<div dir="ltr"><div>Dear all, </div>
<div> I connected a normal phone line to the FXO port but the call is not being processed. The following is the output to asterisk console when I dial 9150 "9 is the prefix I configured and 150 is a local service in to know the current time"</div>
<div><font color="#ff0000">*CLI> -- Executing Dial("SIP/4444-d365", "Zap/1/150") in new stack <br> -- Called 1/150 <br> -- Zap/1-1 answered SIP/4444-d365 </font></div>
<div><br><font color="#ff0000">Here are some more details to help in troubleshooting the problem</font></div>
<div><font color="#ff0000">Dmesg Output is as following:</font></div>
<div><font color="#ff0000">Zapata Telephony Interface Registered on major 196 <br>Code test: code function addr = 0x004894f4 <br>iRxBuffer1 = 0xff803e58 <br>
iTxBuffer1 = 0xff803ed8 <br>ISR installed OK <br>port: 1 port_type: O <font color="#000000">indicate that asterisk is detect two FXO</font> <br>
port: 2 port_type: O <br>port: 3 port_type: - <br>port: 4 port_type: - <br>
port: 5 port_type: - <br>port: 6 port_type: - <br>port: 7 port_type: - <br>
port: 8 port_type: - <br>Testing for DAA... <br> VoiceDAA System: 04 <br>
ISO-Cap is now up, line side: 03 rev 06 <br>Module 0: Installed -- AUTO FXO (FCC mode) <br>Testing for DAA... <br>
VoiceDAA System: 04 <br> ISO-Cap is now up, line side: 03 rev 06 <br>Module 1: Installed -- AUTO FXO (FCC mode) <br>
Found: Blackfin STAMP (8 modules) <br>wcfxs_init_ok = 1 <br>Registered tone zone 0 (United States / North America) <br>
73318 Polarity reversed (0 -> 1)</font> </div>
<div><br>I hashed the following lines in zapata.conf corresponding to the FXS ports which are not installed:</div>
<div> </div>
<div><font color="#ff0000">;signalling=fxo_ks<br>;group=2 <br>;context=internal<br>;channel=> 3-4</font> </div>
<div> </div>
<div>I added the following line to the dial plan under the internal context in the extensions.conf file to enable routing to the FXO port at channel 1.</div>
<div> </div>
<div><font color="#ff0000">exten => _9.,1,Dial(Zap/1/${EXTEN:1})</font></div>
<div><font color="#ff0000"></font> </div>
<div>I added the following section to sip.conf for my IP phone </div>
<div> </div>
<div><font color="#ff0000">[4444] <br>type=friend <br>host=dynamic <br>user=4444 <br>secret=1234 <br>
disallow=all <br>allow=ulaw <br>context=internal</font></div>
<div> </div>
<div>Here are the output of some asterisk commands which might also be useful:</div>
<div><br>*<font color="#ff0000">CLI> zap show channels <br> Chan Extension Context Language MusicOnHold <br> pseudo incoming <br>
1 incoming <br> 2 incoming </font></div>
<div><br><font color="#ff0000">*CLI> zap show cadences <br>r1: 125,125,2000,4000 <br>r2: 250,250,500,1000,250,250,500,4000 <br>
r3: 125,125,125,125,125,4000 <br>r4: 1000,500,2500,5000 </font></div>
<div><font color="#ff0000">*CLI> zap show channel 1 <br>Channel: 1 <br>File Descriptor: 12 <br>
Span: 1 <br>Extension: <br><font color="#009900">Dialing: no</font> <font color="#330000"> why?</font> <br>
Context: incoming <br>Caller ID string: <br>Destroy: 0 <br>
InAlarm: 0 <br>Signalling Type: FXS Kewlstart <br>Owner: Zap/1-1 <br>
Real: Zap/1-1 <br>Callwait: <None> <br>Threeway: <None> <br>
Confno: -1 <br>Propagated Conference: -1 <br>Real in conference: 0 <br>
DSP: yes <br>Relax DTMF: no <br>Dialing/CallwaitCAS: 0/0 <br>
Default law: ulaw <br>Fax Handled: no <br>Pulse phone: no <br>
Echo Cancellation: 128 taps, currently ON <br>DSP cycles last: 752066 worst: 830646 average: 741569 sample: 732930 <br>Actual Confinfo: Num/0, Mode/0x0000 <br>
Actual Confmute: No <br>Actual Hookstate: Onhook </font></div>
<div> </div>
<div> I can measure 52V across the legs of the fxo module. Also when I plug the cable I receive "<font color="#ff0000">Ring on 1/1"</font> message on the console and when I remove the cable I receive <font color="#ff0000">"No Ring on 1/1". </font><font>so the PSTN line is sensed by asterisk.</font></div>
<p>When I try to make a call from my sip phone "4444" on zap 1 port it gave <font color="#ff0000">(no dial tone).</font></p>
<p><br><font color="#FF0000">*CLI> -- Executing Dial("SIP/4444-d365", "Zap/1/150") in new stack <br>-- Called 1/150 <br>-- Zap/1-1 answered SIP/4444-d365 </font></p>
<p>Sometimes it hangs up immediately after answered message. Sometimes it lasts until I hangup SIP session.</p>
<p>Here is the output of Dmesg during this test.</p>
<p><font color="#FF0000">RING on 1/1!<br>136232 Polarity reversed (-1 -> 1)<br>NO BATTERY on 1/1!<br>BATTERY on 1/1 (-)!<br>NO BATTERY on 1/1!<br>BATTERY on 1/1 (-)!<br>NO BATTERY on 1/1!<br>NO RING on 1/1!<br>
142815 Polarity reversed (1 -> -1)<br>
BATTERY on 1/1 (+)!<br>RING on 1/1!<br>149257 Polarity reversed (-1 -> 1)<br>NO BATTERY on 1/1!<br>BATTERY on 1/1 (-)!</font></p>
<p>I wanted to see the voltage variations on the line during the call to justify this behavior. I measured the voltage during my test and there was no variation in voltage across the fxo module legs at all. Is it expected for the voltage to go down when the call start (off hook). It was 52 and remains 52 during all tests. </p>
<p>I really don't know why "<font color="#FF0000">polarity reversed" and "NO Battery on 1/1"</font> messages are generated during the call. </p>
<p>When I make incoming call. There is no messages at all on asterisk console or driver messages on dmesg.</p>
<p><span style="border-collapse:collapse">I tried to avoid the hang up because this false polarity reversed signal. I put the following in zapata.conf. Now it stopped hanging up. But still the same no dial tone and no call progress.<br>
<br><font color="#FF0000">hanguponpolarityswitch=no</font></span></p><p><font color="#FF0000"><span style="border-collapse:collapse">I suspect about the clock that supplied to the FXO card it is 2.048MHZ (E1),however I use USA zone configuration and USA work with standard T1 (1.544MHZ).</span></font></p>
<p><font color="#FF0000"><span style="border-collapse:collapse">am i right?<br></span></font>Any hints ?</p><p>Thanks in advance.</p><p><br></p></div>