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« on: March 24, 2013, 04:51:20 pm »
install 1004, connect some supported phones and wait for the phones to be recognized and get an extension assigned.
Calling from phone to phone will work then. Now add a phone line with the wizard, and fille in the blanks. Protocol SPA. username and phonenumber the same.
click 'Save'. Nothing seems to happen, this is becuase the apache session is killed. restarting apache brings you back and the phone line is created.
Try making incoming call. Call reaches the core, and 'The number you have dialled is not in service' is played.
Made ring all phones setting for all states of the house: still the same.
Checked on outgoing calls. No luck, error 503. Asterisk cannot find an outbound route.
checked the mysql db:
mysql> select * from extensions where context like 'outbound-allroutes';
+-------+--------------------+-------+----------+-------+----------------------------+
| id | context | exten | priority | app | appdata |
+-------+--------------------+-------+----------+-------+----------------------------+
| 12240 | outbound-allroutes | 112 | 1 | Macro | dialout-trunk,/,${EXTEN},, |
| 12241 | outbound-allroutes | 112 | 2 | Macro | outisbusy, |
| 12242 | outbound-allroutes | 911 | 1 | Macro | dialout-trunk,/,${EXTEN},, |
| 12243 | outbound-allroutes | 911 | 2 | Macro | outisbusy, |
+-------+--------------------+-------+----------+-------+----------------------------+
No outbound routes for anything else than 911 and 112. ( and even those don't work)
So would the trunk and SPA3102 then be known at all to Asterisk? Check the db:
mysql> select * from phonelines
-> ;
+----+---------+-------+--------+----------+------+---------+------+-------------+------------+----------+---------+----------+--------------+
| id | enabled | isfax | prefix | protocol | name | host | port | phonenumber | username | password | faxmail | channels | faxheader |
+----+---------+-------+--------+----------+------+---------+------+-------------+------------+----------+---------+----------+--------------+
| 1 | yes | no | 0 | SPA | | dynamic | 5060 | 030879XXXX | 030879XXXX| linuxmce | 2 | 5 | LinuxMCE fax |
+----+---------+-------+--------+----------+------+---------+------+-------------+------------+----------+---------+----------+--------------+
1 row in set (0.00 sec)
So the line is there... Had already changed the port in the SPA to 5060 to match LMCE.
The SPA tries to register with the core, but is rejected becuase the user is not known. May be the SPA should not at all register with the Core when used as a FXO ( phone line ) device?
Then on inbound calls, Asterisk really tries to deliver the call, but to no avail:
<------------>
-- Executing [0308793512@from-sip-external:1] NoOp("SIP/10.1.100.1-00000009", "Received incoming SIP connection from unknown peer to 0308793512") in new stack
-- Executing [0308793512@from-sip-external:2] Set("SIP/10.1.100.1-00000009", "DID=0308793512") in new stack
-- Executing [0308793512@from-sip-external:3] Goto("SIP/10.1.100.1-00000009", "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/10.1.100.1-00000009", "0?from-trunk,0308793512,1") in new stack
-- Executing [s@from-sip-external:2] Set("SIP/10.1.100.1-00000009", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2013-03-24 15:25:01.809 CET.
-- Executing [s@from-sip-external:3] Answer("SIP/10.1.100.1-00000009", "") in new stack
Audio is at 14602
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
-- Executing [s@from-sip-external:4] Wait("SIP/10.1.100.1-00000009", "2") in new stack
-- Executing [s@from-sip-external:5] Playback("SIP/10.1.100.1-00000009", "ss-noservice") in new stack
-- <SIP/10.1.100.1-00000009> Playing 'ss-noservice.gsm' (language 'en')
-- Executing [s@from-sip-external:6] PlayTones("SIP/10.1.100.1-00000009", "congestion") in new stack
-- Executing [s@from-sip-external:7] Congestion("SIP/10.1.100.1-00000009", "5") in new stack
== Spawn extension (from-sip-external, s, 7) exited non-zero on 'SIP/10.1.100.1-00000009'
-- Executing [h@from-sip-external:1] NoOp("SIP/10.1.100.1-00000009", "Hangup") in new stack
Scheduling destruction of SIP dialog '837f2e5f-800dbcd5@10.1.100.6' in 32000 ms (Method: ACK)
Has anybody got this working? And if so how?
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