Setting up CUCME PSTN connection via POTS

Chris Campbell · ‎04-03-2020

Hello all,

I am trying to learn CUCME. I have some test equipment consisting of two 7821 SIP phones, a 2951-VSEC router, a VIC2-4FXO card, and a PVDM32 DSP module. I have gone through some tutorials and have the ability to call from one 7821 to the other. I am now trying to figure out how to make a connection to the PSTN.

To accomplish my PSTN connection, I called up the phone company and asked to add home phone (POTS) to my service. Previously I only had DSL. The POTS service is supposed to be active.

When I try a test outbound call using the command

csim start 2945237

The FXO card port 0 light comes on for less than a second, and then turns off. Trying to call from one of the 7821 phones gets me a fast busy signal.

I do have a POTS phone somewhere in a box, but I'm not sure where it is at this time. Obviously I'd like to verify the POTS connection is working, but I have to find my analog phone before I can do that. What else can I do from the 2951 router to test to make sure the line is active?

Here's what I have:

dial-peer voice 1 pots

description local

destination-pattern 2......

no digit-strip

port 0/1/0

forward-digits 7

!

voice-port 0/1/0

connection plar opx 4001

______________

#show voice port summary

IN OUT

PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC

=============== == ============ ===== ==== ======== ======== ==

0/1/0 -- fxo-ls up dorm idle on-hook y

0/1/1 -- fxo-ls up dorm idle on-hook y

0/1/2 -- fxo-ls up dorm idle on-hook y

0/1/3 -- fxo-ls up dorm idle on-hook y

PWR FAILOVER PORT PSTN FAILOVER PORT

================= ==================

#csim start 2945237

csim: Test started.This command can only be run one at a time.

csim: called number = 2945237, loop count = 1 ping count = 0

csim err:csim_do_test invalid major major(1024) minor(0x0)

csim: loop = 1, failed = 0

csim: call attempted = 1, setup failed = 0, tone failed = 0

#

*Apr 3 17:30:35.718: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Calling Number=, Called Number=2945237, Peer Info Type=DIALPEER_INFO_SPEECH

*Apr 3 17:30:35.718: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=2945237

*Apr 3 17:30:35.718: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

Dial String=2945237, Expanded String=2945237, Calling Number=

Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH

*Apr 3 17:30:35.722: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:

Result=Success(0); Outgoing Dial-peer=1 Is Matched (1 digits)

*Apr 3 17:30:35.722: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Result=Success(0) after DP_MATCH_DEST

*Apr 3 17:30:35.722: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=0

*Apr 3 17:30:35.722: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

Result=SUCCESS(0)

List of Matched Outgoing Dial-peer(s):

1: Dial-peer Tag=1

#show voice port 0/1/0

Foreign Exchange Office 0/1/0 Slot is 0, Sub-unit is 1, Port is 0

Type of VoicePort is FXO

Operation State is DORMANT

Administrative State is UP

The Last Interface Down Failure Cause is Administrative Shutdown

Description is not set

Noise Regeneration is enabled

Non Linear Processing is enabled

Non Linear Mute is disabled

Non Linear Threshold is -21 dB

Music On Hold Threshold is Set to -38 dBm

In Gain is Set to 0 dB

Out Attenuation is Set to 3 dB

Echo Cancellation is enabled

Echo Cancellation NLP mute is disabled

Echo Cancellation NLP threshold is -21 dB

Echo Cancel Coverage is set to 128 ms

Echo Cancel worst case ERL is set to 6 dB

Playout-delay Mode is set to adaptive

Playout-delay Nominal is set to 60 ms

Playout-delay Maximum is set to 1000 ms

Playout-delay Minimum mode is set to default, value 40 ms

Playout-delay Fax is set to 300 ms

Connection Mode is plar opx

Connection Number is 4001

Initial Time Out is set to 15 s

Interdigit Time Out is set to 10 s

Call Disconnect Time Out is set to 60 s

Power Denial Disconnect Time Out is set to 1000 ms

Ringing Time Out is set to 180 s

Wait Release Time Out is set to 30 s

Companding Type is u-law

Region Tone is set for US

Analog Info Follows:

Currently processing none

Maintenance Mode Set to None (not in mtc mode)

Number of signaling protocol errors are 0

Impedance is set to 600r Ohm

Station name None, Station number None

Translation profile (Incoming):

Translation profile (Outgoing):

lpcor (Incoming):

lpcor (Outgoing):

Voice card specific Info Follows:

Signal Type is loopStart

Battery-Reversal is enabled

Number Of Rings is set to 1

Supervisory Disconnect is signal

Answer Supervision is inactive

Hook Status is On Hook

Ring Detect Status is inactive

Ring Ground Status is inactive

Tip Ground Status is inactive

Dial Out Type is dtmf

Digit Duration Timing is set to 100 ms

InterDigit Duration Timing is set to 100 ms

Pulse Rate Timing is set to 10 pulses/second

InterDigit Pulse Duration Timing is set to 750 ms

Percent Break of Pulse is 60 percent

GuardOut timer is 2000 ms

Minimum ring duration timer is 125 ms

Hookflash-in Timing is set to 600 ms

Hookflash-out Timing is set to 400 ms

Supervisory Disconnect Timing (loopStart only) is set to 350 ms

OPX Ring Wait Timing is set to 6000 ms

Secondary dialtone is disabled

Chris Campbell · ‎04-04-2020

Here's an update. I got a call from the phone company tech and he said sales screwed up my order, so the POTS service isn't connected. I also found my analog phone and verified that there's no service. So that explains it. Hopefully they will straighten out some of this next week.

