Cisco 232D CID problems

[alonazar28]

I have 2 different models of Cisco routers configured to work with Asterisk/FreePBX. One is the SPA3102 Voice Gateway and the other is the SPA232D Multi-Line DECT ATA. The SPA3102 seems to work well with both VoIP and PSTN lines. However, the 232D behaves un-reliably under PSTN connections.

When I setup the 232D behind a VoIP line, everything seems to work fairly well (except for caller ID not showing when called via PSTN lines). However, when the 232D is setup behind a POTS line, things start to go wrong. I seem to have the same issues as seen here: https://supportforums.cisco.com/discussion/11946961/spa232d-problems-cid-attached-handset

I can get the caller ID to show up only if I configure the "PSTN Answer Delay" (5) and "PSTN Ring Thru Delay" (1), which is not ideal when working with a PBX. I tried several different caller ID methods and FSK’s that are available in the configuration to no avail. Log files give no indication of any issues that I can see.

In addition, I also cannot get the PSTN line to connect an outgoing call on 50% of the POTS lines I tested on with the 232D. The phone simply reverts back to a dial tone. I tried several units since my company purchased many 232D routers. All the 232D units behave the same. 

Both models are configured the same, as far as I can tell, although my knowledge is extremely limited.

Doe anyone have any advice how to get the 232D to operate the same as the SPA3102 for both POTS and VoIP?

I have attached a log of an incoming call from a PSTN line and a 232D configuration file.

Tanks,

Alon

[Dan Lukes]

I can get the caller ID to show up only if I configure the "PSTN Answer Delay" (5) and "PSTN Ring Thru Delay" (1), which is not ideal when working with a PBX. I tried several different caller ID methods and FSK’s that are available in the configuration to no avail. 

You can't select what CID should be used during call setup. It's your service provider decision. You need to claim what CID method IS used.

According the configuration you disclosed it seems the CID information is transmitted between first and second ring.

Your log confirm my assumptions. The call start with polarity reversal (10:25:03 spa232d FXO: OnHook PolRev) followed by ring (10:25:03 Ring start) followed by CID (10:25:05 CID received on EP 0). Line go to off hook then (10:25:05 spa232d FXO: Off Hook). SIP leg of the call

Line needs to be in on hook state or the CID can't be transmitted. Thus you need to wait for second ring to seize line. You need to decide - either both CID and delay, or neither CID nor delay.

POTS lines has been designed to connect end devices not to interconnect systems. They can be used for it, but behavior is not optimal for such task. Consider other interconnection interfaces, like native SIP trunk or ISDN2 if you wish for interconnect with no compromises.

I also cannot get the PSTN line to connect an outgoing call on 50% of the POTS lines I tested on with the 232D. The phone simply reverts back to a dial tone.

There's just one way to enter dial tone state on POTS. Dial tone follow on hook to off hook transition only. The only even that can be mis-recognized as on-to-off hook transition is the 'flash' feature. Are you using it ? FXS parameters (Voice -> Regional -> Miscelaneous) needs to be configured according PSTN specification/requirements.

Syslog&debug may (but may not) disclose more.

[alonazar28]

Thank you for your response. I think I am still missing something...

The only even that can be mis-recognized as on-to-off hook transition is the 'flash' feature. Are you using it ?

I have ‘Line 1 Signal Hook Flash to PSTN’ set to Disabled, and ‘Off Hook While Calling VoIP’ set to no.

FXS parameters (Voice -> Regional -> Miscelaneous) needs to be configured according PSTN specification/requirements.

I have the two models configured with the same values. The SPA3102 works while the 232D does not.

Line needs to be in on hook state or the CID can't be transmitted. Thus you need to wait for second ring to seize line. You need to decide - either both CID and delay, or neither CID nor delay.

Again, if both models are configured the same and one works, I would expect that the configuration is correct.

I included log files for each model for comparison. I do not have the experience to determine the difference between the models using the logs. Are you able to see any difference between how each model behaves in terms of the issue at hand?

Thank you.

