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Linkus and S50

I have set up Linkus and 3 S20 systems with not problems, I have a S50 and set up the same way and when I call for the app to any desk phone I get a disconnect as soon as I pick up the handset. I am sending a log is some one can explain why I may be getting a disconnect. My linkus app is ext202  and Im dialing my desk set  and its is also 202 I started the log when the linkus app started and ended when I got the disconnect. Thanks

 

support@IPPBX:~$ asterisk -vvvvvvvvvvvvvr

Unable to open specified codec config file '/ysdisk/ysapps/pbxcenter/etc/codec.conf', using built-in defaults

Unable to disable core size resource limit: Operation not permitted

Asterisk 13.7.0, Copyright (C) 1999 - 2014, Digium, Inc. and others.

Created by Mark Spencer <markster@digium.com>

Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.

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.

=========================================================================

Connected to Asterisk 13.7.0 currently running on IPPBX (pid = 2466)

No entry for terminal type "xterm-256color";

using dumb terminal settings.

  == Parsing '/etc/asterisk/pjsip.conf': Found

  == Parsing '/etc/asterisk/pjsip_auth.conf': Found

 

  == Parsing '/etc/asterisk/pjsip_wizard.conf': Found

 

    -- Added contact 'sip:202@10.0.1.77:5077;linkus' to AOR '202' with expiration of 120 seconds

 

 Contact 202/sip:202@10.0.1.77:5077;linkus has been created

 

 Endpoint 202 is now Reachable

 

 Contact 202/sip:202@10.0.1.77:5077;linkus is now Reachable.  RTT: 5.829 msec

 

 Endpoint 202 is now Reachable

 

  == Using SIP RTP Audio TOS bits 184

 

  == Using SIP RTP Audio CoS mark 5

 

    -- Executing [202@DLPN_DialPlan202:1] Macro("PJSIP/202-00000036", "stdexten,202,DIALPARAM_OF_EXTEN202,{DIALOPTIONS}") in new stack

 

    -- Executing [s@macro-stdexten:1] Set("PJSIP/202-00000036", "IsFromOutside=0") in new stack

 

    -- Executing [s@macro-stdexten:2] Set("PJSIP/202-00000036", "PBXDIALNUM=202") in new stack

 

    -- Executing [s@macro-stdexten:3] GotoIf("PJSIP/202-00000036", "0?sys-dial,1") in new stack

 

    -- Executing [s@macro-stdexten:4] GotoIf("PJSIP/202-00000036", "0?set:noset") in new stack

 

    -- Goto (macro-stdexten,s,7)

 

    -- Executing [s@macro-stdexten:7] Set("PJSIP/202-00000036", "CHECKCALLERNUM=202") in new stack

 

    -- Executing [s@macro-stdexten:8] Macro("PJSIP/202-00000036", "realstexten,202,PJSIP/202,tTKkWwXx") in new stack

 

    -- Executing [s@macro-realstexten:1] Set("PJSIP/202-00000036", "CHECKCALLERNUM=202") in new stack

 

    -- Executing [s@macro-realstexten:2] Set("PJSIP/202-00000036", "CHECKREALCALLERNUM=202") in new stack

 

    -- Executing [s@macro-realstexten:3] Set("PJSIP/202-00000036", "TIMEOUT(absolute)=6000") in new stack

 

    -- Channel will hangup at 2018-05-18 05:57:19.853 UTC-8.

 

    -- Executing [s@macro-realstexten:4] GotoIf("PJSIP/202-00000036", "0>0?6:5") in new stack

 

    -- Goto (macro-realstexten,s,5)

 

    -- Executing [s@macro-realstexten:5] YsWalkContext("PJSIP/202-00000036", "whitelist-internal,202,1,202,internal") in new stack

 

    -- Executing [s@macro-realstexten:6] Set("PJSIP/202-00000036", "CALLERID(num)=202") in new stack

 

    -- Executing [s@macro-realstexten:7] GotoIf("PJSIP/202-00000036", "0?conf,1") in new stack

 

    -- Executing [s@macro-realstexten:8] GotoIf("PJSIP/202-00000036", "0>0?follow-me-always,1") in new stack

 

    -- Executing [s@macro-realstexten:9] GotoIf("PJSIP/202-00000036", "0>0?dnd:next") in new stack

 

    -- Goto (macro-realstexten,s,12)

 

    -- Executing [s@macro-realstexten:12] CktStdCall("PJSIP/202-00000036", "srtpfor,PJSIP/202,novalue") in new stack

 

    -- Executing [s@macro-realstexten:13] NoOp("PJSIP/202-00000036", "NO CLIP") in new stack

 

    -- Executing [s@macro-realstexten:14] Set("PJSIP/202-00000036", "CDR(extfield4)=10.0.1.77") in new stack

