H323 Disconnects after 10+ minutes

Discussion in 'General' started by hinzinho, Dec 7, 2009.

  1. hinzinho

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    I am using H323 to talk between Asterisk and Avaya IP Office 500. For some strange reason, when we are talking on a VoIP call, we get disconnected after 10+ minutes. We have two other Elastix box, but none of them are getting disconnected.

    Attached is the log file from the H323. The disconnects happen around 9:46AM and 9:58AM.

    My OOH323.CONF
    Code:
    [general]
    port=1720
    bindaddr=
    gateway=no
    faststart=yes
    h245tunneling=yes
    h323id=ObjSysAsterisk
    e164=100
    callerid=asterisk
    gatekeeper = DISABLE
    context=default
    disallow=all     ;Note order of disallow/allow is important.
    allow=ulaw
    dtmfmode=rfc2833
    progress_setup = 8
    progress_alert = 8
    
    [remoteloc1]
    type=friend
    port=1720
    ip=
    context=from-internal
    disallow=all
    allow=ulaw
    rtptimeout=90
    
    Thank you! http://forum.elastix.org/old_files/H323_log.zip
     
  2. hinzinho

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    I got a few more logs from the Asterisk. From what I can tell, the cause is "condition 20 on ooh323". Any suggestions as to the cause?

    Code:
    Dec 29 10:25:01 	VERBOSE 	[15027] logger.c: 	-- Remote UNIX connection
    Dec 29 10:25:01 	VERBOSE 	[31438] logger.c: 	-- Remote UNIX connection disconnected
    Dec 29 10:26:01 	WARNING 	[31413] chan_ooh323.c: Don't know how to indicate condition 20 on ooh323c_9
    Dec 29 14:42:06  	VERBOSE  	[349] logger.c:  	-- SIP/5034-1b1aa680 is ringing
    Dec 29 14:42:09 	VERBOSE 	[349] logger.c: 	-- SIP/5034-1b1aa680 answered OOH323/denver-eaf3
    Dec 29 14:42:09 	WARNING 	[349] chan_ooh323.c:  Don't know how to indicate condition 20 on ooh323c_18
    Dec 29 15:02:55 VERBOSE [410] logger.c: -- Remote UNIX connection disconnected
    Dec 29 15:04:01 WARNING [349] chan_ooh323.c: Don't know how to indicate condition 20 on ooh323c_18
    Dec 29 15:04:01 VERBOSE [349] logger.c: -- Executing [h@macro-dial:1] Macro("OOH323/denver-eaf3", "hangupcall") in new stack
    Dec 29 15:04:01 VERBOSE [349] logger.c: -- Executing [s@macro-hangupcall:1] ResetCDR("OOH323/denver-eaf3", "w") in new stack
    Dec 29 15:04:01 DEBUG [349] app_macro.c: Executed application: ResetCDR
    Dec 29 15:04:01 VERBOSE [349] logger.c: -- Executing [s@macro-hangupcall:2] NoCDR("OOH323/denver-eaf3", "") in new stack
    Dec 29 15:04:01 DEBUG [349] app_macro.c: Executed application: NoCDR
    Dec 29 15:04:01 VERBOSE [349] logger.c: -- Executing [s@macro-hangupcall:3] GotoIf("OOH323/denver-eaf3", "1?skiprg") in new stack
    Dec 29 15:04:01 VERBOSE [349] logger.c: -- Goto (macro-hangupcall,s,6)
    Dec 29 15:04:01 DEBUG [349] app_macro.c: Executed application: GotoIf
    Dec 29 15:04:01 VERBOSE [349] logger.c: -- Executing [s@macro-hangupcall:6] GotoIf("OOH323/denver-eaf3", "1?skipblkvm") in new stack
    Dec 29 15:04:01 VERBOSE [349] logger.c: -- Goto (macro-hangupcall,s,9)
    Dec 29 15:04:01 DEBUG [349] app_macro.c: Executed application: GotoIf
    Dec 29 15:04:01 VERBOSE [349] logger.c: -- Executing [s@macro-hangupcall:9] GotoIf("OOH323/denver-eaf3", "1?theend") in new stack
    Dec 29 15:04:01 VERBOSE [349] logger.c: -- Goto (macro-hangupcall,s,11)
    Dec 29 15:04:01 DEBUG [349] app_macro.c: Executed application: GotoIf
    Dec 29 15:04:01 VERBOSE [349] logger.c: -- Executing [s@macro-hangupcall:11] Hangup("OOH323/denver-eaf3", "") in new stack
    Dec 29 15:04:01 VERBOSE [349] logger.c: == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'OOH323/denver-eaf3' in macro 'hangupcall'
    Dec 29 15:04:01 VERBOSE [349] logger.c: == Spawn h extension (macro-dial, h, 1) exited non-zero on 'OOH323/denver-eaf3'
    
