Pri Trunking issues

[normrr]

I am having issues with occasional calls that ring the customer station with no audio, with in a second or 2 the same call rings again and audio both ways is fine. The calling party never heard the first answer just a long period of ringback, I also see this in my soft-switch as 20-30 seconds of ringback. I have a SIP Trunk fed 904 2nd Gen onsite that interfaces with a Tadiran Coral IPx 500. This the only site I have seen this at out of the over 60 PRI's I deliver all on 904 platforms. In the logs I see this entry often  2016.02.16 12:09:55 TM.T02 01 IsdnTmStateIdling - clear trunk appearance. Unfortunately I cleared the logs but I have also seen this listed as Unexpected Cleared Trunk Appearance. Has anyone seen this sort of issue before?

[normrr]

wanted to add the new logs

2016.02.16 13:48:08 TM.T02 01 IsdnTmStateOutboundDeliver - rcvd unexpected CallRelease

2016.02.16 13:48:08 TM.T02 01 IsdnTmStateIdling - clear trunk appearance

2016.02.16 13:48:08 TM.T02 01 IsdnTmStateIdle::CallRelease - unexpected

2016.02.16 13:48:08 TM.T02 01 IsdnTmStateIdling - clear trunk appearance

2016.02.16 14:04:43 TM.T02 02 IsdnTmStateIdling - clear trunk appearance

[Anonymous]

Norman, please reply with the following:

configuration of TA

Output of "debug isdn l2-f", "debug voice verbose", and "debug sip stack message" running during one of these calls. If the issue is intermittent, you can use this document to start persistent debug logs:

Enabling Persistent Debug Logging

Thanks

Jay

[normrr]

Thanks jay, It is a very intermittent issue but I just don't see this on the other PRI's. I will need to set up the debug logging. Here is the config file

!

!

! ADTRAN, Inc. OS version R10.10.0.E

! Boot ROM version 14.04.00

! Platform: Total Access 904 (2nd Gen), part number 4212904L1

! Serial number CFG1389694

!

!

hostname "display"

enable password !NMnet09

!

clock timezone -8

!

ip subnet-zero

ip classless

ip routing

!

!

name-server xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx

!

!

no auto-config

!

event-history on

no logging forwarding

no logging email

!

no service password-encryption

!

username "XXXXXX" password "XXXXXXX"

!

banner motd #

        Important 

Web username/password is configured to admin/password.

Enable and Telnet passwords are configured to "password".

Please change them immediately.

The ethernet 0/1 interface is enabled with an address of 10.10.10.1

Telnet/SSH access is also enabled.

#

!

!

no ip firewall alg msn

no ip firewall alg mszone

no ip firewall alg h323

!

!

!

!

!

no dot11ap access-point-control

!

!

!

!

!

!

!

!

!

!

!

!

!

!

interface loop 1

  no ip address

  no shutdown

!

interface eth 0/1

  ip address  xxx.xxx.xxx.xxx  xxx.xxx.xxx.xxx

  media-gateway ip primary

  no awcp

  no shutdown

!

!

!

!

interface t1 0/1

  shutdown

!

interface t1 0/2

  description XXXXXXX

  tdm-group 1 timeslots 1-24 speed 64

  no shutdown

!

!

interface pri 1

  description pri 1

  role network b-channel-restarts enable

  isdn name-delivery setup

  connect t1 0/2 tdm-group 1

  no shutdown

!

!

interface fxs 0/1

  impedance 600r

  rx-gain +0.0

  tx-gain +0.0

  no shutdown

!

interface fxs 0/2

  impedance 600r

  rx-gain +0.0

  tx-gain +0.0

  no shutdown

!

interface fxs 0/3

  impedance 600r

  rx-gain +0.0

  tx-gain +0.0

  no shutdown

!

interface fxs 0/4

  impedance 600r

  rx-gain +0.0

  tx-gain +0.0

  no shutdown

!

