Topic: SAIL / Linksys SPA-3102...anyone have it working?

[veedar]

Has anyone had any luck getting the SPA-3102 running?

The SPA-3102 is the new Linksys version of the former Sipura SPA-3000. I'm following the SPA-3000 instructions to manually install in DocChapter253 but I can not get outgoing or incoming calls to work.

When I try I an outgoing call I see "circuit-busy" and "Everyone is busy/congested at this time" and hear "We're sorry please hang up..."

On the SPA-3102 the status shows as registered. But strangely when I run asterisk -rvvvvvvv from a terminal session I never see any SPA-3102 registration messages.

But in server-manager under Trunklines the state looks connected with this info...
  * Name       : 6179999999
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : mainmenu
  Subscr.Cont. : <Not set>
  Language     :
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  Mailbox      :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : Yes
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : 142
  Insecure     : no
  Nat          : RFC3581
  ACL          : No
  T38 pt UDPTL : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       :
  Addr->IP     : 192.168.242.253 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 6179999999
  SIP Options  : (none)
  Codecs       : 0xc (ulaw|alaw)
  Codec Order  : (alaw:20,ulaw:20)
  Auto-Framing:  No
  Status       : OK (8 ms)
  Useragent    : Linksys/SPA3102-5.1.7(GW)
  Reg. Contact : sip:6179999999@192.168.242.253:5060

Verbosity is at least 9

I'm running SME 7.1.3 with...
selintra-sail-2.1.15-453
smeserver-asterisk-1.4.1-3
smeserver-asterisk-zappri-MPP-1.4.0-4

If anyone has any advice I would be very grateful.

Regards,
Veedar

[SARK devs]

Hi There

It looks as though asterisk and the 3102 aren't quite connecting properly.  If you aren't seeing the registrations on your console then it isn't registering. To be honest we've not played witha 3102 so we don't know the exact differences between it and the old 3K.   However, that's easily remedied; we can talk to the UK distributor on tuesday (monday is a bank holiday here)  and see what,if any, the differences are.

In the meantime check that you have correctly entered the registry info on the 3102.  Also, plug up a phone to it and do *56* to see if asterisk will say your extension number.  watch what happens on the asterisk console log.

Kind Regards

Selintra

[veedar]

Thanks Selintra, I re-entered my registration info and was able to see registration activity in the console....

  -- Registered SIP '5000' at 192.168.242.253 port 5060 expires 180
  -- Registered SIP '6179999999' at 192.168.242.253 port 5060 expires 180

...but still I have my same problems and after expiration I do not see it re-register again.  But it does look like they are there...

asterisk*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status          
6179999999/6179999999      192.168.242.253  D          5060     OK (6 ms)            
5003/xlite                192.168.242.67   D          48994    OK (103 ms)    
5000/5000                  192.168.242.253  D          5060     OK (6 ms)      

Doing a *56* as you requested gives me a busy signal with no console output.  

Thanks for looking into this. I'll await your findings next week.   -Veedar

[SARK devs]

HI,

The registration looks good.  Interesting that you are getting a busy tone with phone plugged into 3102 and no console output.  SAIL generates no busy tones so this must be coming off the 3102 itself.  This leads me to suspect the 3102 dialplan.  Spa3K is effectively a small PBX in its own right.  Switching decisions are made in the dialplan.  What do you have in the 3102?

Best

Selintra

[veedar]

Thanks Selintra, I now have incoming calls and *56* working. I replaced a failed line splitter on my Verizon Telco line. So part of the problem was no dialtone getting to the 3102.

For outgoing calls the "circuit-busy" problem persists. I tried a bunch of different settings but no joy. My "PSTN Line" setting for Dial Plan is...

Dial Plan 2:   (S0<:6179999999>)

Regards,
Veedar

[SARK devs]

Sorry I didn't phrase that very well because there are several fields called dialplan.  The one I'm interested in is at the bottom of the Line  (not PSTN) page. It is next to the "Auto PSTN fallback" field in the SPA 3k.  There is a reference to it (and partial screen shot) in the docs pages at

http://www.selintra.com/docs/cgi-bin/view/Main/DocChapter253

Kind Regards

Selintra

[veedar]

Okay my Dial Plan from "Line 1" reads...

