Topic: Incoming Calls but no outgoing calls

[ronaldson40]

Hi

I am using Linksys SPA 3102 (192.168.1.237) along with my Asterisk Server (192.168.1.129). I followed selintra wiki for setting up the SPA3102 i.e through Manual method rather than setting it automatically through the DHCP via 66.
SPA3102 is connected throug the SIP protocol to Asteriisk,

Presently I am getting incoming calls (via xlite)[callerid also works] but when I try to make an outgoing call to my PSTN (analog phone line), the asterisk CLR says that it cannot find that extension and hangs up which is supposed to mean that there is no pathway to route the calls to the analog line.

I am a resident of Dubai, UAE.

This is my configuration:
On Asterisk:
-- Trunkline
    (PSTN line number) via SPA3012

--Extensions
   5000 : My SPA3102 (local)
   5001 : xlite (local)

-- Routes
    Name : Outgoing
    Primary path : <pstn line no - configured via trunkline)
    Dialing Plans : _XXXXXXX, _050XXXXXXX,_055XXXXXXX
(In Dubai -
Tel no : 7 digit number - no area code required
Mobiles : 050+7digit or 055+7 digit)

When I pick up my xlite, I just able to dial my extensions but not the local numbers and mobile phones. Can this problem arise because of improper configuration of Dial Plan on Line 1 tab under the SPA 3102?

My objective is just to use the normal analog line to setup an office PBX system not VOIP.

Regards
Ronald

-----------------------
SPA3012 Config

Line1 Tab
Dial Plan
(*x.|*xx*|x.)

PSTN line
Dial Plan2
(S0<:xyz>) where xyz represents my telephone number

[SARK devs]

Hello,

I need to see the console output when you make a call.
Also, you show the dial plans like this...

_XXXXXXX, _050XXXXXXX,_055XXXXXXX

..Do you really have those commas (,) in there?  The dial plan does not support commas.

Kind Regards

[ronaldson40]

This is my console output....

Code: [Select]

Connected to Asterisk 1.4.21.1 currently running on toshiba (pid = 3821)
Verbosity is at least 5
    -- Registered SIP '5001' at 192.168.1.245 port 1950 expires 180
    -- Saved useragent "X-Lite release 1011s stamp 41150" for peer 5001
[Jul  6 16:45:38] NOTICE[3883]: chan_sip.c:12669 handle_response_peerpoke: Peer '5001' is now Reachable. (28ms / 3000ms)
    -- Executing [0504686220@internal:1] AGI("SIP/5001-09c0abd0", "selintra|OutCluster|0504686220") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script selintra completed, returning 0
    -- Executing [0504686220@default:1] AGI("SIP/5001-09c0abd0", "selintra|OutRoute|out") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Dial) Options: (SIP/0504686220@3371704)
    -- Called 0504686220@3371704
    -- SIP/3371704-09c14320 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- AGI Script Executing Application: (Background) Options: (were-sorry)
[Jul  6 16:46:21] WARNING[5033]: file.c:602 ast_openstream_full: File were-sorry does not exist in any format
[Jul  6 16:46:21] WARNING[5033]: file.c:912 ast_streamfile: Unable to open were-sorry (format 0x8 (alaw)): No such file or directory
[Jul  6 16:46:21] WARNING[5033]: pbx.c:5722 pbx_builtin_background: ast_streamfile failed on SIP/5001-09c0abd0 for were-sorry
    -- AGI Script Executing Application: (Background) Options: (call-cannot-complete)
[Jul  6 16:46:21] WARNING[5033]: file.c:602 ast_openstream_full: File call-cannot-complete does not exist in any format
[Jul  6 16:46:21] WARNING[5033]: file.c:912 ast_streamfile: Unable to open call-cannot-complete (format 0x8 (alaw)): No such file or directory
[Jul  6 16:46:21] WARNING[5033]: pbx.c:5722 pbx_builtin_background: ast_streamfile failed on SIP/5001-09c0abd0 for call-cannot-complete
    -- AGI Script Executing Application: (Background) Options: (please-hang-up-and-try-again)
[Jul  6 16:46:21] WARNING[5033]: file.c:602 ast_openstream_full: File please-hang-up-and-try-again does not exist in any format
[Jul  6 16:46:21] WARNING[5033]: file.c:912 ast_streamfile: Unable to open please-hang-up-and-try-again (format 0x8 (alaw)): No such file or directory
[Jul  6 16:46:21] WARNING[5033]: pbx.c:5722 pbx_builtin_background: ast_streamfile failed on SIP/5001-09c0abd0 for please-hang-up-and-try-again
    -- AGI Script selintra completed, returning 0
  == Auto fallthrough, channel 'SIP/5001-09c0abd0' status is 'CONGESTION'
    -- Executing [h@default:1] Hangup("SIP/5001-09c0abd0", "") in new stack
  == Spawn extension (default, h, 1) exited non-zero on 'SIP/5001-09c0abd0'