    CLI > core show channeltypes
    Code:
    Type        Description                              Devicestate  Indications  Transfer    
    ----------  -----------                              -----------  -----------  --------    
    SIP         Session Initiation Protocol (SIP)        yes          yes          yes         
    OOH323      Objective Systems H323 Channel Driver    no           yes          no          
    DAHDI       DAHDI Telephony Driver w/PRI w/OPENR2    no           yes          no          
    MGCP        Media Gateway Control Protocol (MGCP)    yes          yes          no          
    WOOMERA     Woomera Channel Driver                   no           yes          yes         
    Agent       Call Agent Proxy Channel                 yes          yes          no          
    IAX2        Inter Asterisk eXchange Driver (Ver 2)   yes          yes          yes         
    Local       Local Proxy Channel Driver               yes          yes          no          
    Phone       Standard Linux Telephony API Driver      no           yes          no          
    ----------
    9 channel drivers registered.
    

    CLI> core show channeltype ooh323
    Code:
    -- Info about channel driver: OOH323 --
      Device State: no
        Indication: yes
         Transfer : no
      Capabilities: -1
       Digit Begin: yes
         Digit End: yes
        Send HTML : no
     Image Support: no
      Text Support: no
    
     
  3. hugo_cba

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    Re: Re:H323 Disconnects after 10+ minutes

    Hi!

    Can you solve this problem?

    I´m actually trying to do this scenario with an Avaya IP Office (400 I think) but i´m trying to get the prog manuals.

    It was problematic the configuration betwen the Elastix and Avaya?

    Best Regards

    P.D: Sorry for my poor english.
     
  4. hinzinho

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    Re: Re:H323 Disconnects after 10+ minutes

    We got it working between IP Office 400s and IP Office 500s. Let me know where you are stuck or need help on.

    As for the disconnects after 10+ minutes.. the problem was related to Cisco router from our ISP.
     
  5. Sagar Shah

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    Hey,

    I got a problem with integrating IPoffice 500 and elastix 2.0.
    I can make outgoing calls easily with the pennytel account i have setup. It is anyways not working with IPoffice 500.

    When i dial an extension of avaya, I get a strange sound and below is the process that happens:-

    Code:
    [root@Elastix ~]# asterisk -rvvvv
    Asterisk 1.6.2.13, Copyright (C) 1999 - 2010 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.
    =========================================================================
      == Parsing '/etc/asterisk/asterisk.conf':   == Found
      == Parsing '/etc/asterisk/extconfig.conf':   == Found
    Connected to Asterisk 1.6.2.13 currently running on Elastix (pid = 10318)
    Verbosity is at least 4
      == Using SIP RTP TOS bits 184
      == Using SIP RTP CoS mark 5
      == Using SIP VRTP TOS bits 136
      == Using SIP VRTP CoS mark 6
        -- Executing [666@from-internal:1] Set("SIP/6000-00000008", "FAX_RX_EMAIL=fax@mydomain.com") in new stack
        -- Executing [666@from-internal:2] Goto("SIP/6000-00000008", "ext-fax,s,1") in new stack
        -- Goto (ext-fax,s,1)
        -- Executing [s@ext-fax:1] Macro("SIP/6000-00000008", "user-callerid,") in new stack
        -- Executing [s@macro-user-callerid:1] Set("SIP/6000-00000008", "AMPUSER=6000") in new stack
        -- Executing [s@macro-user-callerid:2] GotoIf("SIP/6000-00000008", "0?report") in new stack
        -- Executing [s@macro-user-callerid:3] ExecIf("SIP/6000-00000008", "1?Set(REALCALLERIDNUM=6000)") in new stack
        -- Executing [s@macro-user-callerid:4] Set("SIP/6000-00000008", "AMPUSER=6000") in new stack
        -- Executing [s@macro-user-callerid:5] Set("SIP/6000-00000008", "AMPUSERCIDNAME=Sagar") in new stack
        -- Executing [s@macro-user-callerid:6] GotoIf("SIP/6000-00000008", "0?report") in new stack
        -- Executing [s@macro-user-callerid:7] Set("SIP/6000-00000008", "AMPUSERCID=6000") in new stack
        -- Executing [s@macro-user-callerid:8] Set("SIP/6000-00000008", "CALLERID(all)="Sagar" <6000>") in new stack
        -- Executing [s@macro-user-callerid:9] ExecIf("SIP/6000-00000008", "0?Set(CHANNEL(language)=)") in new stack
        -- Executing [s@macro-user-callerid:10] GotoIf("SIP/6000-00000008", "0?continue") in new stack
        -- Executing [s@macro-user-callerid:11] Set("SIP/6000-00000008", "__TTL=64") in new stack
        -- Executing [s@macro-user-callerid:12] GotoIf("SIP/6000-00000008", "1?continue") in new stack
        -- Goto (macro-user-callerid,s,19)
        -- Executing [s@macro-user-callerid:19] NoOp("SIP/6000-00000008", "Using CallerID "Sagar" <6000>") in new stack
        -- Executing [s@ext-fax:2] NoOp("SIP/6000-00000008", "Receiving Fax for: fax@mydomain.com , From: "Sagar" <6000>") in new stack
        -- Executing [s@ext-fax:3] StopPlayTones("SIP/6000-00000008", "") in new stack
        -- Executing [s@ext-fax:4] ReceiveFAX("SIP/6000-00000008", "/var/spool/asterisk/fax/1303045112.8.tif") in new stack
        -- Auto fallthrough, channel 'SIP/6000-00000008' status is 'UNKNOWN'
        -- Executing [h@ext-fax:1] GotoIf("SIP/6000-00000008", "1?failed") in new stack
        -- Goto (ext-fax,h,103)
        -- Executing [h@ext-fax:103] NoOp("SIP/6000-00000008", "FAX FAILED for: fax@mydomain.com , From: "Sagar" <6000>") in new stack
        -- Executing [h@ext-fax:104] Macro("SIP/6000-00000008", "hangupcall,") in new stack
        -- Executing [s@macro-hangupcall:1] GotoIf("SIP/6000-00000008", "1?noautomon") in new stack
        -- Goto (macro-hangupcall,s,3)
        -- Executing [s@macro-hangupcall:3] NoOp("SIP/6000-00000008", "TOUCH_MONITOR_OUTPUT=") in new stack
        -- Executing [s@macro-hangupcall:4] GotoIf("SIP/6000-00000008", "1?skiprg") in new stack
        -- Goto (macro-hangupcall,s,7)
        -- Executing [s@macro-hangupcall:7] GotoIf("SIP/6000-00000008", "1?skipblkvm") in new stack
        -- Goto (macro-hangupcall,s,10)
        -- Executing [s@macro-hangupcall:10] GotoIf("SIP/6000-00000008", "1?theend") in new stack
        -- Goto (macro-hangupcall,s,12)
        -- Executing [s@macro-hangupcall:12] Hangup("SIP/6000-00000008", "") in new stack
      == Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/6000-00000008' in macro 'hangupcall'
      == Spawn extension (ext-fax, h, 104) exited non-zero on 'SIP/6000-00000008'
    Elastix*CLI>
    