 

    -- Executing [s@macro-realstexten:15] GotoIf("PJSIP/202-00000036", "0?send-notify:judge-fork") in new stack

 

    -- Goto (macro-realstexten,s,17)

 

    -- Executing [s@macro-realstexten:17] GotoIf("PJSIP/202-00000036", "0?forking:dial") in new stack

 

    -- Goto (macro-realstexten,s,21)

 

    -- Executing [s@macro-realstexten:21] Dial("PJSIP/202-00000036", "PJSIP/202,30,tTKkWwXxb(local_add_diversion,add_diversion,1()),,0") in new stack

 

[2018-05-18 04:17:19] WARNING[27764][C-00000018]: rtp_engine.c:1327 ast_rtp_instance_early_bridge_make_compatible: Seeded SDP of 'PJSIP/202-00000037'[mode 0] with that of 'PJSIP/202-00000036'[mode 20]

 

    -- PJSIP/202-00000037 Internal Gosub(local_add_diversion,add_diversion,1) start

 

    -- Executing [add_diversion@local_add_diversion:1] GotoIf("PJSIP/202-00000037", "0?2:3") in new stack

 

    -- Goto (local_add_diversion,add_diversion,3)

    -- Executing [add_diversion@local_add_diversion:3] NoOp("PJSIP/202-00000037", "end add diversion") in new stack

 

    -- Executing [add_diversion@local_add_diversion:4] GotoIf("PJSIP/202-00000037", "?pjsip_add_header,1") in new stack

 

[2018-05-18 04:17:19] NOTICE[27764][C-00000018]: app_stack.c:1082 gosub_run: PJSIP/202-00000037 Abnormal 'Gosub(local_add_diversion,add_diversion,1)' exit.  Popping routine return locations.

 

    -- Called PJSIP/202/sip:202@10.0.1.94:5060

  == Using SIP RTP Audio TOS bits 184

  == Using SIP RTP Audio CoS mark 5

 

    -- PJSIP/202-00000037 is ringing

 

[2018-05-18 04:17:21] WARNING[10340]: channel.c:5506 set_format: Unable to find a codec translation path: (ulaw) -> (ilbc)

[2018-05-18 04:17:21] WARNING[10340]: channel.c:5506 set_format: Unable to find a codec translation path: (ilbc) -> (ulaw)

 

    -- PJSIP/202-00000037 answered PJSIP/202-00000036

 

    -- Channel PJSIP/202-00000037 joined 'simple_bridge' basic-bridge <c6d38992-bd85-49bf-9b65-d50ab539ffd0>

 

    -- Channel PJSIP/202-00000036 joined 'simple_bridge' basic-bridge <c6d38992-bd85-49bf-9b65-d50ab539ffd0>

 

       > 0x723aa648 -- Probation passed - setting RTP source address to 10.0.1.94:12102

 

[2018-05-18 04:17:21] WARNING[27764][C-00000018]: channel.c:5506 set_format: Unable to find a codec translation path: (ulaw) -> (ilbc)

 

    -- Channel PJSIP/202-00000036 left 'simple_bridge' basic-bridge <c6d38992-bd85-49bf-9b65-d50ab539ffd0>

 

    -- Channel PJSIP/202-00000037 left 'simple_bridge' basic-bridge <c6d38992-bd85-49bf-9b65-d50ab539ffd0>

[2018-05-18 04:17:21] WARNING[27801][C-00000018]: channel.c:5506 set_format: Unable to find a codec translation path: (ulaw) -> (ilbc)

 

  == Spawn extension (macro-realstexten, s, 21) exited non-zero on 'PJSIP/202-00000036' in macro 'realstexten'

 

  == Spawn extension (macro-stdexten, s, 8) exited non-zero on 'PJSIP/202-00000036' in macro 'stdexten'

 

  == Spawn extension (DLPN_DialPlan202, 202, 1) exited non-zero on 'PJSIP/202-00000036'

 

    -- Executing [h@DLPN_DialPlan202:1] Hangup("PJSIP/202-00000036", "") in new stack

 

  == Spawn extension (DLPN_DialPlan202, h, 1) exited non-zero on 'PJSIP/202-00000036'

 

  == Parsing '/etc/asterisk/pjsip.conf': Found

  == Parsing '/etc/asterisk/pjsip_auth.conf': Found

 

  == Parsing '/etc/asterisk/pjsip_wizard.conf': Found

 

[2018-05-18 04:17:40] WARNING[21295]: res_pjsip_outbound_registration.c:966 handle_registration_response: Maximum retries reached when attempting outbound registration to 'sip:voip.sotelsystems.com' with client 'sip:4703176671@voip.sotelsystems.com', stopping registration attempt

 

IPPBX*CLI> 

 

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