SPA3102 PSTN Line DTMF tones issue

[pero95305]

Hi
I have problems with DTMF tones through analogue line
I have my two old gateways, one is GSM/VOIP Setuata211g, and 2nd is SPA3102
I was using SPA3102 for many years to make calls from analogue phone through VOIP or landline providers without any problems, I just setup and forget.
Now my current landline provider is giving me unlimited calls to any landline or mobile number in EU. So, because I’m traveling a lot around country my idea was to hook up this 2 gateways peer-to-peer in local home network, and make free international calls from my mobile phone. Where I’m dialling GSM gateway number form my mobile phone, GSM gateway is transferring call to SPA, then SPA is picking up call and I’m dialling *2 or *3, (speed dial setup in dial plan) and calls going through landline.
But problem is with DTMF tones generated by SPA3102. Tones are not recognized by landline operator.
My configuration is
GSM / VOIP gateway is setup as peer-to-peer and all GSM incoming calls are transfer to SPA3102 to PSTN Line tab (192.168.1.35:6789)
Then, when I’m hearing tone generated by SPA I’m start dialling full phone number or speed dial number from dial plan (e.g *2). The SPA is recognising the tones generated by mobile phone or soft phone installed in my PC.( I used soft phone to call straight through IP to the SPA to eliminated any problems with GSM gateway).
When I’m pressing *2, SPA is recognising tones and is starting sending own DTMF tones on landline, but the tones are not recognized by landline operator. Always I’m hearing “The number is incorrect”
But when I’m using analogue phone connected to SPA with this same dial plan coped to “Line 1” Tab everything works correctly, I didn’t have any problem with dialling. All calls are going through.
Only one number is recognizing every time without any problem, this number has digits 0 1 4 6 7 8 9, phone numbers with 2 3 and 5 are not recognized.
I tried different settings in ‘Regional’ Tab , ‘PSTN Line’ tab etc. I do factory resets few times and I start from scratch, I spend weeks to try find answer in internet without any results.
Please for help or any advice.

My SPA3102 Hardware 145(a) Software 5.2.13(GW002)

[Howard Wittenberg]

There are many things that could cause a problem sending DTMF. It would help if we could see your SPA3102 configuration and a sip debug trace of a test call that fails. There are a number of settings that impact the transmission of DTMF tones. The sip trace would show the digits received over the voip line.

You say using an analog phone connected to the SPA can send the number to the attached PSTN line successfully, where receiving it over voip causes a failure. This would seem to indicate that there is a problem with the transmission over the voip network.

Transmitting dtmf tones over voip generally works best if they are sent by the rfc2833 protocol which Cisco calls AVT. This would be a setting on the sending unit. Other protocols are Inband and INFO and should also be tried.

I believe Speed Dials on a SPA3102 are triggered by sending a single digit, not a single digit prefaced with an *. If a call is processed by the Line 1 dial plan it uses Speed Dials setup on the User 1 Tab, if the call is processed by the PSTN Line dial plan it uses Speed Dials setup on the PSTN User Tab.

Edit:  You may not be using the Speed Dial capability of the SPA3102 but rather building the speed dial into the dial plan where you setup (<*2:16317918378>|<*3:18005069511>| etc.  You still need to establish for certain that you are receiving the dialing instructions over voip correctly before you concentrate on any settings that may affect the voip-to-pstn gateway analog dtmf dialing on the pstn line especially since you say it works perfectly from an analog phone attached to the SPA3102.  They both go to the same place, however there is a capability to have a different dial plan on the PSTN Line Tab for a Line 1 caller and a voip caller. In addition, the * has a special meaning when physically dialing an ip address so you don't see it often referenced in a dial plan.

To run a syslog on a failing call you download and install a syslog program on your pc and make settings on your SPA3102. You can download a simple syslog program here:
https://supportforums.cisco.com/document/36921/using-slogsrvexe-utility

To capture the trace you setup your pc's local network ip address on the SPA3102 System Tab under Debug Server, set the Debug level to 3 and on the PSTN Line Tab you set the Sip Debug Option to FULL.

[pero95305]

Hi

Thanks for quick replay. How I told before I’m traveling a lot so I didn’t have a chance to do this through the week.  I done factory reset and, start from scratch.

I checked/tested almost all settings in PSTN Line tab, include:

Audio Configuration:    DTMF TX Mode, DTMF TX Method, DTMF TX Strict Hold Off Time

Dial Plan:     (xx.|<*2:0123456789><:@gw0>|<*3:0987654321><:@gw0>)

International Control:     DTMF Playback Level, DTMF Playback Twist

In Regional tab I do adjustments in Call Progress Tones.

All settings and suggestions I found on forums and checked them good few times.

It looks like SPA3102 is generates the tones which are not recognized by landline provider. Only I can hear that the number is incorrect.  Only one number is recognized every time, this is (*3 in PSTN dial plan) this number has digits 0 1 4 6 7 8 9.

To be honest I need only to use 5 numbers added to PSTN dial plan and dialled by choosing *2, *3, *4, *5, *6.

The syslog attached is just after factory reset, with basic settings in PSTN to make transfer calls from VOIP to PSTN.

[Dan Lukes]

As a blind shot - may be the generated DTMF tones are so short to be recognized reliably. Try

DTMF Tx Mode: Strict
DTMF Tx Strict Hold Off Time: 90

[Howard Wittenberg]

Pero95305,

The sip trace looks normal.