    
    
    
    Elastix*CLI> core show channeltypes
    Type        Description                              Devicestate  Indications  Transfer
    ----------  -----------                              -----------  -----------  --------
    IAX2        Inter Asterisk eXchange Driver (Ver 2)   yes          yes          yes
    Agent       Call Agent Proxy Channel                 yes          yes          no
    Bridge      Bridge Interaction Channel               no           no           no
    WOOMERA     Woomera Channel Driver                   no           yes          yes
    Phone       Standard Linux Telephony API Driver      no           yes          no
    MGCP        Media Gateway Control Protocol (MGCP)    yes          yes          no
    USTM        UNISTIM Channel Driver                   no           yes          no
    OOH323      Objective Systems H323 Channel Driver    no           yes          no
    DAHDI       DAHDI Telephony Driver w/PRI & MFC/R2    no           yes          no
    SIP         Session Initiation Protocol (SIP)        yes          yes          yes
    Local       Local Proxy Channel Driver               yes          yes          no
    ----------
    11 channel drivers registered.
    
    
    
    
    
    Elastix*CLI> core show channeltype ooh323
    -- Info about channel driver: OOH323 --
      Device State: no
        Indication: yes
         Transfer : no
      Capabilities: -1
       Digit Begin: yes
         Digit End: yes
        Send HTML : no
     Image Support: no
      Text Support: no
    
    
    
    Someone please help!!
     
  6. hinzinho

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    How are you connecting to IPOffice 500? Sip or H323? Post your trunk configuration. Are you able to make calls from the Avaya to Asterisk?
     
  7. Sagar Shah

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    Thanks a lot for your reply.

    The avaya machine (Ipoffice 500) is connected to a linksys switch and the elastix machine is also connected to the same switch with a leased line.

    I am not able to make calls from avaya as well.

    the trunk configuration is as follows:-

    Code:
    
    [general]
    port=1720
    bindaddr=10.2.1.253
    gateway=no
    faststart=yes
    h245tunneling=yes
    h323id=ObjSysAsterisk
    e164=100
    callerid=asterisk
    gatekeeper = DISABLE
    context=default
    disallow=all ;Note order of disallow/allow is important.
    allow=g729
    allow=gsm
    allow=ulaw
    dtmfmode=rfc2833
    progress_setup = 8
    progress_alert = 8
    dtmfmode=rfc2833
    
    [AvayaPBX]
    type=friend
    ip=10.2.1.250
    port=1720
    context=from-internal
    disallow=all
    allow=ulaw
    rtptimeout=60
    e164=50
    
    Regards,
    Sagar
     

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