[FernandoHerrera]

I have had my SPA-3102 for several months, as client of an Asterisk-based network of mine composed of some 20 VoIP ATAs (all of them Sipura or Linksys). For the first time, I am trying to use the telco port, but this has been a complete failure.

Port 1 (FXS) has extension number 6000 and port 2 (FXO) has extension number 6500.

The fundamental problem is that I never hear the telco dial tone, just a quiet dial tone provided by the Linksys box, and most importantly: the dial tone DOES NOT allow me to dial out.

There are two things that I have tried:

(1) Dial 6500, I can use a PIN or not. I get the 3102 internal dial tone. This works fine up to here.

(2) Set up Dial plans such as: "911S0 <:@gw0>" (I actually use 119 which is the time in this country). I see this log message:

Last Called Number: 119@gw0

In both cases above, the last accomplishment is to hear the 3102 dial tone, and I cannot progress any further. As I start dialing a phone number, the dial tone shuts down (as it should) and that's the end of it. A few seconds later I get a busy tone, regardless of what I dial (or even if I don't dial anything).

I have tried enabling an disabling the "VoIP-To-PSTN Gateway" option, with no difference.

I have tried:

VoIP Access List: 127.0.0.1 to allow access to the localhost. Still can't dial out.

The unit is running the latest firmware 5.1.7(GW), and just in case I reinstalled it. I did a reset to factory. The problem remains.

TIA for your kind assistance,

-Ramon F. Herrera

[hwittenb]

I would concentrate on getting the 119 dialing working first. On the Line 1 and the PSTN tabs it should say either G711u or Use Preferred Codec Only NO, on the PSTN tab it should say One Stage Dialing Yes and the Voip-to-Pstn gateway pointers to dial plans should point to a valid one like (xx.). The PSTN tab Line Enable and the Voip-to-PSTN Gateway Enable should be YES.

Some things to check for troubleshooting:

1. After you complete a call to 119 what does the SPA3102 Info tab say was the "Last PSTN Disconnect Reason". The SPA checks a number of things to try to detect a PSTN disconnect. Maybe you are getting a false reading as soon as you take the FXO port off hook. If so, try to disable the check for that reason for the disconnect.

2. The SPA won't take the FXO port off hook if it thinks it is busy. It does this by checking the line voltage. The voltage needs to be higher than the "Line In Use Voltage". You can read the on-hook and off-hook voltage at the appropriate time on the Info tab by refreshing the screen. Typical PSTN voltages are 48v on hook, 7v off hook. PBX voltages will differ.

3. If the two reasons above look OK, then I would run a Sip Debug trace by downloading the program from the Sipura web site:
http://www.sipura.com/Documents/faq/Section_2.html#9
Be sure to set both the Debug Level to 3 on the System Tab and the Sip Debug Option to Full on the PSTN tab. This should show the progress of the call.

[FernandoHerrera]

Thank you so much, hwittenb!!

The problem was indeed the CPC signal, which apparently is not provided by the Telco here in Venezuela, and the Linksys unit was trying to detect it.

Another really good tip you gave me is the syslog for Windows. As I am mostly a server guy, I was about to set up a Linux server at home (my regular servers are in another continent, behind a firewall) just to catch those precious syslog lines.

My next project is to master the arcane art of dialplan writing. Seems like a lot of fun...

Thanks again! This is the DEFINITE BEST forum for VoIP, I wish I had discovered it earlier.

-Ramon

[voip_addict]

I am having a very similar problem and hopefully I'm posting this in the right place.



I dial #9 to make a call ..

I get a fake dial tone from the 3102. I Then dial only to have the 3102 not dial anything over the PSTN line. Then I get a rapid busy signal on the handset. This is making me insane.

More Info:
Not sure if this can help.. but When registration fails for one reason or another the 3102 doesn't fall back to PSTN either. All I get on the handset is the same rapid busy signal sound. If I listen to that rapid busy signal sound it changes tone after about 10 seconds.

More Info:
Incoming calls from VoIP or from PSTN ring through on the handset connected to the 3102 with no problems. Outbound calls from the handset to VoIP work fine too.