TONY SMITH · ‎04-06-2020

Just a suggestion because I find analogue PSTN lines a pain nowadays. See if you can find a SIP service to sign up to, and get your 2921 to register with that for your external calls. Be careful with security, but if you use one of the pre-paid services even the worst sort of dial through fraud can only consume the credit you've already paid for.

By the way don't think that toll fraud is limited to IP services. I knew a company where they had analogue lines connected to a router, which was also an Internet gateway. A fraudster managed to access the router and ran up huge bills making calls on the analogue lines.

Chris Campbell · ‎04-10-2020

I really appreciate the warning about toll fraud. Right now, this router is behind my another router which is configured as a firewall, and I do a good job of security, so I think I'm okay, but it is something to watch for.

I got the phone company to straighten out my order and after that, everything is working fine.

In the meantime, I am trying to figure out why the system doesn't complete a call when I pick up the receiver and dial the number, but if I punch it in with the buttons and press the call button, it works fine. That applies to calls inside my house, and calls outgoing to the PSTN. Any ideas?

TONY SMITH · ‎04-11-2020

That could be an overlap in your dial plan. When you say it doesn't work dialling off hook, do you mean it comes up with an error before you've finished dialling? If so then note exactly what digits you've dialled, and look for some match for those digits.

Chris Campbell · ‎04-12-2020

It doesn't give me any errors at all... it just hangs there with dead air and doesn't complete the call.

Chris Campbell · ‎04-14-2020

I think I'm onto something after printing out some debugging information. It's saying the called number is 4 and more digits are needed. Yes, that's true. My test setup used 4-digit extensions in the house. I'm wondering why it's not seeing the rest of the digits. I may try setting up extensions that are just one digit and see how that works. But I'm still not understanding why it's cutting off after a single digit.

*Apr 14 14:36:18.560: //1128/2316FB8E85B0/CCAPI/cc_api_call_setup_ind_common:

Set Up Event Sent;

Call Info(Calling Number=4001(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),

Called Number=4(TON=Unknown, NPI=Unknown))

*Apr 14 14:36:18.560: //1128/2316FB8E85B0/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP

*Apr 14 14:36:18.560: //1128/2316FB8E85B0/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:100, container:1C12228C

*Apr 14 14:36:18.560: //1128/2316FB8E85B0/CCAPI/cc_process_call_setup_ind:

Event=0x164234E8

*Apr 14 14:36:18.560: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:

Try with the demoted called number 4

*Apr 14 14:36:18.560: //1128/2316FB8E85B0/CCAPI/ccCallSetContext:

Context=0x177110C4

*Apr 14 14:36:18.560: //1128/2316FB8E85B0/CCAPI/cc_process_call_setup_ind:

>>>>CCAPI handed cid 1128 with tag 40001 to app "_ManagedAppProcess_Default"

*Apr 14 14:36:18.560: //1128/2316FB8E85B0/CCAPI/ccCallSetupAck:

Call Id=1128

*Apr 14 14:36:18.560: //1128/2316FB8E85B0/CCAPI/cc_api_set_transfer_info:

Transfer Number=, Transfer Reason=0x0

*Apr 14 14:36:18.560: //-1/2316FB8E85B0/DPM/dpMatchPeersCore:

Calling Number=, Called Number=4, Peer Info Type=DIALPEER_INFO_SPEECH

*Apr 14 14:36:18.560: //-1/2316FB8E85B0/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=4

*Apr 14 14:36:18.560: //-1/2316FB8E85B0/DPM/dpMatchCore:

Dial String=4, Expanded String=4, Calling Number=

Timeout=FALSE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH

*Apr 14 14:36:18.560: //-1/2316FB8E85B0/DPM/MatchNextPeer:

Result=MORE_DIGITS_NEEDED(1); Outgoing Dial-peer=40001

*Apr 14 14:36:18.560: //-1/2316FB8E85B0/DPM/dpMatchCore:

Result=1

*Apr 14 14:36:18.560: //-1/2316FB8E85B0/DPM/dpMatchPeersCore:

Result=Partial Matches(1) after DP_MATCH_DEST

*Apr 14 14:36:18.560: //-1/2316FB8E85B0/DPM/dpMatchSafModulePlugin:

dialstring=4, saf_enabled=1, saf_dndb_lookup=0, dp_result=1

*Apr 14 14:36:18.560: //-1/2316FB8E85B0/DPM/dpMatchPeersMoreArg:

Result=MORE_DIGITS_NEEDED(1)

*Apr 14 14:36:18.560: //1128/2316FB8E85B0/CCAPI/ccSetDigitTimeouts:

Initial Digit Timeout=10000(ms), Inter Digit Timeout=10000(ms)

*Apr 14 14:36:18.560: //1128/2316FB8E85B0/CCAPI/ccSetDigitTimeouts:

Call Entry(Inter Digit Timeout=10000(ms), Initial Digit Timeout=10000(ms))

*Apr 14 14:36:18.560: //1128/2316FB8E85B0/CCAPI/ccCallReportDigits:

(callID=0x468, digit_event=0x1, enable=TRUE, consume=FALSE)

*Apr 14 14:36:18.560: //1128/2316FB8E85B0/CCAPI/ccCallReportDigits:

Enabled=TRUE, Call Id=1128

*Apr 14 14:36:18.560: //-1/xxxxxxxxxxxx/SIP/Event/ccsip_spi_subscribe_client: Queued event from SIP SPI : SIPSPI_EV_CC_S

Chris Campbell · ‎04-14-2020

I changed the dn of the other phone to 4 and it works like I expect. Interesting!