H323 Disconnects after 10+ minutes

hinzinho

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#1
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
 

hinzinho

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#2
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
 

hugo_cba

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#3
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.
 

hinzinho

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#4
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.
 

Sagar Shah

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#5
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!!
 

hinzinho

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#6
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?
 

Sagar Shah

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#7
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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