A little more about my configuration:

Line 1: FWD account with IPKALL did forwarded to it. (requires registration)
PSTN Line: My VoIP account for outbound calls (require registration)

Dial Plan:
In Dial Plan I have <#9,;>[*x]x.<:@gw0>



Any assistance is greatly appreciated,

VoIP_Addict





NOTE:
Ive been reading these forums for about a year and have had NO success. At one stage at lost my user password... created another forum user account and posted a question only to receive a message that I was a newbie and should be reading the forums more before asking questions.
Anyway.. at least 5 months have gone by.. hopefully I don't get accused of joining and asking a question on the same as 'resigning up.

[hwittenb]

Originally Posted by voip_addict

I dial #9 to make a call ..
In Dial Plan I have <#9,;>[*x]x.<:@gw0>

If you didn't make a typographical error, the dial plan should be:
<#9,:>[*x]x.<:@gw0>
where following the #9, is a colon not a semicolon as you have shown.

If you are trying to dial an * on the FXO port the relevant dial plan on the PSTN tab must also allow an * in the dial string.

[FernandoHerrera]

VoIP Addict:

The first unusual thing that I noticed is your strange dialplan:

> In Dial Plan I have <#9,;>[*x]x.<:@gw0>

According to the many dialplans I have seen on the net, there are only *two* valid ways to use the angle bracket notation:

(1) Single bracket pair:

911 <:@gw0>

(2) Dual bracket pair:

<911|171> <:@gw0>

The meaning of first case above should be obvious, while the second is something that I would use in this country, where the emergency number is 171, in order to help a confused US visitor.

Anyway, my point is that the dual bracket case always involves a substitution and therefore the "|" symbol us always used. The dialplan you posted doesn't match any of the two above forms.

My other comment is about the "Last PSTN Disconnect Reason". Have you looked at it?

Additionally, you should take a look at the "One Stage Dialing" parameter, but only after you have your main issue resolved. That parameter brings some convenience but it cannot explain the lack-of-connection symptom that you are experiencing.

-Ramon

[FernandoHerrera]

After reading the last post from hwittenb, allow me to reprhase what I wrote earlier.

Here is the rule:

In every case where the angle bracket notation is used, there has to be one colon ":" inside each set of angle brackets.

I wish I could find some formal description of the dialplan notation, as it still manages to get me confused. I have learned it (or semi-leaned it) by just looking at examples on the net, which aren't always necessarily correct.

Thanks again, hwittenb...

-Ramon

[voip_addict]

Originally Posted by hwittenb

If you didn't make a typographical error, the dial plan should be:
<#9,:>[*x]x.<:@gw0>
where following the #9, is a colon not a semicolon as you have shown.

If you are trying to dial an * on the FXO port the relevant dial plan on the PSTN tab must also allow an * in the dial string.

Oops! That was a typo ~ dial plaln has a semi colon

[voip_addict]

Well thank you both for you help.

You know I just realized something.. a while back I thought I mastered Sipura dial plans. I felt so good it was a battle just to keep the big grin off my face.

Well after not looking at them for a while.. I have to spend a few days reading up on them cause I forgot half of what I knew about them.

.. And I was considering buying a bicycle. Good thing I didn't.. probably would have had to start off with training wheels again!!!


VoIP_Addict

[rookiebot]

i'm facing an issue here of similiar nature ... i want certain numbers dialled to be diverted to the pstn ... but not happening ... not clear on whats going wrong - cud only see RSE:GetServerAddrErr(gw0 ,0)=-101. Tried most of whats said above. But dint work out. These are the details of my config-

Dial Plan:
( **275*x. | *75xx | 05[05]xxxxxxxS0 < : @gw0 > | [1-9]xxxxxxS0 < : @gw0 > | 0[24678]xxxxxxxS0< : @gw0 > | *xx. | *xxx | xxxxxxxxxxxx. )

