Topic: SAIL 3.2.0-14 and Trunks ?

[fpausp]

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:

Code: [Select]

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

[SARK devs]

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.

Code: [Select]

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

Code: [Select]

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 

[fpausp]

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

[fpausp]

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

[fpausp]

I got problems with the SPA3102, the outbound-connection goes down after a view seconds (~30).

Code: [Select]

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

[SARK devs]

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