VoIP Gateways – issues of modern telephony

Cisco / Linksys series – PAP2T, SPA2102, PAP112, SPA122 are propably most widely used around the world for VoIP communication. Most common issues related to VoIP Gateways are unreachable phone in incoming calls, DTMF not passing through or not working properly with alarm/elevator systems.

Problem 1: VoIP gateway not reachable on incoming call.
Mostly this issue can be solved by adjusting the ‘Registration’ parameter. As a default it is set to 3600 seconds. I highly recommend chaning it to ‘120’. Solved!

Problem 2: gateway working not properly behind NAT.
Two functionalities: ‘NAT Mapping Enable’ and ‘NAT Keep Alive Enable’. Chaning it to ‘yes’ should solve the problem.

Problem 3: Long waiting time before making a call.
Here you need to do some experiments with values. There are many factors here, so be careful and change one parameter at the time. I’ve noticed that simple, analog phones are making calls faster than modern, cordless phones which you experience often in homes. First of all: ‘Interdigit Long Timer’ and ‘Interdigit Short Timer’. These values determines time which gateway waits between user pressing numbers on keypad. I recommend setting it to values ‘5’ and ‘3’. These seems working well. Next – move to ‘Dial Plan’. There are many manuals about proper configuration. Here (Poland) this one works best: (*xx|112S0|99[7-9]S0|0[347]0x.!|[1-9]xxxxxxxxS0|[x*][x*].)
After these changes calling number time decreases from ~10 to ~2-3 seconds.

Problem 4: Alarm systems
Common problem. Mainly – DTMF is an issue here. My parameters works for me, but it may not fit into your solution.

Codec: G711u or G711a
Silence Supp Enable: no
DTMF Process Info: yes
DTMF Process AVT: yes
DTMF TX Method: InBand (it modulates DTMF into RTP packets) – important issue!
DTMF TX Mode: Normal
DTMF Playback level: -16
DTMF Playback length: .1
Detect ABCD/Playback ABCD: yes

PSTN parameters:

Ring Frequency: 20
Ring Voltage: 85
CWT Frequency: 440@-10
Ring Waveform: trapezoid
Synchronized Ring: no
Port Impedance: 600
Output/Input port gain: -3

Problem 5: Communication with elevators (emergency calling)
Same here. DTMF issue all along. Configuration parameters which you see were gathered for about 2 years of experience with different manufacturers. Specifications which I was provided were far from reality. Cut to the case:

Working modules:
Linksys SPA2102 / Linksys PAP2T / Grandstream GXW4024

Not working:
Cisco SPA112 / Cisco SPA122

PSTN specification:

Ring waveform: Trapezoid
Ring frequency: 25
Ring voltage: 85-95
Ring idle voltage: 50-60
FXS Port Input Gain: -3
FXS Port Output Gain: -3
FXS Port Impedance: 600

DTMF specs:

DTMF Playback level: -16
DTMF Playback length: .1
Detect ABCD: enabled
Playback ABCD: enabled
DTMF Process Info: yes
DTMF Process AVT: yes
DTMF Tx Method: InBand


Preferred: G711a
Silence Supp Enabled: no
Echo Canc Enable: yes
Echo Canc Adapt Enable: yes

Configuration snip for Grandstream:

Program — Bootloader — Core — Base —

Dial Tone: f1=425@,f2=425@-13,c=0/0;
Ringback Tone: f1=425@,f2=425@-19,c=2000/4000;
Busy Tone: f1=425@-24,f2=425@-24,c=500/500;
Reorder Tone: f1=425@-24,f2=425@-24,c=250/250;
Confirmation Tone: f1=425@-11,f2=425@-11,c=100/100-100/100-100/100;
Call Waiting Tone: f1=425@-13,c=425/10000-300/10000-0/0;
Prompt Tone: f1=425@-13,f2=425@-13,c=0/0;

Preferred DTMF method:
(in listed order)
Priority 1: RFC2833
Priority 2: SIP INFO
Priority 3: In-audio

Disable DTMF Negotiation: Yes (use above DTMF order without negotiation)
Disable Call-Waiting: Yes
Dial Plan: {xxxxxxxxx}
Use First Matching Vocoder in 200OK SDP: Yes
Preferred Vocoder:(in listed order)
choice 1: PCMA
choice 2: PCMU
Voice Frames per TX: 1
Symmetric RTP: No
Fax mode: Pass-Through
Fax tone detection mode: Callee
Jitter buffer type: Adaptive
Jitter buffer length: Medium
Gain: TX 0db RX, 0db TX
Disable Line Echo Canceller (LEC): No

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