Dial Plan:   (*x.|*xx*|x.)

And here's the console output when I try to make an outgoing call via XLite...

asterisk*CLI>
    -- Executing [6171234567@internal:1] AGI("SIP/5003-08a87598", "selintra|OutCluster|6171234567") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script selintra completed, returning 0
    -- Executing [6171234567@default:1] AGI("SIP/5003-08a87598", "selintra|OutRoute|RouteName") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Set) Options: (CALLERID(NUMBER)=6179999999))
    -- AGI Script Executing Application: (Dial) Options: (SIP/6171234567@6179999999)
    -- Called 6171234567@6179999999
    -- SIP/6179999999-08acb788 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- AGI Script Executing Application: (Background) Options: (were-sorry)
    -- <SIP/5003-08a87598> Playing 'were-sorry' (language 'en')
    -- AGI Script Executing Application: (Background) Options: (call-cannot-complete)
[May 27 14:43:12] WARNING[4282]: file.c:553 ast_openstream_full: File call-cannot-complete does not exist in any format
[May 27 14:43:12] WARNING[4282]: file.c:804 ast_streamfile: Unable to open call-cannot-complete (format 0x4 (ulaw)): No such file or directory
[May 27 14:43:12] WARNING[4282]: pbx.c:5668 pbx_builtin_background: ast_streamfile failed on SIP/5003-08a87598 for call-cannot-complete
    -- AGI Script Executing Application: (Background) Options: (please-hang-up-and-try-again)
    -- <SIP/5003-08a87598> Playing 'please-hang-up-and-try-again' (language 'en')
    -- AGI Script selintra completed, returning 0
  == Auto fallthrough, channel 'SIP/5003-08a87598' status is 'CONGESTION'
asterisk*CLI>

[veedar]

Now this is interesting...to see what would happen, under "Line 1" I set  "Line enable" to "no". And now both outgoing and incoming calls are successful. Of course the phone I have plugged into the 3102 no longer works. It would seem there is a conflict somewhere.

[SARK devs]

Sounds like you're almost there.  Without a unit here it's difficult to advise you further.  

There is a very good forum on the Spa units on Voxilla.  You may want to take a look at that.

Best

Selintra

[veedar]

Thanks Selintra, I got it working by changing the Line1 port from 5060 to 5061. Chapter253 says to use 5060 for both Line1 and PSTN Line but I could not make that work.

http://www.selintra.com/docs/cgi-bin/view/Main/DocChapter253

Also as a suggestion it would be nice if there was a drop down menu when editing an extension where you could pick your unavailable / busy / welcome message.  The drop down menu for this could be populated from the Greetings panel.

[SARK devs]

Nice work,

Yes, that does look like an error in the docs.  I dimly seem to remember something about ports on the 3000 so I'm sure you're right.  Also, we like the idea of the dropdown so we'll add it to the build list - don't hold your breath tho' - the build list is just about long enough to span the globe at the moment.

No rest for the wicked...

Best

S

[ntblade]

Been following the instructions but so far I can't get the provisioning to work;
I gave the 3102 static IP address, enabled the TFTP srever in globals, added extension and trunk but nothing shows in asterisk or the 3102 web insterface.

Any clues?  Meanwhile I'll try manual setup.

Edit>
Followed the Manual setup and no success.  I can't even see the device try to register.

To be sure... the device should be plugged into the network port and not the Internet port right?

In the instructions it says http://www.selintra.com/docs/cgi-bin/view/Main/DocChapter253

Plug the spa3K into your network switch using an RJ45 patch lead. It will get an address from DHCP. To discover the IP address, plug a regular phone into the ext RJ11 port. LEAVE THE LINE PORT UNATTACHED. Take the phone off-hook and dial ****. The spa3K has its own on-board IVR. You will hear the spa3k reception greeting. Dial 110# and the spa3k will speak its IP address.

This is only true of the WAN port and the web interface can't be reach from that one.

Edit>
DOH! Plugged into WAN interface and now we're cooking.

N

[SARK devs]

Hi NT

You might do better with -505.   It has the new linksys provisioning support in it (including the 3102).

Best

S