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Need help ASAP - 4351 with NIM-2MFT-T1/E1 - ISDN PRI - one-way audio problem

voip7372
Level 4
Level 4

This is the first time I've configured an ISDN PRI (USA) on a 4000 series router with the new NIM-2MFT-T1/E1 module and PVDM4.   I have a PVDM4 on the motherboard as well as one on the T1 module.

My problem is I have NO audio coming IN from the outside call (ISDN PRI/T1).  Audio going OUT on the ISDN PRI is fine - if I speak into my desk phone at the office, I can hear my voice on my mobile phone when I call it).  

Audio internally between phones is fine and between phones at other locations is fine (if I call internally from this location to another location).

If anyone can help me solve this, I'd greatly appreciate it!  I'm onsite for this office move right now and it's not working which is a huge problem.  I see no errors when I do a debug isdn q931.  The T1 controller is up with no errors and the ISDN status shows good (MULTIPLE_FRAME_ESTABLISHED)

show controller t1 0/1/0 br
T1 0/1/0 is up.
Applique type is Channelized T1
Cablelength is long gain36 0db
Description: ISDN-PRI-1
No alarms detected.
alarm-trigger is not set
Soaking time: 3, Clearance time: 10
AIS State:Clear LOS State:Clear LOF State:Clear
Framing is ESF, Line Code is B8ZS, Clock Source is Line Primary.
Data in current interval (6 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs

show isdn st
Global ISDN Switchtype = primary-ni
ISDN Serial0/1/0:23 interface
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 = 7
Total Allocated ISDN CCBs = 0

Also, when I look at the web page for the phone that made the call (during a live call), it's not showing any received packets.  For some reason, it's just not recieving anything from the outside call. This is the past that has me stumped.  I'm not sure why it's not getting any audio from the outside call on the ISDN PRI / T1 module... 

Sender packets 204
Sender octets 32640
Sender codec G.711u
Sender reports sent 1
Sender report time sent 10:56:50am
Rcvr lost packets 0
Avg jitter 0
Receiver codec G.711u
Receiver reports sent 0
Receiver report time sent 00:00:00
Rcvr packets 0
Rcvr octets 0

Here's my config.  Some of the IP's changed so I don't post our internal IPs in public).  The T1 port I'm using is in slot 0/1/0

card type t1 0 1
network-clock synchronization automatic
network-clock input-source 1 controller t1 0/1/0
isdn switch-type primary-ni
trunk group PSTN-TRUNK-GRP
 hunt-scheme sequential both up
controller T1 0/1/0
clock source line primary
 description ISDN-PRI-1
 framing ESF
 linecode B8ZS
pri-group timeslots 1-24
voice-card 0/1
dsp services dspfarm
no local-bypass
voice-card 0/4
dsp services dspfarm
no local-bypass
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol t38 nse force fallback pass-through g711ulaw
 sip
 bind media source-interface gi0/0/1.2
 bind control source-interface gi0/0/1.2
 session transport tcp
 registrar server expires max 600 min 60
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
codec preference 4 g729br8
sccp local GigabitEthernet0/0/1.2
sccp ccm 172.16.1.100 identifier 1 priority 1 version 7.0+
sccp ccm 172.16.1.101 identifier 2 priority 2 version 7.0+
sccp ccm 172.16.1.102 identifier 3 priority 3 version 7.0+
sccp
dspfarm profile 1 mtp
 description MTP Resources
 codec g711ulaw
 maximum sessions hardware 8
 associate application SCCP
 no shutdown
dspfarm profile 2 conference
 description Conferencing Resources
 codec g711ulaw
 codec g711alaw
 codec g729r8
 codec g729br8
 codec g729ar8
 codec g729abr8
 maximum sessions 6
 associate application SCCP
 no shut
dspfarm profile 3 transcode
 description Transcoding Resources g711 / g729
 codec g711ulaw
 codec g711alaw
 codec g729r8
 codec g729br8
 codec g729ar8
 codec g729abr8
 maximum sessions 8
 associate application SCCP
 no shut
sccp ccm group 1
description US CUCM CLUSTER
associate ccm 1 priority 1
associate ccm 2 priority 2
associate ccm 3 priority 3
associate profile 1 register MTP-US-SCE
associate profile 2 register CFB-US-SCE
associate profile 3 register XCODE-US-SCE
sip-ua
 no remote-party-id
 retry invite 2
 timers buffer-invite 5000
interface Serial0/1/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn map address ^011 plan unknown type unknown
 no cdp enable
no shut
voice translation-rule 1
 rule 1 /^\(..........\)$/ /+1\1/
voice translation-rule 2
rule 1 /^\+1/ //
voice translation-rule 3
rule 1 // // type any unknown plan any unknown
voice translation-profile TP-Incoming
 translate called 1
voice translation-profile TP-Outgoing
 translate calling 2
 translate called 3
voice-port 0/1/0:23
 translation-profile incoming TP-Incoming
 translation-profile outgoing TP-Outgoing
controller t1 0/1/0
 trunk-group PSTN-TRUNK-GRP timeslots 1-24 preference 1
voice class uri CUCM sip
 host ipv4:172.16.1.100
 host ipv4:172.16.1.101
 host ipv4:172.16.1.102
dial-peer voice 1 voip
description INCOMING-TO-CUCM-SUB1
preference 1
destination-pattern +1..........
session protocol sipv2 
session target ipv4:172.16.1.100
session transport tcp
dtmf-relay rtp-nte
no vad
voice-class codec 1
dial-peer voice 2 voip
description INCOMING-TO-CUCM-SUB2
preference 2
destination-pattern +1..........
session protocol sipv2 
session target ipv4:172.16.1.101 
session transport tcp
dtmf-relay rtp-nte
no vad
voice-class codec 1
dial-peer voice 3 voip
description INCOMING-TO-CUCM-PUB
preference 3
destination-pattern +1..........
session protocol sipv2
session target ipv4:172.16.1.102
session transport tcp
dtmf-relay rtp-nte
no vad
voice-class codec 1
dial-peer voice 10 pots
 description LOCAL 7 DIGIT CALLS
 trunkgroup PSTN-TRUNK-GRP
 destination-pattern 9[2-9]......
dial-peer voice 11 pots
 description LOCAL 10 DIGIT CALLS
 trunkgroup PSTN-TRUNK-GRP
 destination-pattern 9[2-9]..[2-9]......
dial-peer voice 12 pots
 description LONG DISTANCE CALLS
 trunkgroup PSTN-TRUNK-GRP
 destination-pattern 91[2-9]..[2-9]......
 forward-digits 11
dial-peer voice 13 pots
 description INTERNATIONAL CALLS
 trunkgroup PSTN-TRUNK-GRP
destination-pattern 9011T
 prefix 011
dial-peer voice 14 pots
 description 911 CALLS
 trunkgroup PSTN-TRUNK-GRP
destination-pattern 911
 forward-digits 3
dial-peer voice 15 pots
 description 911 CALLS WITH EXTRA 9
 trunkgroup PSTN-TRUNK-GRP
 destination-pattern 9911
 forward-digits 3
dial-peer voice 50 voip
incoming called-number .
dtmf-relay rtp-nte
voice-class codec 1
no vad
dial-peer voice 51 pots
Incoming called-number .
direct-inward-dial
voice register global
system message SRST Mode
max-dn 50
max-pool 1
dialplan-pattern 1 +1.......... extension-length 4 demote

