I cann't access for this link..

Hi,

please try this link, it should work if your not logged in.

http://www.cisco.com/en/US/products/sw/voicesw/ps2169/products_configuration_guide_chapter09186a00806674c3.html#wp1351962

Or search for

"Cisco Unified SRST 4.0: Setting Up the Network"

on www.cisco.com

Bye

Jo

didn't help

As I use IOS later 12.3 I used

applicatiob

service default

but this didn't help. What can be the problem??

Post your full config here and we can confirm exactly how it is set up and make config suggestions.

!

controller E1 0/0/0

framing NO-CRC4

pri-group timeslots 1-31 service mgcp

!

interface Serial0/0/0:15

no ip address

encapsulation hdlc

isdn switch-type primary-net5

isdn incoming-voice voice

isdn bind-l3 ccm-manager

no cdp enable

!

voice-port 0/0/0:15

cptone RU

!

voice-port 0/1/0

!

voice-port 0/1/1

!

ccm-manager fallback-mgcp

ccm-manager redundant-host 10.10.0.2

ccm-manager mgcp

ccm-manager music-on-hold

ccm-manager config server 10.10.0.1

ccm-manager config

!

mgcp

mgcp call-agent 10.10.0.1 2427 service-type mgcp version 0.1

mgcp rtp unreachable timeout 1000 action notify

mgcp modem passthrough voip mode nse

mgcp package-capability rtp-package

mgcp package-capability sst-package

mgcp package-capability pre-package

no mgcp package-capability res-package

no mgcp package-capability fxr-package

no mgcp timer receive-rtcp

mgcp sdp simple

mgcp rtp payload-type g726r16 static

!

mgcp profile default

!

!

!

dial-peer voice 4 pots

tone ringback alert-no-PI

destination-pattern [2346].....

no digit-strip

direct-inward-dial

port 0/0/0:15

!

dial-peer voice 80 pots

incoming called-number .

direct-inward-dial

port 0/0/0:15

!

!

!

!

call-manager-fallback

max-conferences 8 gain -6

ip source-address 10.11.10.1 port 2000

max-ephones 128

max-dn 256

transfer-pattern 0.

transfer-pattern 0..

transfer-pattern 2.....

transfer-pattern 3.....

transfer-pattern 4.....

transfer-pattern 6.....

transfer-pattern 82.......

transfer-pattern 80..........

!

!

end

TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT

4 pots up up [2346]..... 0 up 0/0/0:15

20001 pots up up 1027$ 0 50/0/1

20002 pots up up 1001$ 0 50/0/2

20003 pots up up 1026$ 0 50/0/3

20004 pots up up 1030$ 0 50/0/4

20005 pots up up 1002$ 0 50/0/5

20006 pots up up 1707$ 0 50/0/6

20007 pots up up 1025$ 0 50/0/7

20008 pots up up 1031$ 0 50/0/8

20009 pots up down 0 50/0/9

20010 pots up down 0 50/0/10

20011 pots up down 0 50/0/11

20012 pots up down 0 50/0/12

20013 pots up down 0 50/0/13

20014 pots up down 0 50/0/14

20015 pots up down 0 50/0/15

20016 pots up down 0 50/0/16

20017 pots up down 0 50/0/17

20018 pots up down 0 50/0/18

20019 pots up down 0 50/0/19

20020 pots up down 0 50/0/20

20021 pots up down 0 50/0/21

20022 pots up down 0 50/0/22

Cisco IOS Version 12.3(11r)T2

Please add the command 'call application alternate default' to the main config. This will allow the ISDN D channel operation to drop back to local router processing. Without this command, the router will be trying to make a connection to the call manager all the time, so the D channel will not come up .

Your ephones register in the range 1XXX (4 digits) so once the D channel is working correctly in SRST, you might also need to translate the called number that comes from the telco to this 4 digit range.

Once you add the above command, wait a moment and enabled 'debug isdn q931', then make an inwards call. Capture this log and paste it here so we can confirm the number range being sent to you.

Feb 4 13:16:27.701: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8 callref = 0x005D

Sending Complete

Bearer Capability i = 0x8090A3

Standard = CCITT

Transfer Capability = Speech

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0xA18381

Preferred, Channel 1

Progress Ind i = 0x8281 - Call not end-to-end ISDN, may have in-band info

Called Party Number i = 0x81, '422864'

Plan:ISDN, Type:Unknown

Feb 4 13:16:27.757: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x805D

Cause i = 0x809B - Destination out of order

Feb 4 13:16:31.773: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8 callref = 0x005E

Sending Complete

Bearer Capability i = 0x8090A3

Standard = CCITT

Transfer Capability = Speech

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0xA18381

Preferred, Channel 1

Progress Ind i = 0x8281 - Call not end-to-end ISDN, may have in-band info

Called Party Number i = 0x81, '422864'

Plan:ISDN, Type:Unknown

Feb 4 13:16:31.797: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x805E

Cause i = 0x809B - Destination out of order

Feb 4 13:16:41.438: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8 callref = 0x005F

Sending Complete

Bearer Capability i = 0x8090A3

Standard = CCITT

Transfer Capability = Speech

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0xA18381

Preferred, Channel 1

Progress Ind i = 0x8281 - Call not end-to-end ISDN, may have in-band info

Called Party Number i = 0x81, '422864'

Plan:ISDN, Type:Unknown

Feb 4 13:16:41.446: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x805F

Cause i = 0x809B - Destination out of order

Outgoing calls - say that line is busy

call application alternate default - is added

Very good - D channel is now up :-)

For inwards calls, the number range provided by your telco is 6 digits :

Called Party Number i = 0x81, '422864'

But this called number does not correspond to any registered ephone numbers :

20001 pots up up 1027$ 0 50/0/1

20002 pots up up 1001$ 0 50/0/2

20003 pots up up 1026$ 0 50/0/3

20004 pots up up 1030$ 0 50/0/4

20005 pots up up 1002$ 0 50/0/5

20006 pots up up 1707$ 0 50/0/6

20007 pots up up 1025$ 0 50/0/7

20008 pots up up 1031$ 0 50/0/8

So you need to translate the called number/s from the telco to your internal range.

Let us suppose the numbers from the telco are in the range 422XXX, and you want to map these over to your extension range of 1XXX for SRST.

The easiest way is to use a translation profile to change the 422 to 1, and leave the trailing digits unchanged.

!

voice translation-rule 111

rule 1 /^422/ /1/

!

voice translation-profile test

translate called 111

!

voice-port 0/0/0:15

translation-profile incoming test

!

Almost there! Change the translation rule to suit your number and give it another go.

for outwards calls to work, add the following config -

!

call-manager-fallback

secondary-dialtone 0

!

dial-peer voice 2222 pots

destination-pattern 0T

port 0/0/0:15

!

To make calls to the PSTN , dial 0 then the rest of number . After 10 second timeout, the call should be sent out the PRI without the leading 0.

call-manager-fallback

secondary-dialtone 0

!

dial-peer voice 2222 pots

destination-pattern 0T

port 0/0/0:15

!

This config also didn't help me.. :(

I wait for 10 seconds and receive busy signal

debug isdn q931 didn't show anything

I think You are right :)

but by the words, internal and external numbers like this:

1004 422864

1005 422865

1006 422866

so voice translation-rule 111

rule 1 /^42286/ /100/

Am I right?