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Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: fpausp on July 16, 2012, 10:20:51 PM
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Hi,
I have sail 3.2.0-14 running and like to use three different trunks:
1. SIP-Trunk (fairytel.at)
2. SPA3102
3. Portech MV-370 GSM-GW
Please take a look on my logs:
All Calls have been done with a SoftClient (Yate) and my mobile-phone (A1 - 0680XXXXXXX)
# Fairytel SIP-Trunk - Outbound Call - The Phone (Mobile) is ringing
sme8*CLI>
== Using SIP RTP CoS mark 5
-- Executing [0680XXXXXXX@internal:1] AGI("SIP/401-0000000d", "sarkhpe,OutCos,0680XXXXXXX,,") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
-- <SIP/401-0000000d>AGI Script sarkhpe completed, returning 0
-- Executing [0680XXXXXXX@401opencos:1] AGI("SIP/401-0000000d", "sarkhpe,OutCluster,0680XXXXXXX,,") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
-- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=default)
-- <SIP/401-0000000d>AGI Script sarkhpe completed, returning 0
-- Executing [0680XXXXXXX@qrxvtmny:1] AGI("SIP/401-0000000d", "sarkhpe,OutRoute,fairytel,,") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
-- AGI Script Executing Application: (Set) Options: (__filename=1342467393-0680XXXXXXX-401.wav)
-- AGI Script Executing Application: (MixMonitor) Options: (/var/spool/asterisk/monitor/1342467393-0680XXXXXXX-401.wav,,/bin/sh /opt/sark/scripts/selmix mixmon /var/spool/asterisk/monitor/1342467393-0680XXXXXXX-401.wav)
== Begin MixMonitor Recording SIP/401-0000000d
-- AGI Script Executing Application: (Set) Options: (TIMEOUT(absolute)=14400)
Channel will hangup at 2012-07-17 01:36:33.106 CEST.
-- AGI Script Executing Application: (Dial) Options: (SIP/0680XXXXXXX@peer3290,,T)
== Using SIP RTP CoS mark 5
-- Called SIP/0680XXXXXXX@peer3290
-- SIP/peer3290-0000000e is making progress passing it to SIP/401-0000000d
[Jul 16 21:36:33] WARNING[10687]: res_rtp_asterisk.c:2041 ast_rtp_read: RTP Read too short
-- SIP/peer3290-0000000e is making progress passing it to SIP/401-0000000d
-- <SIP/401-0000000d>AGI Script sarkhpe completed, returning 4
== Spawn extension (qrxvtmny, 0680XXXXXXX, 1) exited non-zero on 'SIP/401-0000000d'
-- Executing [h@qrxvtmny:1] Hangup("SIP/401-0000000d", "") in new stack
== Spawn extension (qrxvtmny, h, 1) exited non-zero on 'SIP/401-0000000d'
== MixMonitor close filestream
== Executing [/bin/sh /opt/sark/scripts/selmix mixmon /var/spool/asterisk/monitor/1342467393-0680XXXXXXX-401.wav]
== End MixMonitor Recording SIP/401-0000000d
-- Got SIP response 500 "I'm terribly sorry, server error occurred (1/SL)" back from 213.185.165.114:5060
-- Remote UNIX connection
-- Remote UNIX connection disconnected
[Jul 16 21:37:49] NOTICE[3092]: chan_sip.c:20192 handle_response_peerpoke: Peer 'peer3290' is now Lagged. (3020ms / 3000ms)
[Jul 16 21:37:59] NOTICE[3092]: chan_sip.c:20192 handle_response_peerpoke: Peer 'peer3290' is now Reachable. (21ms / 3000ms
# Fairytel SIP-Trunk - Inbound Call - There is a Voice-message who tells me the number does not exist ...
# Portech MV370 GSM-GW - Outbound Call - The Phone (Mobile) is ringing
sme8*CLI>
== Using SIP RTP CoS mark 5
-- Executing [0680XXXXXXX@internal:1] AGI("SIP/401-0000000f", "sarkhpe,OutCos,0680XXXXXXX,,") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
-- <SIP/401-0000000f>AGI Script sarkhpe completed, returning 0
-- Executing [0680XXXXXXX@401opencos:1] AGI("SIP/401-0000000f", "sarkhpe,OutCluster,0680XXXXXXX,,") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
-- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=default)
-- <SIP/401-0000000f>AGI Script sarkhpe completed, returning 0
-- Executing [0680XXXXXXX@qrxvtmny:1] AGI("SIP/401-0000000f", "sarkhpe,OutRoute,portech,,") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
-- AGI Script Executing Application: (Set) Options: (__filename=1342467611-0680XXXXXXX-401.wav)
-- AGI Script Executing Application: (MixMonitor) Options: (/var/spool/asterisk/monitor/1342467611-0680XXXXXXX-401.wav,,/bin/sh /opt/sark/scripts/selmix mixmon /var/spool/asterisk/monitor/1342467611-0680XXXXXXX-401.wav)
== Begin MixMonitor Recording SIP/401-0000000f
-- AGI Script Executing Application: (Set) Options: (TIMEOUT(absolute)=14400)
Channel will hangup at 2012-07-17 01:40:11.984 CEST.
