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.<\/p>\n
Problem 1: VoIP gateway not reachable on incoming call. <\/strong> Problem 2: gateway working not properly behind NAT.<\/strong> Problem 3: Long waiting time before making a call.<\/strong> Problem 4: Alarm systems<\/strong> Codec: G711u or G711a PSTN parameters:<\/p>\n Ring Frequency: 20 Problem 5: Communication with elevators (emergency calling)<\/strong> Working modules: Not working: PSTN specification:<\/p>\n Ring waveform: Trapezoid DTMF specs:<\/p>\n DTMF Playback level: -16 Audio:<\/p>\n Preferred: G711a Configuration snip for Grandstream:<\/strong><\/p>\n Program — 1.0.8.4 Bootloader — 1.0.0.15 Core — 1.0.8.2 Base — 1.0.8.4<\/p>\n ADVANCED SETTINGS BASIC SETTINGS PROFILE 1 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\u2026<\/p>\n","protected":false},"author":1,"featured_media":185,"comment_status":"open","ping_status":"open","sticky":false,"template":"","format":"standard","meta":{"footnotes":""},"categories":[67,62],"tags":[298,304,306,300,308,314,316,318,320,322,324,326,328,330,332,336],"class_list":["post-187","post","type-post","status-publish","format-standard","has-post-thumbnail","hentry","category-electronics","category-networking","tag-long-delay","tag-dialplan-en","tag-dtmf-en","tag-elevators","tag-interdigit-en","tag-otis-en","tag-pap2t-en","tag-platan-en","tag-satel-en","tag-schindler-en","tag-sipura-en","tag-slican-en","tag-spa112-en","tag-spa122-en","tag-spa2102-en","tag-voip-en"],"aioseo_notices":[],"aioseo_head":"\n\t\t\n\t\n\t\n\t\n\t\n\t\t\n\t\t\n\t\t\n\t\t\n\t\t\n\t\t\n\t\t\n\t\t\n\t\t\n\t\t\n\t\t\n\t\t\n\t\t
\nMostly 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!<\/p>\n
\nTwo functionalities: 'NAT Mapping Enable’ and 'NAT Keep Alive Enable’. Chaning it to 'yes’ should solve the problem.<\/p>\n
\nHere 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*].)
\nAfter these changes calling number time decreases from ~10 to ~2-3 seconds.<\/p>\n
\nCommon problem. Mainly – DTMF is an issue here. My parameters works for me, but it may not fit into your solution.<\/p>\n
\nSilence Supp Enable: no
\nDTMF Process Info: yes
\nDTMF Process AVT: yes
\nDTMF TX Method: InBand (it modulates DTMF into RTP packets) – important issue!
\nDTMF TX Mode: Normal
\nDTMF Playback level: -16
\nDTMF Playback length: .1
\nDetect ABCD\/Playback ABCD: yes<\/p>\n
\nRing Voltage: 85
\nCWT Frequency: 440@-10
\nRing Waveform: trapezoid
\nSynchronized Ring: no
\nPort Impedance: 600
\nOutput\/Input port gain: -3<\/p>\n
\nSame 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:<\/p>\n
\nLinksys SPA2102 \/ Linksys PAP2T \/ Grandstream GXW4024<\/p>\n
\nCisco SPA112 \/ Cisco SPA122<\/p>\n
\nRing frequency: 25
\nRing voltage: 85-95
\nRing idle voltage: 50-60
\nFXS Port Input Gain: -3
\nFXS Port Output Gain: -3
\nFXS Port Impedance: 600<\/p>\n
\nDTMF Playback length: .1
\nDetect ABCD: enabled
\nPlayback ABCD: enabled
\nDTMF Process Info: yes
\nDTMF Process AVT: yes
\nDTMF Tx Method: InBand<\/p>\n
\nSilence Supp Enabled: no
\nEcho Canc Enable: yes
\nEcho Canc Adapt Enable: yes<\/p>\n
\nDial Tone: f1=425@,f2=425@-13,c=0\/0;
\nRingback Tone: f1=425@,f2=425@-19,c=2000\/4000;
\nBusy Tone: f1=425@-24,f2=425@-24,c=500\/500;
\nReorder Tone: f1=425@-24,f2=425@-24,c=250\/250;
\nConfirmation Tone: f1=425@-11,f2=425@-11,c=100\/100-100\/100-100\/100;
\nCall Waiting Tone: f1=425@-13,c=425\/10000-300\/10000-0\/0;
\nPrompt Tone: f1=425@-13,f2=425@-13,c=0\/0;<\/p>\n
\nPreferred DTMF method:
\n(in listed order)
\n Priority 1: RFC2833
\n Priority 2: SIP INFO
\n Priority 3: In-audio<\/p>\n
\nDisable DTMF Negotiation: Yes (use above DTMF order without negotiation)
\nDisable Call-Waiting: Yes
\nDial Plan: {xxxxxxxxx}
\nUse First Matching Vocoder in 200OK SDP: Yes
\nPreferred Vocoder:(in listed order)
\n choice 1: PCMA
\n choice 2: PCMU
\nVoice Frames per TX: 1
\nSymmetric RTP: No
\nFax mode: Pass-Through
\nFax tone detection mode: Callee
\nJitter buffer type: Adaptive
\nJitter buffer length: Medium
\nGain: TX 0db RX, 0db TX
\nDisable Line Echo Canceller (LEC): No<\/p>\n","protected":false},"excerpt":{"rendered":"