SysLog:
DateTime, Source, Facility, Priority, Message
8:42:08 PM, 127.0.0.1, 19, 7, 20:42:08 \SPA-IP\ [0]Off Hook
8:42:10 PM, 127.0.0.1, 18, 7, 20:42:10 \SPA-IP\ 2. Report digit 0 (1)(40 ms)
8:42:10 PM, 127.0.0.1, 18, 7, 20:42:10 \SPA-IP\ 2. Report digit 5 (1)(40 ms)
8:42:11 PM, 127.0.0.1, 18, 7, 20:42:11 \SPA-IP\ 2. Report digit 0 (1)(40 ms)
8:42:11 PM, 127.0.0.1, 18, 7, 20:42:11 \SPA-IP\ 2. Report digit 5 (1)(40 ms)
8:42:11 PM, 127.0.0.1, 18, 7, 20:42:11 \SPA-IP\ 2. Report digit 9 (1)(40 ms)
8:42:11 PM, 127.0.0.1, 18, 7, 20:42:11 \SPA-IP\ 2. Report digit 9 (1)(40 ms)
8:42:12 PM, 127.0.0.1, 18, 7, 20:42:12 \SPA-IP\ 2. Report digit 5 (1)(40 ms)
8:42:12 PM, 127.0.0.1, 18, 7, 20:42:12 \SPA-IP\ 2. Report digit 3 (1)(40 ms)
8:42:13 PM, 127.0.0.1, 18, 7, 20:42:13 \SPA-IP\ 2. Report digit 7 (1)(40 ms)
8:42:13 PM, 127.0.0.1, 18, 7, 20:42:13 \SPA-IP\ 2. Report digit 2 (1)(40 ms)
8:42:13 PM, 127.0.0.1, 18, 7, 20:42:13 \SPA-IP\ [5060]STUN trying 0
8:42:13 PM, 127.0.0.1, 18, 7, 20:42:13 \SPA-IP\ [16384]STUN trying 0
8:42:13 PM, 127.0.0.1, 18, 7, 20:42:13 \SPA-IP\ [16385]STUN trying 0
8:42:13 PM, 127.0.0.1, 18, 7, 20:42:13 \SPA-IP\ [16386]STUN trying 0
8:42:13 PM, 127.0.0.1, 18, 7, 20:42:13 \SPA-IP\ [16387]STUN trying 0
8:42:13 PM, 127.0.0.1, 18, 7, 20:42:13 \SPA-IP\ [5060]STUN trying 1
8:42:13 PM, 127.0.0.1, 18, 7, 20:42:13 \SPA-IP\ [16384]STUN trying 1
8:42:13 PM, 127.0.0.1, 18, 7, 20:42:13 \SPA-IP\ [16385]STUN trying 1
8:42:13 PM, 127.0.0.1, 18, 7, 20:42:13 \SPA-IP\ [16386]STUN trying 1
8:42:13 PM, 127.0.0.1, 18, 7, 20:42:13 \SPA-IP\ [16387]STUN trying 1
8:42:13 PM, 127.0.0.1, 18, 7, 20:42:13 \SPA-IP\ [5060]STUN trying 2
8:42:13 PM, 127.0.0.1, 18, 7, 20:42:13 \SPA-IP\ [16384]STUN trying 2
8:42:13 PM, 127.0.0.1, 18, 7, 20:42:13 \SPA-IP\ [16385]STUN trying 2
8:42:13 PM, 127.0.0.1, 18, 7, 20:42:13 \SPA-IP\ [16386]STUN trying 2
8:42:13 PM, 127.0.0.1, 18, 7, 20:42:13 \SPA-IP\ [16387]STUN trying 2
8:42:14 PM, 127.0.0.1, 18, 7, 20:42:14 \SPA-IP\ [0:0]CC:STUN OK:c0a80002->d0650e94, 5060->62832 16385->63068
8:42:14 PM, 127.0.0.1, 18, 7, 20:42:14 \SPA-IP\ Calling:0505975399 @gw0 :0
8:42:14 PM, 127.0.0.1, 18, 7, 20:42:14 \SPA-IP\ [0:0]AUD ALLOC CALL (port=16385)
8:42:14 PM, 127.0.0.1, 18, 7, 20:42:14 \SPA-IP\ [0:0]RTP Rx Up
8:42:17 PM, 127.0.0.1, 18, 7, 20:42:17 \SPA-IP\ RSE:GetServerAddrErr(gw0 ,0)=-101
8:42:17 PM, 127.0.0.1, 18, 7, 20:42:17 \SPA-IP\ TP:?Tx->0
8:42:17 PM, 127.0.0.1, 19, 7, 20:42:17 \SPA-IP\ [0]FM Alert Stop RxTx (c=0024a404;a=0)
8:42:18 PM, 127.0.0.1, 18, 7, 20:42:17 \SPA-IP\ [0:0]AUD Rel Call
8:42:18 PM, 127.0.0.1, 18, 7, 20:42:17 \SPA-IP\ CC:Failed w/ Calling
8:42:18 PM, 127.0.0.1, 18, 7, 20:42:17 \SPA-IP\ Sess Terminated
8:42:29 PM, 127.0.0.1, 19, 7, 20:42:29 \SPA-IP\ [0]On Hook