FYI - UPDATE:

This was the resolution:

Problem solved.  Bug....added this to the NIM module...

https://bst.cloudapps.cisco.com/bugsearch/bug/CSCuu86175/?referring_site=bugquickviewredir

This only applies if we’re using zone based firewall, which we are.  'brown' is specific to our zone based firewall config, just so you know.  That's not a general entry anyone else would use.

 

interface Service-Engine0/1/0

zone-member security brown

5 Replies 5

Aseem Anand
Cisco Employee
Cisco Employee

Hi,

Can you configure the following:

1. voice rtp send-recv is added on the VG to enable voice cut through.

2. Run the command show controller e1 0/0 and verify if there are slip errors or not.

3. run the debug isdn q931 for an outgoing call and as well as for an incoming call and check the channel used. Sometimes the issue can be with a particular channel.

4. Check if the clocking is configured correctly. You can try changing it to line or internal.

5. try changing the framing and see if it gets you a better result.

Aseem

1.   I added the voice rtp send-recv statement but it had no affect.

2. I cleared the counters and watched it and I see no slips or any other errors when I look at the show controller command.  

3. It uses channel 1 first for outgoing calls and channel 23 first for incoming,  Same problem on both channels.

4. Clocking is configured using the examples I found on the Cisco page and it appears to be sync'd correctly.  Output shown below.

5. I haven't tried changing the framing but it must be ESF since it's ISDN PRI and I had the telco confirm that.

show network-clocks sync
Symbols: En - Enable, Dis - Disable, Adis - Admin Disable
NA - Not Applicable
* - Synchronization source selected
# - Synchronization source force selected
& - Synchronization source manually switched

Automatic selection process : Enable
Equipment Clock : 2048 (EEC-Option1)
Clock Mode : QL-Disable
ESMC : Disabled
SSM Option : 1
T0 : T1 0/1/0
Hold-off (global) : 300 ms
Wait-to-restore (global) : 300 sec
Tsm Delay : 180 ms
Revertive : No

Nominated Interfaces

Interface SigType Mode/QL Prio QL_IN ESMC Tx ESMC Rx
Internal NA NA/Dis 251 QL-SEC NA NA
*T1 0/1/0 NA NA/Dis 1 QL-SEC NA NA

Let me ask this just to be sure we even have the correct module for what we're doing.

Presently, we have the NIM-2MFT-T1/E1 module in our 4351 router.  

...But I noticed there's a different T1 module on Cisco's web site that actually has ISDN in its name:  NIM-2CE1T1-PRI

Which module is correct for ISDN PRI T1 circuits in the USA???

 NIM-2MFT-T1/E1 is supported on the 4000 series router.  You can refer to the link below:

http://www.cisco.com/c/en/us/products/routers/4000-series-integrated-services-routers-isr/relevant-interfaces-and-modules.html

Aseem

This was the resolution:

Problem solved.  Bug....added this to the NIM module...

https://bst.cloudapps.cisco.com/bugsearch/bug/CSCuu86175/?referring_site=bugquickviewredir

 

This only applies if we’re using zone based firewall, which we are.  'brown' is specific to our zone based firewall config, just so you know.  That's not a general entry anyone else would use.

 

interface Service-Engine0/1/0

zone-member security brown

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