[Help] Asterisk@home 2.4 and Vega 50 configuration questions

Hello,

I am trying to get 2 8 port Vegastream Vega 50’s (one FXS, one FXO) to work with *@home 2.4. Right now I’m focusing on getting the FXS to work. So far I’ve been able to get the PBX to ring an extension on the Vega from a SIP soft phone however I can’t seem to get the Vega to initiate a call to the PBX. Any help on that would be greatly appreciated! I’ll include any files anyone needs to help me figure this out. I realize it is probably a dialing rule issue in the Vega but I can’t which other forums to post this on. I’m willing to try anything!

Thank you,

Tom

OK, an update. I’ve got the FXS working (thanks to a VERY nice guy at Netxusa for sending me the dialing plan manual), sort of. The current problem is when I place a call from the FXS device it goes to the IVR. I need to figure out how to let * know that it is an extension. Any ideas anyone?

Tom

Here is a chunk of my log file.

Feb 3 17:43:01 DEBUG[2399] chan_sip.c: Checking SIP call limits for device
Feb 3 17:43:01 DEBUG[2399] chan_sip.c: build_route: Contact hop:
Feb 3 17:43:01 VERBOSE[3179] logger.c: – Executing Goto(“SIP/192.168.0.5-08ca9ae0”, “from-pstn-timecheck|s|1”) in new stack
Feb 3 17:43:01 VERBOSE[3179] logger.c: – Goto (from-pstn-timecheck,s,1)
Feb 3 17:43:01 DEBUG[3179] pbx.c: Expression result is '1’
Feb 3 17:43:01 VERBOSE[3179] logger.c: – Executing GotoIf(“SIP/192.168.0.5-08ca9ae0”, “1?from-pstn-reghours|s|1:”) in new stack
Feb 3 17:43:01 VERBOSE[3179] logger.c: – Goto (from-pstn-reghours,s,1)
Feb 3 17:43:01 DEBUG[3179] pbx.c: Expression result is '0’
Feb 3 17:43:01 VERBOSE[3179] logger.c: – Executing GotoIf(“SIP/192.168.0.5-08ca9ae0”, “0?from-pstn-reghours-nofax|s|1:2”) in new stack
Feb 3 17:43:01 VERBOSE[3179] logger.c: – Goto (from-pstn-reghours,s,2)
Feb 3 17:43:01 VERBOSE[3179] logger.c: – Executing Answer(“SIP/192.168.0.5-08ca9ae0”, “”) in new stack
Feb 3 17:43:01 VERBOSE[3179] logger.c: – Executing Wait(“SIP/192.168.0.5-08ca9ae0”, “1”) in new stack
Feb 3 17:43:01 DEBUG[2399] chan_sip.c: Stopping retransmission on ‘0010-E24C-81E7E1DF-0@192.168.0.10’ of Response 82313018: Match Found
Feb 3 17:43:01 DEBUG[2399] chan_sip.c: Auto destroying call ‘000C-0A77-BBFE46CB-99@192.168.0.10’
Feb 3 17:43:01 NOTICE[3179] rtp.c: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.0.10
Feb 3 17:43:01 DEBUG[2399] chan_sip.c: Stopping retransmission on ‘7375a7d77a8fc6fa42e267f20d5f4f76@127.0.0.1’ of Request 163: Match Not Found
Feb 3 17:43:02 VERBOSE[3179] logger.c: – Executing SetVar(“SIP/192.168.0.5-08ca9ae0”, “intype=aa_1”) in new stack
Feb 3 17:43:02 VERBOSE[3179] logger.c: – Executing Cut(“SIP/192.168.0.5-08ca9ae0”, “intype=intype|-|1”) in new stack
Feb 3 17:43:02 DEBUG[3179] pbx.c: Expression result is ‘0’
Feb 3 17:43:02 VERBOSE[3179] logger.c: – Executing GotoIf(“SIP/192.168.0.5-08ca9ae0”, “0?7:9”) in new stack
Feb 3 17:43:02 VERBOSE[3179] logger.c: – Goto (from-pstn-reghours,s,9)
Feb 3 17:43:02 DEBUG[3179] pbx.c: Expression result is ‘0’
Feb 3 17:43:02 VERBOSE[3179] logger.c: – Executing GotoIf(“SIP/192.168.0.5-08ca9ae0”, “0?10:12”) in new stack
Feb 3 17:43:02 VERBOSE[3179] logger.c: – Goto (from-pstn-reghours,s,12)
Feb 3 17:43:02 DEBUG[3179] pbx.c: Expression result is ‘0’
Feb 3 17:43:02 VERBOSE[3179] logger.c: – Executing GotoIf(“SIP/192.168.0.5-08ca9ae0”, “0?13:15”) in new stack
Feb 3 17:43:02 VERBOSE[3179] logger.c: – Goto (from-pstn-reghours,s,15)