-- AGI Script Executing Application: (Dial) Options: (SIP/0680XXXXXXX@peer3372,,T)
== Using SIP RTP CoS mark 5
-- Called SIP/0680XXXXXXX@peer3372
-- SIP/peer3372-00000010 is ringing
-- <SIP/401-0000000f>AGI Script sarkhpe completed, returning 4
== Spawn extension (qrxvtmny, 0680XXXXXXX, 1) exited non-zero on 'SIP/401-0000000f'
-- Executing [h@qrxvtmny:1] Hangup("SIP/401-0000000f", "") in new stack
== Spawn extension (qrxvtmny, h, 1) exited non-zero on 'SIP/401-0000000f'
== Executing [/bin/sh /opt/sark/scripts/selmix mixmon /var/spool/asterisk/monitor/1342467611-0680XXXXXXX-401.wav]
== End MixMonitor Recording SIP/401-0000000f
# Portech MV370 GSM-GW - Inbound Call - The Extension 401 is not ringing
sme8*CLI>
== Using SIP RTP CoS mark 5
[Jul 16 21:56:59] NOTICE[3092]: chan_sip.c:22081 handle_request_invite: Call from 'peer3372' (192.168.XXX.XXX:5060) to extension '401' rejected because extension not found in context 'mainmenu'.
# SPA3102 - Outbound Call - The Phone (Mobile) is ringing
sme8*CLI>
== Using SIP RTP CoS mark 5
-- Executing [0680XXXXXXX@internal:1] AGI("SIP/401-00000013", "sarkhpe,OutCos,0680XXXXXXX,,") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
-- <SIP/401-00000013>AGI Script sarkhpe completed, returning 0
-- Executing [0680XXXXXXX@401opencos:1] AGI("SIP/401-00000013", "sarkhpe,OutCluster,0680XXXXXXX,,") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
-- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=default)
-- <SIP/401-00000013>AGI Script sarkhpe completed, returning 0
-- Executing [0680XXXXXXX@qrxvtmny:1] AGI("SIP/401-00000013", "sarkhpe,OutRoute,spa3102,,") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
-- AGI Script Executing Application: (Set) Options: (__filename=1342467950-0680XXXXXXX-401.wav)
-- AGI Script Executing Application: (MixMonitor) Options: (/var/spool/asterisk/monitor/1342467950-0680XXXXXXX-401.wav,,/bin/sh /opt/sark/scripts/selmix mixmon /var/spool/asterisk/monitor/1342467950-0680XXXXXXX-401.wav)
== Begin MixMonitor Recording SIP/401-00000013
-- AGI Script Executing Application: (Set) Options: (TIMEOUT(absolute)=14400)
Channel will hangup at 2012-07-17 01:45:50.523 CEST.
-- AGI Script Executing Application: (Dial) Options: (SIP/0680XXXXXXX@peer1787,,T)
== Using SIP RTP CoS mark 5
-- Called SIP/0680XXXXXXX@peer1787
-- SIP/peer1787-00000014 is ringing
-- SIP/peer1787-00000014 answered SIP/401-00000013
[Jul 16 21:45:50] WARNING[11119]: res_rtp_asterisk.c:2041 ast_rtp_read: RTP Read too short
-- Executing [h@qrxvtmny:1] Hangup("SIP/401-00000013", "") in new stack
== Spawn extension (qrxvtmny, h, 1) exited non-zero on 'SIP/401-00000013'
-- <SIP/401-00000013>AGI Script sarkhpe completed, returning 4
== Spawn extension (qrxvtmny, 0680XXXXXXX, 1) exited non-zero on 'SIP/401-00000013'
== MixMonitor close filestream
== Executing [/bin/sh /opt/sark/scripts/selmix mixmon /var/spool/asterisk/monitor/1342467950-0680XXXXXXX-401.wav]
== End MixMonitor Recording SIP/401-00000013
-- Remote UNIX connection
-- Remote UNIX connection disconnected
# SPA3102 - Inbound Call - The Extension 401 is not ringing
sme8*CLI>
== Using SIP RTP CoS mark 5
[Jul 16 21:46:43] NOTICE[3092]: chan_sip.c:22081 handle_request_invite: Call from 'peer1787' (192.168.XXX.XXX:5060) to extension '192.168.XXX.XXX:5060' rejected because extension not found in context 'mainmenu'.
