Topic: SPA2000 problem

[del]

Hi everyone, after having over a year off due to illness I decided to boot up my SME/SARK server, it all worked OK but was very out of date with both SME and SAIL so I decided to start from scratch and install SME8.0 and SAIL from the ISO on Selintra's web site, everything seemed to work OK except  I can't get my SPA2000 to ring in  I don't have any dial tone on the phone connected to it but I can still dial out and make a call OK. If I dial into the server the phone doesn't ring out, it doesn't ring if I dial it from another extension (Xlite) but it does ring the Xlite extension (even though their is no dial tone)  Any ideas? I've checked and double checked the SPA2000 settings and can't see anything wrong. The PBX says it's connected and the SPA2000 says it's registered. Any help is appreciated 

Del

[SARK devs]

Hello Del

Hope you are OK now.  It looks like you may have more than one issue here.  The fact that you don't get dialtone is clearly a problem and I guess you'll need to look at the Spa manual to figure out why.  To see why phones can't dial the spa extension you will need to watch the Asterisk console to see what is happening when you make a call.  If you still can't see why then post the console output here and we'll take a look at it.

Kind Regards

S

[del]

Hi Selintra,

I'm a lot better now thanks, the out put is:

login as: root
root@192.168.0.10's password:
Last login: Tue Oct 11 18:44:12 2011 from pc-00003.ddj.homelinux.com
[root@sark-pbx ~]# asterisk -vvvvvr
Asterisk 1.8.5.0, Copyright (C) 1999 - 2011 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for detail                                                                                               

                                                             s.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.8.5.0 currently running on sark-pbx (pid = 3167)
Verbosity is at least 5
    -- Remote UNIX connection
  == Using SIP RTP CoS mark 5
    -- Executing [402@internal:1] AGI("SIP/401-00000009", "sarkhpe,OutCos,402") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
    -- <SIP/401-00000009>AGI Script sarkhpe completed, returning 0
    -- Executing [402@401closedcos:1] AGI("SIP/401-00000009", "sarkhpe,OutCluster,402") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=default)
    -- <SIP/401-00000009>AGI Script sarkhpe completed, returning 0
    -- Executing [402@qrxvtmny:1] AGI("SIP/401-00000009", "sarkhpe,InCall,") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Dial) Options: (SIP/402,20,ktT)
[Oct 11 18:50:24] WARNING[30790]: app_dial.c:2196 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- AGI Script Executing Application: (Voicemail) Options: (402,u)
    -- <SIP/401-00000009> Playing 'vm-theperson.gsm' (language 'en-gb')
[Oct 11 18:50:24] NOTICE[30790]: channel.c:4128 __ast_read: Dropping incompatible voice frame on SIP/401-00000009 of format alaw since our native format has changed to 0x4

(ulaw)
    -- <SIP/401-00000009> Playing 'digits/4.gsm' (language 'en-gb')
    -- <SIP/401-00000009> Playing 'digits/0.gsm' (language 'en-gb')
    -- <SIP/401-00000009> Playing 'digits/2.gsm' (language 'en-gb')
    -- <SIP/401-00000009> Playing 'vm-isunavail.gsm' (language 'en-gb')
    -- <SIP/401-00000009> Playing 'vm-intro.gsm' (language 'en-gb')
       > doing dnsmgr_lookup for 'sipgate.co.uk'
       > ast_get_srv: SRV lookup for '_sip._udp.sipgate.co.uk' mapped to host sipgate.co.uk, port 5060
       > doing dnsmgr_lookup for 'sipgate.co.uk'
       > ast_get_srv: SRV lookup for '_sip._udp.sipgate.co.uk' mapped to host sipgate.co.uk, port 5060
       > doing dnsmgr_lookup for 'sipgate.co.uk'
       > ast_get_srv: SRV lookup for '_sip._udp.sipgate.co.uk' mapped to host sipgate.co.uk, port 5060
       > doing dnsmgr_lookup for 'sipgate.co.uk'
       > ast_get_srv: SRV lookup for '_sip._udp.sipgate.co.uk' mapped to host sipgate.co.uk, port 5060
       > doing dnsmgr_lookup for 'sipgate.co.uk'
       > ast_get_srv: SRV lookup for '_sip._udp.sipgate.co.uk' mapped to host sipgate.co.uk, port 5060
       > doing dnsmgr_lookup for 'sipgate.co.uk'
       > ast_get_srv: SRV lookup for '_sip._udp.sipgate.co.uk' mapped to host sipgate.co.uk, port 5060
    -- <SIP/401-00000009> Playing 'beep.gsm' (language 'en-gb')
    -- Recording the message
    -- x=0, open writing:  /var/spool/asterisk/voicemail/default/402/tmp/L9YdWF format: wav49, 0x9e52f48
    -- User hung up
    -- Recording was 0 seconds long but needs to be at least 2 - abandoning
    -- <SIP/401-00000009>AGI Script sarkhpe completed, returning 4
  == Spawn extension (qrxvtmny, 402, 1) exited non-zero on 'SIP/401-00000009'
    -- Executing [h@qrxvtmny:1] Hangup("SIP/401-00000009", "") in new stack
  == Spawn extension (qrxvtmny, h, 1) exited non-zero on 'SIP/401-00000009'
sark-pbx*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
[root@sark-pbx ~]#

Also the voicemail cuts off after the message telling you to leave a message is finished 

Regards,
Del

[del]

Update, today we had a power cut and when it came back on everything works OK, still no actual dial tone though and I can't see any settings in the SPA2000 that look any different from when I had dial tone