Solved: Re: 0x82AC18 - Requested circuit/channel not available

stoofer · ‎11-06-2021

hello guys

we are getting an ISDN error on incoming calls on our T1 trunks connected to an old PBX, any ideas what might be the reason for these errors ?

thanks in advace.

*Nov 2 00:16:25.028: ISDN Se0/1/0:23 Q931: RX <- SETUP pd = 8 callref = 0x01DB

Bearer Capability i = 0x8090A2

Standard = CCITT

Transfer Capability = Speech

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0xA98397

Exclusive, Channel 23

Calling Party Number i = 0x2183, '5201205545'

Plan:ISDN, Type:National

Called Party Number i = 0xA1, '912503507781'

Plan:ISDN, Type:National

*Nov 2 00:16:25.028: ISDN Se0/1/0:23 Q931: Received SETUP callref = 0x81DB callID = 0x000F switch = primary-ni interface = Network

*Nov 2 00:16:25.028: ISDN Se0/1/0:23 **ERROR**: CCPRI_NegotiateBchan: b channel busy 23, event 0x90 excl? 1

*Nov 2 00:16:25.028: %ISDN-6-CHAN_UNAVAILABLE: Interface Se0/1/0:23 Requested Channel 23 is not available

VG01#

*Nov 2 00:16:25.032: ISDN Se0/1/0:23 **ERROR**: CCPMSG_InCall: ChannelID IE invalid 44(0x2C) rejecting call

*Nov 2 00:16:25.032: ISDN Se0/1/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x81DB

Cause i =0x82AC18 - Requested circuit/channel not available

VG01#

*Nov 2 00:16:50.696: ISDN Se0/1/0:23 **ERROR**: CCPCC_TApplnAckExpiry: Application Ack Timer expired. b channel 23 cref 0x81D9 call_id 0x000E

*Nov 2 00:16:50.696: %ISDN-4-STATUS: Interface Se0/1/0:23 Application AckTimer expired. st 0x1407 ev 0xE cid 0x81D9 cref 0x0

VG01#

*Nov 2 00:16:50.696: ISDN Se0/1/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x81D9

VG01#

Roger Kallberg · ‎11-06-2021

Thanks, however that’s missing the serial interface part.

Based on your description I would suggest that you reconfigure the controllers to not use MGCP as that’s a control protocol that uses CM for layer 3 signalling.

View solution in original post

Roger Kallberg · ‎11-06-2021

The SIP error is related to ISDN Requested circuit/channel not available.

As you have setup your ISDN cirques to be controlled by MGCP is there even a CM in your system landscape?

If you do not have a Call Manager in the picture reconfigure your controllers with this.

trunk group Siemens_PBX
max-calls any 24
hunt-scheme sequential
!
controller T1 0/0/0
cablelength long 0db
pri-group timeslots 1-24
trunk-group Siemens_PBX timeslots 1-24 preference 2
!
controller T1 0/1/0
cablelength long 0db
pri-group timeslots 1-24
trunk-group Siemens_PBX timeslots 1-24 preference 1

You will have to remove the current configuration of the pri-group before you can reconfigure this. To do remove this you’ll have to shut down the voice ports and controllers, then remove the current pri-group by “no pri-group timeslots 1-24 service mgcp”.

Please be advised as this would remove and then recreate your voice ports and serial interfaces you would need to reconfigure these with the applicable configuration that you currently have on these.

interface Serial0/0/0:23
no ip address
encapsulation hdlc
no cdp enable
isdn switch-type primary-ni
isdn timer T310 120000
isdn protocol-emulate network
isdn incoming-voice voice
!
interface Serial0/1/0:23
no ip address
encapsulation hdlc
no cdp enable
isdn switch-type primary-ni
isdn timer T310 120000
isdn protocol-emulate network
isdn incoming-voice voice

View solution in original post

Roger Kallberg · ‎11-09-2021

That is good news. For your remaining issue you could do this in a SIP profile that you attach on the inbound direction on your SIP dial peer from the service provider.

voice class sip-profiles 10
 request ANY sip-header From modify "From:(.*)(<sip:.*@.*>)" "From: \2" 
 response ANY sip-header From modify "From:(.*)(<sip:.*@.*>)" "From: \2"
!
voice service voip
 sip
  sip-profiles inbound
!
dial-peer voice 100 voip
 voice-class sip profiles 10 inbound

For detailed info on dial peers and how they operate have a look at this fantastic document In Depth Explanation of Cisco IOS and IOS-XE Call Routing - Cisco

You can also verify the working of a SIP profile by using this tool. SIP-Profile Test Tool (cisco.com)

View solution in original post

Roger Kallberg · ‎11-06-2021

Can you please share your configuration for the port, interface, controller and dial peers from the gateway?

stoofer · ‎11-06-2021

!
voice-port 0/0/0:23
!
voice-port 0/1/0:23
!
!

