/0/0:15

isdn switch-type primary-net5

isdn incoming-voice voice

!

interface serial 0/0/0:15

isdn switch-type primary-net5

isdn incoming-voice voice

isdn protocol-emulate network

!


!

dial-peer voice 1 pots

description - enable Direct Inward Dial for PSTN trunk

incoming-called-number .

direct-inward-dial

port 0/0/0:15

!

dial-peer voice 2 pots

description - enable Direct Inward Dial for PBX trunk

incoming-called-number .

direct-inward-dial

port 0/0/1:15

!

dial-peer voice 3 pots

description - Calls from PSTN towards PBX, number range is 5551XXX - 5552XXX

destination-pattern 555[1,2]...

forward-digits all

port 0/0/1:15

!

dial-peer voice 3 voip

description - Calls from PSTN towards CCM, number range is 5553XXX

destination-pattern 5553...

session target ipv4:10.1.1.1

codec g711ulaw

dtmf-relay h245-alphanumeric

no vad

!

dial-peer voice 4 voip

description - calls from PBX to CCM using 4 digit numbers in range 3XXX

destination-pattern 3...

session target ipv4:10.1.1.1

codec g711ulaw

dtmf-relay h245-alphanumeric

no vad

!

dial-peer voice 5 pots

description - outgoing PSTN calls using leading 0 access code. Access code is stripped off

destination-pattern 0

port 0/0/0:15

!


We have many customers using this feature and it works well. It's a good way to transistion old PBX systems to VOIP, which is even more necessary now given the news that Ericsson have sold off their enterprise PBX group ...


http://www.computerworld.com.au/index.php/id;765736722



HI pcameron, thank very much for your time and help.


I just have two question about outgoing PSTN calls.


1. In the dial-peer voice 5 pots on the destination-pattern, should I need to add T after 0 (access code)in order to send the digits to PSTN?


2. When users from PBX call to PSTN dial 0T, is there necesary to configure a dial-peer pots for port 0/0/1:15??


Could you please help to get this clear.


Thank very much in advance.

Regads

Walter

pcameron Thu, 02/21/2008 - 14:54
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  • Cisco Employee,

1) For dial peer 5, just the destination pattern of 0 is adequate - the Q931 setup messages that come from the PBX and the CCM are in 'enbloc' format (setup message has a complete called number), so the gateway router knows there are no further digits to follow and it will match the leading 0 on the called number with this dial peer. The 0 will be stripped off the other numbers sent out the PSTN connection.


2) For PBX users, you need an outgoing POTS dial peer to send their calls to the main PSTN port. You might as well use the existing dial peer that the CCM is using for outgoing calls. As mentioned above, as long as the PBX sends the full called number enbloc, this dial peer is adequate.


Hi pcameron, and thank very much for helping me.


I am working in the scenario:

1. The calls from PSTN to PBX are forwarding through the h323 gateway, working OK.

2. The customers requested that the GDN 2507100 was received on CCM DN 2000, working OK.

3. From CCM to PSTN the calls are working fine using Access Code 64.


I just have one issue the calls from PBX TO PSTN through the gateway DO NOT WORK.

I run debug isdn q931 and I dont see any called number from PBX SIDE and I received from PSTN SIDE a disconection because of Invalid number or Incomplete number.


You enclosed my configuration beside the debug.

Do you think it might be a Overlap on PBX SIDE ERICSON MD110, because I'm using Enbloc on the gateway?


Regards

Walter Flores





Correct Answer
pcameron Fri, 02/22/2008 - 05:55
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  • Cisco Employee,

Looks like the PBX is sending overlap -


*Feb 22 01:17:34.956: ISDN Se0/0/1:15 Q931: RX <- SETUP pd = 8 callref = 0x1777

Bearer Capability i = 0x8090A3

Standard = CCITT

Transfer Capability = Speech

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0xA18396

Preferred, Channel 22

Progress Ind i = 0x8183 - Origination address is non-ISDN

Calling Party Number i = 0x0181, '1025'

Plan:ISDN, Type:Unknown

Called Party Number i = 0x81, '64' ********

Plan:ISDN, Type:Unknown



There is only the access code of '64' and the sending complete flag is not set in the setup message.


Add the command 'isdn overlap-recieving' under interface serial 0/0/1:15 to allow the gateway to buffer the recieved digits.



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