!

isdn-group 1

  min-channels 1

  max-channels 23

  connect pri 1

!

isdn-number-template 0 prefix "" plan 1 type 2 NXX-NXX-XXXX

isdn-number-template 1 prefix "" plan 1 type 2 NXX-XXXX

isdn-number-template 2 prefix "" plan 1 type 2 1-NXX-NXX-XXXX

!

!

!

timing-source internal

!

timing-source internal secondary

!

!

!

!

!

!

ip route 0.0.0.0 0.0.0.0 xxx.xxx.xxx.xxx

!

ssh-server pubkey-chain

!

no tftp server

no tftp server overwrite

http server

no http secure-server

no snmp agent

no ip ftp server

no ip scp server

no ip sntp server

!

!

!

!

!

!

!

!

sip

sip udp 5060

no sip tcp

!

!

!

voice feature-mode network

voice forward-mode network

!

!

!

!

!

!

!

!

voice dial-plan 1 local NXX-NXX-XXXX

voice dial-plan 2 long-distance 1-NXX-NXX-XXXX

!

!

voice email-list emergency

  description “Display"

  address!

!

!

voice codec-list "SIP to PRI"

  default

  codec g711ulaw

  codec g711alaw

  codec g722

  codec g729

!

voice codec-list "sip to analog"

  codec g711ulaw

  codec g711alaw

  codec g722

!

!

!

voice trunk T01 type sip

  description "SIP to XXXXXX"

  sip-server primary voip.XXXXXXXXXX.com

  registrar primary XXX.XXX.XXX.XXX

  no registrar require-expires

  outbound-proxy primary voip.XXXXXXXXXX.com

  domain "Display"

  dial-string source to

  max-number-calls 23

  codec-list "SIP to PRI" both

  authentication username "XXXXXXXXX" password "XXXXXXXXX"

  no diversion-supported

  transfer-mode local

!

voice trunk T02 type isdn

  description "PRI to PBX"

  resource-selection linear ascending

  caller-id-override number-inbound XXXXXXXXXXX

  connect isdn-group 1

  early-cut-through

  modem-passthrough

  rtp delay-mode adaptive

!

!

voice grouped-trunk "XXXXXXXX SIP TO PRI"

  trunk T01

  accept NXX-XXXX cost 0

  accept 1-NXX-NXX-XXXX cost 0

  accept 1-800-NXX-XXXX cost 0

  accept 1-888-NXX-XXXX cost 0

  accept 1-877-NXX-XXXX cost 0

  accept 1-866-NXX-XXXX cost 0

  accept 1-855-NXX-XXXX cost 0

  accept 011-$ cost 0

  accept 411 cost 0

  accept 611 cost 0

  accept 911 cost 0

  accept 10-10-XXX-$ cost 0

  accept NXX-NXX-XXXX cost 0

  reject 976-XXXX

  reject 1-900-NXX-XXXX

  reject 1-976-NXX-XXXX

  reject NXX-976-XXXX

!

!

voice grouped-trunk "PRI TO PBX"

  trunk T02

  accept $ cost 0

  accept NXX-NXX-XXXX cost 0

  accept 1-NXX-NXX-XXXX cost 0

  accept 1-800-NXX-XXXX cost 0

  accept 1-888-NXX-XXXX cost 0

  accept 1-877-NXX-XXXX cost 0

  accept 1-866-NXX-XXXX cost 0

  accept 1-855-NXX-XXXX cost 0

  accept 011-$ cost 0

  accept 411 cost 0

  accept 611 cost 0

  accept 911 cost 0

  accept 0-NXX-NXX-XXXX cost 0

  accept 10-10-XXX-$ cost 0

  reject NXX-976-XXXX

  reject 1-900-NXX-XXXX

  reject 1-976-NXX-XXXX

!

!

voice user 300

  connect fxs 0/1

  first-name "XXXXXX"

  last-name "XXXX"

  password "1234"

  caller-id-override external-numberXXXXXXXXX

  modem-passthrough

  no echo-cancellation

  codec-list "SIP to PRI"

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

sip qos dscp 46

!

!

!

ip rtp symmetric-filter

!

!

!

line con 0

  no login

!

line telnet 0 4

  login

  password !NMnet09

  shutdown

line ssh 0 4

  login local-userlist

  shutdown

!