[Dan Lukes]

I have the two models configured with the same values. The SPA3102 works while the 232D does not. 

It may mean either

1. both devices have wrong configuration, but because of different tolerance and sensitivity the one hardware works while second does not. In such case you need to configure (at least) SPA232D correctly (but same configuration should work even for SPA3102 as well)

or

2. there's a bug in the hardware/firmware of SPA232D. In such case only Cisco technicians may help - call SMB Support Center.

Again, if both models are configured the same and one works

You claimed even SPA232D can receive CID - and the log provided is confirming it. Thus I'm unsure what you mean saying "one works". Is it the SPA3102 that doesn't work for you ?

Are you able to see any difference between how each model behaves in terms of the issue at hand?

I'm still unsure about the issue related to incoming call.

It take 2 seconds from "Start CNDD" to "FXO: Off Hook" on SPA232D as well s on SPA3102. It take another 3 seconds to receive "SIP/2.0 180 Ringing" from line 1000 SIP peer. Still same for SPA232D as well SPA3102. Another 11 seconds and we hit "SIP/2.0 200 OK" from line 1000 SIP peer - the call matured to connected state. Again, no difference between SPA232D and SPA3102.

So the behavior as well as timing seems to be the same (within 1s precision) on both models. Despite you has claimed SPA232D behavior "not ideal when working with a PBX" I'm unable to identify a true difference.

May be you should be more specific about the issue you are facing with incoming call.

I will make other comment with analysis of outgoing call logs later.

[alonazar28]

To me both routers also seem exactly the same. The issue is that NO caller ID shows up on the handset when using the 232D but there is a caller ID showing up on the handset when using the SPA3102. I realize the CID comes through the SIP on both (as the logs show) but only using the SPA3102, does the CID display on the handset. 

[Dan Lukes]

OK, now it's clear.

Unfortunately, this part of session can't be compared. SPA3102 doesn't log so much about FXS operations. So we have log from SPA232D only:

[0]Ring cad event 1 pol 0
CID: OnHookTx Pol
[0]CID CID_ST_POLREV_POST_DELAY
uchDisplayCIDFSK(), EP 1 lid 0 buflen 99 overhead 60 SZ_MAX_USERDATA 200 offhook 0
uchDisplayCIDFSK(), FSK Caller ID standard is 0(bell 202)
uchDisplayCIDFSK(), SeizeFreq 0x16 MarkFreq 0xc
[0]CID Start DTMF/FSK, CID_ST_ACTIVE
uchAppCb(), Event 65 received EP 1 lid 0
CH_ASYNC_CIT_TRANSMITTED 
[0]CID CID:DONE
[0]CID CID_ST_ACTIVE_POST_DELAY
[0]CID CID_ST_IDLE
[0]Ring cad event 0 pol 0

In short, CID has been sent between first and second ring, FSK/Bell protocol used. The FXS line parameters are configurable in Voice -> Miscellaneous -> Miscelaneous and  Voice -> Line 1 -> FXS Port Polarity Configuration. It need to be configured according your analog phone requirements. Wrong impedance and gain may cause the CID signal to be so distorted to be recognized. Phone may have specific requirements related to polarity reversal policy. Of course, Caller ID Method and FSK Standard needs to be set according particular phone's requirements as well.

I have no SPA3102 thus I don't know if it can be configured the same way as SPA232D or some options are hardwired/not configurable on SPA3102.

For configurable items you should use the same. For those not configurable on SPA3102, you should just guess & try.

You can call SMB support center and ask for help, of course.

[Dan Lukes]

[posted by accident, but not allowed to delete]

[Dan Lukes]

According outgoing call - are you using speaker mode (e.g. "loud" mode) of the phone during dialing ? If yes ...

... Can you try to dial with handset (e.g. no loud voice) ?

According the LOG, it seems you has dialed #9250

The phone has fired "fast busy tone" after the '2' digit has been received. The FXO has not been active yet, so PSTN dialing seems not to be part of the issue.

Is the number in question allowed by dial plan ? Have you #92 configured as a Vertical Service Activation Code or an other code ?