Dan Lukes suggestion about a test increasing the DTMF timing sounds good. I would change the setting on the PSTN Line Tab
PSTN Dial Digit Len to 0.15/0.15 (150ms on 150ms off). The default is 0.1/0.1.

Another DTMF setting to try increasing would be the delay before you start sending the DTMF digits after you take the FXO port off hook. That would be "PSTN Dialing Delay:". Change it to 2 seconds. The default is 1 second.

There is also the SPA to PSTN gain which is a volume setting. It's probably OK but you never know for sure.

Background:

You are calling from a PC softphone running MicroSip 3.14 using direct ip dialing over your local network from 192.168.1.127:51276 to 192.168.1.35:5061 using G711u codec and AVT for the DTMF transmission over the voip digital circuit.

In the trace you have 6 calls that connect, you dial several digits, the dial plan times out and then the SPA3102 takes the FXO port off hook and dials the digits The calls end normally. The SPA3102 trace capability doesn't actually show the digits dialed on the FXO port.

DTMF Digits received over voip line

Call 1: 0 1 1 2 2 3 3 3 4 4 5 5 5 6 6 6 7 7 7 8 8 9 9

Call 2: * 2 2

Call 3: 0 0 9 8 8 7 7 6 6 6 5 5 5 4 4 4 3 3 3 2 2 1 1

Call 4: * * 3 3

Call 5: * 2

Call 6: * 3

The last *2 call and *3 calls should match in dial plan you indicated.

[pero95305]

Howard Wittenberg

All changes without any success.

I'm calling from a PC softphone MicroSip 3.14 using direct ip just for the test, and for normal use I have GSM gateway Matrix SETU ATA211g but this same the call is over local network, using direct IP. In both cases is G711u codec and AVT for the DTMF transmission.
I tried to make call to two mobile phones. First I dial numbers manually, and after, this same numbers was dialed by using dial plan (*2 and*3)

I don't know what kind of digits are actually dialed on the FXO port.

The strange is when I'm calling through PSTN Line Tab using direct ip dialing over local network, from softphone MicroSip or from GSM gateway, and I'm using dial plan (xx.|<*2:0123456789><:@gw0>|<*3:0987654321><:@gw0>) located in PSTN Line Tab. I can't make call, all the time I hearing "we sorry your number is not completely dial, check number and try again".

But when I'm calling through Line 1 Tab using analogue phone and this same dial plan (xx.|<*2:0123456789><:@gw0>|<*3:0987654321><:@gw0>) located in Line 1 Tab. Everything working OK.

It looks like SPA3102 has issue to dial right DTMF tone on the FXO port but only from PSTN Line.

[Howard Wittenberg]

For what it is worth, the difference between dialing from the PSTN line Tab and the Line 1 Tab out the FXO port is the following:

Dialing from a phone attached to the SPA3102 the Line 1 Tab the Line 1 dial plan resolves the *2 or *3 substitution and sends a Sip Invite to the PSTN Line tab with the number to be dialed in the Sip Invite as "one stage dialing". The PSTN Line Tab receives the Sip Invite with the number to be dialed and passes it thru the dial plan (if any) under the VoIP-To-PSTN Gateway setup as the dial plan for "Line 1 VoIP Caller DP" which may or may not be the same dial plan you have configured for VoIP Caller Default DP. Then the number to be dialed is sent to the logic to take the FXO port off hook and dial the number.

When you are calling over the network from the distant caller your incoming Sip Invite is only calling ip_address:port_number. When the call is connected the SPA3102 returns dial tone and you send the *2 or *3 over the call circuit using the AVT protocol. This is processed by the dial plan you have setup as the VoIP Caller Default DP. Then the number to be dialed is sent to the logic to take the FXO port off hook and dial the number.

If the SPA3102 has some hidden bug inside the firmware you probably are not going to get it fixed as the SPA3102 is past the Cisco sale end of life and is in the final stages of Warranty and Service Contract Support. Cisco has no replacement product for the feature you are using. You need to find an alternative with the SPA3102 that works.

The weakest part of your setup is the transmission of the DTMF using the AVT protocol. We have pretty much established that that is working OK, although it could be sending double digits sometimes for a single key depression and that would cause trouble.

I know of no reason why the *2 and *3 substitution should not work, however I would not use something that starts with an asterisk (*) because the SPA3102 uses the * plus 2 digits to issue "Vertical Service Activation Codes" although these codes do not start with 2 or 3. However it is not in common use. The most common combination to use would be #2 or #3.

I also do not think the @gw0 in the PSTN Line Tab dial plans cause any trouble, however @gw0 is not needed there and is not common. The only destination for something dialed from an incoming call to the PSTN Line Tab is the FXO port. Obviously the @gw0 is needed, however, for a dial plan on the Line 1 Tab.

You are using direct ip dialing so another alternative would be to setup "one stage dialing" where you dial the number on your distant sip client and you have the client setup to send the number to an unregistered proxy (the SPA3102 PSTN Line Tab) as 12345678@ip_address:port in the Sip Invite and you do not send any DTMF. To do this you need to include a userid on the PSTN Line Tab. When a number on the incoming Sip Invite does not match the userid setup on the PSTN Line Tab "One Stage Dialing" is invoked.

[pero95305]

No, still this same story. Only what I'm getting is

"number not in service" or

"we sorry your number is not completely dial, check number and try again".

[Howard Wittenberg]

Sounds like something changed.  What were the changed setting when you got the messages from the PSTN service?

[pero95305]

No.

all the time I have this same messages before and after this changes.