!

!

!

!

end

[normrr]

I am still having issue with about 10 % of calls failing or timing out (408) and the customer tells me that sometimes they get an unassigned when they try to make an out bound call. I know this is trunk related. I can seem to find the messaging error.

[Anonymous]

Norman, in the example you provided, the 900 sends 2 setup messages with no response:

06:31:22.844 ISDN.L2_FMT PRI 1 ==============================================

06:31:22.844 ISDN.L2_FMT PRI 1  T Sapi:00 C/R:C Tei:00 INFO Ns:65  Nr:81  P:0

06:31:22.844 ISDN.L2_FMT PRI 1    Prot:08  CRL:2 CRV:1307

06:31:22.845 ISDN.L2_FMT PRI 1    M - 05 SETUP      

06:31:27.015 ISDN.L2_FMT PRI 1  T Sapi:00 C/R:C Tei:00 INFO Ns:66  Nr:81  P:0

06:31:27.015 ISDN.L2_FMT PRI 1    Prot:08  CRL:2 CRV:1307

06:31:27.015 ISDN.L2_FMT PRI 1    M - 05 SETUP      

We start to tear down the call and we don't receive anything from the provider until about 20 seconds later:

06:31:43.442 ISDN.L2_FMT PRI 1  R Sapi:00 C/R:R Tei:00 INFO Ns:81  Nr:67  P:0

06:31:43.442 ISDN.L2_FMT PRI 1    Prot:08  CRL:2 CRV:9307

06:31:43.444 ISDN.L2_FMT PRI 1    M - 02 CALL_PROC  

It appears the provider is taking too long to process the call. Thanks

Jay

[normrr]

Again Jay, Thanks for all the help over the last couple of years. Yes I am getting a delay from the PBX, I have updated the firmware and see 408 coming back. The real question is why on only this PBX do I get intermittent delay and the customer get the occasional unassigned when making an outbound call. I have reconfigured every which way with no change and the Vendor has no tech support from Tadiran. I think its a messaging issue but cant pin it down.

[Anonymous]

Norman, unfortunately there isn't a lot more we can look at from the 900 perspective. We're sending a properly formatted message to the PBX and not receiving a timely response. Even if the PBX didn't like the format of the message, it should reply with some sort of error and not just wait. It even attempts to complete the call, it's just doing it far too late for it to matter. Thanks

Jay

[normrr]

Thanks Jay, I think I have found a resource that wont charge me $90/hr to talk about the Tadiran config.

[Anonymous]

normrr

What did you ever find out about the Tadiran config. We are have what sounds like the exact same problem with a customer and a Mitel system.

Thank you

Josh

[Anonymous]

A couple of things to double check.

1.  Check PBX to make sure it gets it's timing from the Adtran since Adtran is set to generate timing

2.  You may want to set the LBO on the T1 to match the cable length correctly, if the length is detected incorrectly it can cause timing issues

3.  Make sure the PBX is using the same setting as the Adtran is using on the T1 int i.e. Coding, Framing (run show run int t1 0/2 verb)

4.  Make sure the PBX is using the same setting as the Adtran is using on the PRI int i.e. isdn switch type (run show run int PRI 1 verb)

5. Try changing role network b-channel-restarts to disable that's the setting I use with most PBXs (Including Mitel System to other poster)

6.  Use show int t1 0/2 and look for errors like loss of frame, loss of signal etc.

7.  Use show int pri 1 and look for errors like CRC, Input, Frame, etc.

8.  Additional recommendation you really only want your Voice Grouped Trunk going to the PBX to accept number directly on the PBX it shouldn't accept any other numbers

9.  Also if the adtran is using resource ascending make sure the PBX is using resource descending otherwise you run a great risk of inbound and outbound colliding

10.  Make sure the PBX has all 23 channels enabled

11.  Also I see your SIP trunk is set to 23 Max Calls but if you are running a PRI plus Analog the analog lines could talk up talk path that the PBX is expecting to have available.

Hope some of this helps

John Wable