I have presently put only _050XXXXXXX in the DP for the route.
I have a question. Since I am new to PBX systems, is there a prefix that I need to dial to direct my calls given the dialplan used on SPA3102. I remember sometime back, you needed to dial 8 to dial via PSTN.

[peterpan746]

I might help you with part of the question.  Unlike the legacy systems (Samsung, NEC etc.) you don't have to dial a digit to seize the line.  Therefore in your dial plan the first digit is actually the first digit of the number you are dialing.

I had a problem with that until Selintra pointed this out to me. I'll be watching this closely since in the near future I want to connect a 3102 to my system as well.

I hope some of my ranting helped you.

Best regards

[ronaldson40]

Thanks...

I would like to add one more info...

When I used SPA3102 standalone with it connected to my router and the xlite connected to it via the ethernet port, outgoing and incoming calls work(immediately after a factory reset). Its only when I interface it with asterisk does the SPA3012(on extension 5000) does it not work. This is just for your information.

[SARK devs]

All I can tell you from the data you've given is that SARK is working correctly.  It has processed your call and handed it to a SIP channel called 3371704.   You can see it in the trace...

Code: [Select]

Called 0504686220@3371704

SARK has handed off the number you dialled to the channel you nominated in your route.  The SIP UA at 3371704 has rejected the call and given back a SIP return code of CONGESTION/BUSY and refused the invite.

The problem lies at the SIP UA, which I assume is your 3102.

If you have correctly followed the guidelines on our /docs pages then it will work.  If you have set the 3102 up differently then it probably won't.

Kind Regards

[ronaldson40]

Hi Selintra

Thanks for the reply.

I followed upon the point you made about the misconfiguration of SPA3102.

I tried the steps listed on http://www.selintra.com/docs/cgi-bin/view/Main/DocChapter253#Setting_up_the_spa_3000_as_an_As but it did not work.

Then I tried going to SPA3102 support page on linksys.com.
http://www.linksys.com/servlet/Satellite?c=L_CASupport_C2&childpagename=US%2FLayout&cid=1169083369263&pagename=Linksys%2FCommon%2FVisitorWrapper

I followed the steps listed in the answers for the following questions :

How can I make an Outbound PSTN Call on SPA-3102?
http://linksys.custhelp.com/cgi-bin/linksys.cfg/php/enduser/std_adp.php?p_faqid=5175&lid=6866569263B09

How to forward PSTN Callers to a VoIP number on SPA-3102?
http://linksys.custhelp.com/cgi-bin/linksys.cfg/php/enduser/std_adp.php?p_faqid=5159&lid=6862769263B08

ITS WORKING!!!



I can now make inbound and outbound calls  via the SPA3102 and the askterisk also reports success...

Thanks once again for the support!

I would like to know how to make Asterisk now pick a call and play an IVRS message.
Presently I have made ext 5001 as the operator on Global settings page, so my xlite rings when a call arrives from the PSTN.

[SARK devs]

I would like to know how to make Asterisk now pick a call and play an IVRS message.

IVR setup is covered in the docs htttp://selintra.com/docs

Kind Regards