02-04-2008 02:20 AM - edited 03-15-2019 08:35 AM
when phones registers with SRST on gateway , appear dial-peers like 1001$,1002$,1003$ . Phones can call internal numbers, but they don't see PRI
Here is the dial-peer:
dial-peer voice 4 pots
tone ringback alert-no-PI
destination-pattern [234].....
no digit-strip
direct-inward-dial
port 0/0/0:15
02-04-2008 02:57 AM
Hi,
I am assuming that when you say they don't see PRI you mean that they cannot make outgoing calls.
This sounds like a problem in you SRST config.
Please see the following guide on how to configure SRST for MGCP.
http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a0080144630.shtml
Hope this helps.
If it does please rate.
Regards
Dave
02-04-2008 03:01 AM
Hi,
try
call application alternate [application-name]
or
service [alternate-name | default] service-name location
in the dialpeer config.
See
Bye
Jo
02-04-2008 03:39 AM
I cann't access for this link..
02-04-2008 03:56 AM
Hi,
please try this link, it should work if your not logged in.
Or search for
"Cisco Unified SRST 4.0: Setting Up the Network"
Bye
Jo
02-04-2008 04:41 AM
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??
02-04-2008 04:43 AM
Post your full config here and we can confirm exactly how it is set up and make config suggestions.
02-04-2008 04:49 AM
!
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
02-04-2008 05:01 AM
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
02-04-2008 05:09 AM
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.
02-04-2008 05:19 AM
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
02-04-2008 05:36 AM
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.
02-04-2008 05:41 AM
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.
02-04-2008 06:30 AM
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
02-04-2008 05:44 AM
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?
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