Feb 3 17:43:02 VERBOSE[3179] logger.c: – Executing Goto(“SIP/192.168.0.5-08ca9ae0”, “aa_1|s|1”) in new stack
Feb 3 17:43:02 VERBOSE[3179] logger.c: – Goto (aa_1,s,1)
Feb 3 17:43:02 WARNING[3179] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_EQ, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:
= ANSWER
^
Feb 3 17:43:02 WARNING[3179] ast_expr2.fl: If you have questions, please refer to doc/README.variables in the asterisk source.
Feb 3 17:43:02 DEBUG[3179] pbx.c: Expression result is ‘0’
Feb 3 17:43:02 VERBOSE[3179] logger.c: – Executing GotoIf(“SIP/192.168.0.5-08ca9ae0”, “0?4”) in new stack
Feb 3 17:43:02 DEBUG[3179] pbx.c: Not taking any branch
Feb 3 17:43:02 VERBOSE[3179] logger.c: – Executing Answer(“SIP/192.168.0.5-08ca9ae0”, “”) in new stack
Feb 3 17:43:02 VERBOSE[3179] logger.c: – Executing Wait(“SIP/192.168.0.5-08ca9ae0”, “1”) in new stack
Feb 3 17:43:03 VERBOSE[3179] logger.c: – Executing SetVar(“SIP/192.168.0.5-08ca9ae0”, “LOOPED=1”) in new stack
Feb 3 17:43:03 DEBUG[3179] pbx.c: Expression result is ‘0’
Feb 3 17:43:03 VERBOSE[3179] logger.c: – Executing GotoIf(“SIP/192.168.0.5-08ca9ae0”, “0?hang|1”) in new stack
Feb 3 17:43:03 DEBUG[3179] pbx.c: Not taking any branch
Feb 3 17:43:03 VERBOSE[3179] logger.c: – Executing SetVar(“SIP/192.168.0.5-08ca9ae0”, “DIR-CONTEXT=default”) in new stack
Feb 3 17:43:03 VERBOSE[3179] logger.c: – Executing DigitTimeout(“SIP/192.168.0.5-08ca9ae0”, “3”) in new stack
Feb 3 17:43:03 VERBOSE[3179] logger.c: – Set Digit Timeout to 3
Feb 3 17:43:03 VERBOSE[3179] logger.c: – Executing ResponseTimeout(“SIP/192.168.0.5-08ca9ae0”, “7”) in new stack
Feb 3 17:43:03 VERBOSE[3179] logger.c: – Set Response Timeout to 7
Feb 3 17:43:03 VERBOSE[3179] logger.c: – Executing BackGround(“SIP/192.168.0.5-08ca9ae0”, “custom/aa_1”) in new stack
Feb 3 17:43:03 VERBOSE[3179] logger.c: – Playing ‘custom/aa_1’ (language ‘en’)
Feb 3 17:43:05 DEBUG[2399] chan_sip.c: Stopping retransmission on ‘7375a7d77a8fc6fa42e267f20d5f4f76@127.0.0.1’ of Request 163: Match Not Found
Feb 3 17:43:08 VERBOSE[3179] logger.c: == Spawn extension (aa_1, s, 9) exited non-zero on ‘SIP/192.168.0.5-08ca9ae0’
Feb 3 17:43:08 VERBOSE[3179] logger.c: – Executing Hangup(“SIP/192.168.0.5-08ca9ae0”, “”) in new stack
Feb 3 17:43:08 VERBOSE[3179] logger.c: == Spawn extension (aa_1, h, 1) exited non-zero on ‘SIP/192.168.0.5-08ca9ae0’
Feb 3 17:43:08 DEBUG[3179] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Feb 3 17:43:08 DEBUG[3179] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES (‘2006-02-03 17:43:01’,’“01” <01>’,‘01’,‘s’,‘aa_1’, ‘SIP/192.168.0.5-08ca9ae0’,’’,‘Hangup’,’’,7,7,‘ANSWERED’,3,’’)
Feb 3 17:43:08 DEBUG[3179] chan_sip.c: update_call_counter() - decrement call limit counter
Feb 3 17:43:08 DEBUG[3179] chan_sip.c: update_call_counter() - decrement call limit counter

Tom

Nevermind, I found it. I forgot to make the POTS user in the Vega the same as the extension number in *.

Next, on to the FXO Vega.

Tom

OK, are there any handy guides out there that give a breakdown of how to set-up a SIP multi port FXO device (I have a 8-port Vega 50 FXO) with *? Any guide, even if its for another vendors device may help. I have looked at the Sipura 3000 instructions and didn’t have much luck with those. Anyone?

Did you ever get this done? How do you like the vega products? I to will be using this device and am interested to hear you results.

How is the sound quality? PSTN quality?

How was the support?

Ease of setup?