CISCO SPA112 Problems Voice Gateway with DTMF

Antonio B · ‎09-08-2021

Hello, everyone,

 

I have the problem that I connected 2 analog telephones to the SPA112 via the 2 FXS ports.

I have a SIP channel on both lines that are also registered.

The two analog telephones are standard devices, a Siemens Optipoint and a Gigaset DA 210

these are only intended as a test. In the future, a Siedle DoorCom and an analog mobile part will be connected to these two channels. This handset should transmit a code which then opens the door. Unfortunately I only found a configuration report from the company Siedle for the Linksys PAP2T. And unfortunately I am still very green when it comes to VoIP / SIP and analog devices or voice gateways.

I have faced the problem that when I transmit DTMF tones or when I trigger them on the analog phone, they are not passed on. I found this out via an IVR in which DTMF, so to speak, triggers a connection to another call number / sub-office.

Could someone give me a clue or share solutions and experiences on this topic with me?

I know unfortunately it is probably very special.

 

I would like to thank everyone in advance who found the time to take on my problem 

Thanks for your Help

bye Antonio

Dan Lukes · ‎09-08-2021

It's not so special, it's rather common. There is more than one method to pass DTMF signalling over SIP. You need to configure method recognized by the peer (the PBX/call control device the SPA1xx device is registered to). If you're using in-band then compressed codec MUST NOT be used. Use PCMA (PCMU in USA and Japan) instead.

Antonio B · ‎09-09-2021

Good morning Dan, and thank you for your answer. What is the difference between PCMA and PCMU? I am here in Bavaria and have found a configuration report for the Linksys PAP2T from the manufacturer of the door intercom (Siedle) which also includes a door opener (and which I would like to control via one of the two FXS ports with a DTMF sequence of numbers) probably the predecessor of the SPA112 was. I have followed this report here. Four settings were recommended there, but they don't mean anything to me? Once it was recommended to set the interdigit short timer from 3 to 1 and the interdigit long timer from 10 to 2 in the regional settings under Controll Timer Values and under the Miscellaneous settings it was recommended to record the FXS port impedance of 600 (ohms, I assume) 220 + 820 || 115nF, the FXS Port Output Gain from -3 (1 + n) to 10 (In-Home, Multi) - although I don't understand the meaning of In-Home, Multi - and the Caller ID FSK standard (modulation ) from Bell202 to v.23. Further settings were not described. I made the settings and tested them on an IVR with two analog telephones and two SIP TRUNKS as mentioned in my post. In your opinion, are these settings of the report helpful or rather obstructive? I thank you for your support. Greetings Antonio

Dan Lukes · ‎09-09-2021

What is the difference between PCMA and PCMU? I am here in Bavaria

Not the best place for depth-in description of difference. Read G.711 or consider them just two uncompressed codec. Use PCMA in Bavaria unless your upstream phone service provider asked you to use PCMU.

it was recommended to set the interdigit short timer from 3 to 1 and the interdigit long timer from 10 to 2 in the regional settings under Controll Timer Values

Unrelated to DTMF, it's about dialing.

under the Miscellaneous settings it was recommended to record the FXS port impedance of 600 (ohms, I assume) 220 + 820 || 115nF, the FXS Port Output Gain from -3 (1 + n) to 10 (In-Home, Multi) - although I don't understand the meaning of In-Home, Multi

There's no universal settings of "one fit all" kind. Port impedance is parameter of analog loop between SPA112 and analog phone (or other phone-like device like intercom) connected to it. Configure whatever impedance on both ends, but it must be the same on both ends of analog loop. Configuration of this may affect reliability of DTMF transmission, but only log captured from SPA112 will tell us it's the true cause of your issue or not.

Caller ID FSK standard (modulation ) from Bell202 to v.23.

This is related to caller ID transmission from SPA112 to the analog device connected. It must match connected device's requirements. If you can see caller id on incomming call, then it works. Unrelated to DTMF transmission.

You mentioned no most important configuration option - DTMF Tx Method

According description:

Method to transmit DTMF signals to the far end:

Inband = Send DTMF using the audio path;
INFO = Use the SIP INFO method
AVT = Send DTMF as AVT events
Auto = Use Inband or AVT or INFO based on outcome of codec negotiation

The method configured must match your upstream provider requirements or DTMF may not be passed at all.

Catch logs from SPA112 and catch SIP packets on the wire - those related to the failing call (ask your LAN administrator for help if necesarry). It may disclose more about your issue. See Debug and syslog Messages from SPA1x2 and SPA232D ATA