Is there a global way for the recect problem and shell we do that globally ?
I got no log for the incomming call on the sip-trunk ...
regards
fpausp
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Hi
for the first two, you are getting a 500 from the far end server after an rtp error. This usually indicates something wrong with your comms gear (or theirs), check your router and ethernet settings.
res_rtp_asterisk.c:2041 ast_rtp_read: RTP Read too short
...
Got SIP response 500 "I'm terribly sorry, server error occurred (1/SL)" back from 213.185.165.114:5060
the third instance 'Portech MV370 GSM-GW - Outbound Call' is less clear. Run the call again but do the following at the asterisk console first
agi debug
core set global DEBUG ON
post the output
In the last case 'Portech MV370 GSM-GW - Inbound Call' you are attempting to call extension 401 on the inbound side of the system (mainmenu); extensions are defined on the other side(internal). You can fix this by either creating a DDI of 401 or some other number and setting you GSM device to send to that.
Finally, I didnot understand your last question about globals - can you please re-state?
Kind Regards
S
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Hi,
Thanks for your reply, I think I have it all running, I need more time to test it ...
Just one Problem/Bug with the trunks, they overwrite themself with the data of the first trunk, I will do more tests to tomorrow...
regards
fpausp
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Hi,
Everything works great, just the inbound calls of the sip-trunk (fairytel.at) are not working but I guess it is the provider ...
Thanks a lot for your help !
Best
fpausp
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I got problems with the SPA3102, the outbound-connection goes down after a view seconds (~30).
sme8*CLI>
== Using SIP RTP CoS mark 5
-- Executing [02631XXXXX@internal:1] AGI("SIP/5000-00000076", "sarkhpe,OutCos,02631XXXXX,,") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
-- <SIP/5000-00000076>AGI Script sarkhpe completed, returning 0
-- Executing [02631XXXXX@5000opencos:1] AGI("SIP/5000-00000076", "sarkhpe,OutCluster,02631XXXXX,,") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
-- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=default)
-- <SIP/5000-00000076>AGI Script sarkhpe completed, returning 0
-- Executing [02631XXXXX@qrxvtmny:1] AGI("SIP/5000-00000076", "sarkhpe,OutRoute,fairytel,,") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
-- AGI Script Executing Application: (Set) Options: (__filename=1342959878-02631XXXXX-5000.wav)
-- AGI Script Executing Application: (MixMonitor) Options: (/var/spool/asterisk/monitor/1342959878-02631XXXXX-5000.wav,,/bin/sh /opt/sark/scripts/selmix mixmon /var/spool/asterisk/monitor/1342959878-02631XXXXX-5000.wav)
== Begin MixMonitor Recording SIP/5000-00000076
-- AGI Script Executing Application: (Set) Options: (TIMEOUT(absolute)=14400)
Channel will hangup at 2012-07-22 18:24:38.898 CEST.
-- AGI Script Executing Application: (Dial) Options: (SIP/02631XXXXX@peer1636,,T)
== Using SIP RTP CoS mark 5
-- Called SIP/02631XXXXX@peer1636
-- SIP/peer1636-00000077 is making progress passing it to SIP/5000-00000076
-- SIP/peer1636-00000077 is making progress passing it to SIP/5000-00000076
-- SIP/peer1636-00000077 answered SIP/5000-00000076
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Executing [h@qrxvtmny:1] Hangup("SIP/5000-00000076", "") in new stack
== Spawn extension (qrxvtmny, h, 1) exited non-zero on 'SIP/5000-00000076'
-- <SIP/5000-00000076>AGI Script sarkhpe completed, returning 4
== Spawn extension (qrxvtmny, 02631XXXXX, 1) exited non-zero on 'SIP/5000-00000076'
== MixMonitor close filestream
== Executing [/bin/sh /opt/sark/scripts/selmix mixmon /var/spool/asterisk/monitor/1342959878-02631XXXXX-5000.wav]
== End MixMonitor Recording SIP/5000-00000076
Please tell me how I can get a detailed log/debug from spa3102 ?
Best
fpausp
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If it is closing at around 30 seconds this is usually (but not always) a sign that a firewall pinhole is closing. Check your firewall settings between the spa and whatever it is talking to (your asterisk box I guess).
Kind REgards
S