!
trunk group Siemens_PBX
max-calls any 30
hunt-scheme sequential
!
controller T1 0/0/0
cablelength long 0db
pri-group timeslots 1-24 service mgcp
trunk-group Siemens_PBX timeslots 1-24 preference 2
!
controller T1 0/1/0
cablelength long 0db
pri-group timeslots 1-24 service mgcp
trunk-group Siemens_PBX timeslots 1-24 preference 1
!
voice translation-rule 11
rule 1 /.+$...........$$/ /\1/
!
voice translation-rule 21
rule 1 /^9$.*$/ /\1/
!
!
voice translation-profile SIP_INCOMING
translate called 11
!
voice translation-profile SIP_OUTGOING
translate called 21
!
dial-peer voice 91 voip
description *** DIAL PEER FOR OUTGOING PSTN CALLS ***
translation-profile incoming SIP_OUTGOING
destination-pattern 9T
session protocol sipv2
session target ipv4:*.*.*.*:5060
voice-class sip early-offer forced
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 911 voip
description *** DIAL PEER FOR 911CALLS ***
destination-pattern 911
session protocol sipv2
session target ipv4:*.*.*.*:5060
voice-class sip early-offer forced
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 100 voip
description *** INBOUND CALLS FROM SERVICE PROVIDER ***
translation-profile incoming SIP_INCOMING
session protocol sipv2
incoming called-number .
voice-class codec 1
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 1000 pots
trunkgroup Siemens_PBX
description pots calls from pbx
incoming called-number .
forward-digits all
!
dial-peer voice 2000 pots
trunkgroup Siemens_PBX
description calls to Siemens PBX
destination-pattern 1402.......
!

Roger Kallberg · ‎11-06-2021

Thanks, however that’s missing the serial interface part.

Based on your description I would suggest that you reconfigure the controllers to not use MGCP as that’s a control protocol that uses CM for layer 3 signalling.

stoofer · ‎11-06-2021

Hello Roger,

here are the serial interface configs, pls advice and thanks in advance!

!
interface Serial0/0/0:23
no ip address
encapsulation hdlc
no cdp enable
isdn switch-type primary-ni
isdn timer T310 120000
isdn protocol-emulate network
isdn incoming-voice voice
!
interface Serial0/1/0:23
no ip address
encapsulation hdlc
no cdp enable
isdn switch-type primary-ni
isdn timer T310 120000
isdn protocol-emulate network
isdn incoming-voice voice
!

stoofer · ‎11-06-2021

thanks for your help so far and advice that these errors are related

here is the other error we are getting from the sip side when receiving calls from the provider

*Nov 1 20:14:54.065: //415/822D05688518/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 123.54.19.21:5060;branch=z9hG4bK08Bbec859587c2f451e
From: "Unavailable" <sip:+3112964334@123.54.19.21:5060>;tag=gK0816e333
To: <sip:+12533023294@10.123.4.8:5060>;tag=32CE7CE4-973
Date: Thu, 01 Nov 2021 20:14:54 GMT
Call-ID: 235415833_62895109@123.54.19.21
CSeq: 404063 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.7.3.M3
Reason: Q.850;cause=44
Session-ID: b8f3b157290f53bb89bbb43de2984ae7;remote=c889e4fb91db50a9b2059df423048be8
Content-Length: 0

Roger Kallberg · ‎11-06-2021

The SIP error is related to ISDN Requested circuit/channel not available.

As you have setup your ISDN cirques to be controlled by MGCP is there even a CM in your system landscape?

If you do not have a Call Manager in the picture reconfigure your controllers with this.

trunk group Siemens_PBX
max-calls any 24
hunt-scheme sequential
!
controller T1 0/0/0
cablelength long 0db
pri-group timeslots 1-24
trunk-group Siemens_PBX timeslots 1-24 preference 2
!
controller T1 0/1/0
cablelength long 0db
pri-group timeslots 1-24
trunk-group Siemens_PBX timeslots 1-24 preference 1

You will have to remove the current configuration of the pri-group before you can reconfigure this. To do remove this you’ll have to shut down the voice ports and controllers, then remove the current pri-group by “no pri-group timeslots 1-24 service mgcp”.

Please be advised as this would remove and then recreate your voice ports and serial interfaces you would need to reconfigure these with the applicable configuration that you currently have on these.

interface Serial0/0/0:23
no ip address
encapsulation hdlc
no cdp enable
isdn switch-type primary-ni
isdn timer T310 120000
isdn protocol-emulate network
isdn incoming-voice voice
!
interface Serial0/1/0:23
no ip address
encapsulation hdlc
no cdp enable
isdn switch-type primary-ni
isdn timer T310 120000
isdn protocol-emulate network
isdn incoming-voice voice

stoofer · ‎11-06-2021

Hi Roger,

In the old setup there was a CM but now there is none anymore ! We are
migrating and we inherited the vg and the old Siemens pbx, and we are
trying to make this work using a new provider (sip).
Can you advice which change I have to make ?

Many thanks for your time and help !

Roger Kallberg · ‎11-06-2021

Look at my last response.

As you do not have a CM anymore you can not use MGCP to control your ISDN circuits. This is the root cause of your problem.

stoofer · ‎11-08-2021

Hi Roger,

Outbound calls coming from the old PBX are giving this error : (I blanked out the phone nrs for privacy)

*Nov 8 23:37:51.712: ISDN Se0/1/0:23 Q931: RX <- SETUP pd = 8 callref = 0x0441
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98397
Exclusive, Channel 23
Calling Party Number i = 0x2183, '6********'
Plan:ISDN, Type:National
Called Party Number i = 0xA1, '9***********'
Plan:ISDN, Type:National
*Nov 8 23:37:51.712: ISDN Se0/1/0:23 Q931: Received SETUP callref = 0x8441 callID = 0x0005 switch = primary-ni interface = Network
*Nov 8 23:37:51.712: ISDN Se0/1/0:23 **ERROR**: CCPRI_NegotiateBchan: b channel busy 23, event 0x90 excl? 1
*Nov 8 23:37:51.712: %ISDN-6-CHAN_UNAVAILABLE: Interface Se0/1/0:23 Requested Channel 23 is not available
CISCO-VG##
*Nov 8 23:37:51.712: ISDN Se0/1/0:23 **ERROR**: CCPMSG_InCall: ChannelID IE invalid 44(0x2C) rejecting call
*Nov 8 23:37:51.712: ISDN Se0/1/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x8441
Cause i = 0x82AC18 - Requested circuit/channel not available
CISCO-VG##
*Nov 8 23:38:08.908: ISDN Se0/1/0:23 **ERROR**: CCPCC_TApplnAckExpiry: Application Ack Timer expired. b channel 23 cref 0x843F call_id 0x0004
*Nov 8 23:38:08.908: %ISDN-4-STATUS: Interface Se0/1/0:23 Application AckTimer expired. st 0x1407 ev 0x4 cid 0x843F cref 0x0
CISCO_VG#
*Nov 8 23:38:08.908: ISDN Se0/1/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x843F
CISCO_VG#
*Nov 8 23:38:29.628: //-1/xxxxxxxxxxxx/CCAPI/ccAppShutdownMode:

!
controller T1 0/0/0
cablelength long 0db
pri-group timeslots 1-24
trunk-group Siemens_PBX timeslots 1-24 preference 2
!
!
controller T1 0/1/0
cablelength long 0db
pri-group timeslots 1-24
trunk-group Siemens_PBX timeslots 1-24 preference 1
!

!
interface Serial0/0/0:23
no ip address
encapsulation hdlc
no cdp enable
isdn switch-type primary-ni
isdn timer T310 120000
isdn protocol-emulate network
isdn incoming-voice voice
!
interface Serial0/1/0:23
no ip address
encapsulation hdlc
no cdp enable
isdn switch-type primary-ni
isdn timer T310 120000
isdn protocol-emulate network
isdn incoming-voice voice
!

Roger Kallberg · ‎11-08-2021

Can you please show the output from show isdn status so that we can see the status of your T1 ISDN connections?

stoofer · ‎11-08-2021

Roger, I am not sure if my other post about the dial-peers is linked/related to this issue ?

https://community.cisco.com/t5/ip-telephony-and-phones/need-help-with-dialpeers/m-p/4498949

here is the output from sh isdn stat command :

Global ISDN Switchtype = primary-ni
ISDN Serial0/0/0:23 interface
******* Network side configuration *******
dsl 0, interface ISDN Switchtype = primary-ni
Layer 1 Status:
ACTIVE
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 0 CCBs = 0
The Free Channel Mask: 0x807FFFFF
Number of L2 Discards = 0, L2 Session ID = 1
ISDN Serial0/1/0:23 interface
******* Network side configuration *******
dsl 1, interface ISDN Switchtype = primary-ni
Layer 1 Status:
ACTIVE
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 1 CCBs = 0
The Free Channel Mask: 0x807FFFFF
Number of L2 Discards = 0, L2 Session ID = 1
Total Allocated ISDN CCBs = 0

Roger Kallberg · ‎11-08-2021

Likely it’s related, but it does not have to be. Your ISDN interfaces are at least up, that’s a good start. Please do a debug isdn q931, debug voip ccapi inout and debug ccsip message to see what happens with the call when it hits the router. Please post the output as an attached file so that it’s easier to check.

stoofer · ‎11-08-2021

output attached Roger, I masked out the phone nrs for privacy

Roger Kallberg · ‎11-08-2021

The call disconnects with

Cause i = 0x8281 - Unallocated/unassigned number

From what it looks the call does not match an outbound dial peer and that’s what causes the call to fail.