[asterisk-users] Fwd: Back to back E1 - asterisk <=> toshiba pbx -Call droping

Thu Mar 8 07:16:07 MST 2007

Before studying your configs, what have you tried so far?

Did you change this?  

Also, try changing your second span timing from span=2,2,0,ccs,hdb3,crc4
to span=2,0,0,ccs,hdb3,crc4.

Here is the documentation on voip-info for why it may be the cause of
your issues

http://www.voip-info.org/wiki/view/Zaptel.conf+span+syntax

span definition format: 
span=(spannum),(timing),(LBO),(framing),(coding) 

spannum= Number of the span. 

timing= How to synchronize the timing devices. 
0: to not use this span as sync source 
1: to use as primary sync source 
2: to set as secondary and so forth 

Use '1' if you want to use the circuit as your primary sync source. If
'0' is used asterisk will try to provide timing to the span (say, if you
were connecting to a legacy PBX). If Asterisk is connected directly to
the telco you will want to use '1' to accept timing from them. If
youhave multiple spans, set them as 2, 3, 4, etc. 

Problems with timing manifest themselves different ways - with static,
pops, and channels or calls regularly dropping.

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Vidura
Senadeera
Sent: Thursday, March 08, 2007 1:01 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Fwd: Back to back E1 - asterisk <=> toshiba
pbx -Call droping

---------- Forwarded message ----------
From: Vidura Senadeera <vidurased at gmail.com>
Date: Mar 8, 2007 11:27 AM 
Subject: Re: Back to back E1 - asterisk <=> toshiba pbx - Call droping
To: asterisk-users at lists.digium.com

Hi steve and All,

I'm attaching cat proc/interrupts out put, lspci -bv out put, zapa.conf,
zaptel.conf for your information

Thanks so much for the feedback and I do accordingly. Hope to get rid
off this isue any how.

To day also reported 10 call drops within 2 hours of period.

fook forward to have your support on this regard.

Thanks & Regards,

Vidura Senadeera,

Network Engineer,

Debug Solutions

Sri Lanka.

Tel - +94114520036

Mobile - +94777766596

Web - www.debug.lk <http://www.debug.lk/>  

	Message: 16
	Date: Wed, 7 Mar 2007 05:05:36 -0500
	From: "Steve Totaro" < stotaro at asteriskhelpdesk.com
<mailto:stotaro at asteriskhelpdesk.com> >
	Subject: RE: [asterisk-users] Back to back E1 - asterisk <=>
toshiba
	       pbx -   Calldroping issue
	To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
	       <asterisk-users at lists.digium.com>
	Message-ID:
	       <
DFB93BD730105941BD1A782A1EE9E95CCC76 at 1-0fa9e300af524.asteriskhelpdesk.co
m
<mailto:DFB93BD730105941BD1A782A1EE9E95CCC76 at 1-0fa9e300af524.asteriskhel
pdesk.com> >

	Content-Type: text/plain; charset="us-ascii"

	As these problems are very time sensitive and frustrating, I
suggest you 
	document each change you make and do them one at a time so you
can
	actually know what the problem was and not introduce new
problems in the
	process.

	Find someone who is on the phone quite a bit and will give you
an honest 
	evaluation of the call dropping situation (unless you yourself
are
	experiencing this issue too).  Some people are so quick to say,
"It is
	still happening" without starting the evaluation from a clean
slate 
	after each change.

	You may want to check your Asterisk log for more insight.
	/var/log/asterisk/full.  Also you can turn on debugging on one
span at a
	time and see if you can find something there

	Do you have a resetinterval set in zapata.conf?  If you can
isolate the
	dropped calls to the reset interval (watch the console, it will
scroll
	with each channel being reset) then set resetinterval=never.  If
there 
	is no entry for resetinterval, add it and set it to never since
it is
	defaulted to on.

	Also, try changing your second span timing from
span=2,2,0,ccs,hdb3,crc4
	to span=2,0,0,ccs,hdb3,crc4.  This in combination with your
first span 
	should accept timing from the Telco and then supply it to your
Toshiba,
	I would actually try this first.

	Another thought, It seems you have quite a lot of hardware in
that box.
	I am not sure how much is too much, but that would probably just
rear 
	it's ugly head as poor audio.

	Thanks,
	Steve Totaro
	http://www.asteriskhelpdesk.com
<http://www.asteriskhelpdesk.com/>  

	_____

	From: asterisk-users-bounces at lists.digium.com
	[mailto: asterisk-users-bounces at lists.digium.com
<mailto:asterisk-users-bounces at lists.digium.com> ] On Behalf Of Vidura
	Senadeera
	Sent: Wednesday, March 07, 2007 2:15 AM 
	To: support at digium.com
	Cc: asterisk-users at lists.digium.com
	Subject: [asterisk-users] Back to back E1 - asterisk <=> toshiba
pbx - 
	Calldroping issue

	Hi Team,

	I have integrated asterisk with Toshiba analog PBX. NOw the live
setup
	is going.

	Now I am facing call droping problem. It's happening ample time.
10-20 
	calls are droping every day.

	What could be the reason. I attached latest zapata.conf file for
your
	information.

	This is being a huge issue.

	Highly appreciate your help on this regard. 

	Thanks & Regards,

	Vidura Senadeera.

	On 1/26/07, Vidura Senadeera <vidurased at gmail.com > wrote:

	Dear Marco,

	There is a huge problem i'm facing. 

	My asterisk server included with TDM2451E and 2 TE110p cards.
One E1 i
	conected to the telco. other E1 port i'm using to
cros-connection with 
	toshiba pbx. My telco E1 d channels communicating well. but
toshiba pbx 
	E1 not getting. d-channels are not getting up.

	what could be the issue. i'm using asterisk -1.2.14 and zaptel
1.2.12.

	notes - if i put, zap show channels in asterisk cli. its only
showing
	the first 31 channels. but with ztcfg -vvv it showing al the
channels.

	my configs are

	# Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0"
HDB3/CCS/CRC4 RED 

	# ============ Suntel E1 connection ========== 

	span=1,1,0,ccs,hdb3,crc4
	bchan=1-15,17-31
	dchan=16

	# Span 2: WCT1/1 "Digium Wildcard TE110P T1/E1 Card 1"
	# ============ Legacy PBX E1 connection ======= 

	span=2,2,0,ccs,hdb3,crc4
	bchan=32-46,48-62
	dchan=47

	# Span 3: WCTDM/0 "Wildcard TDM2400P Prototype Board 1"
	fxoks=63
	fxoks=64
	fxoks=65
	fxoks=66
	fxoks=67
	fxoks=68
	fxoks=69 
	fxoks=70
	fxoks=71
	fxoks=72 
	fxoks=73
	fxoks=74
	fxoks=75
	fxoks=76
	fxoks=77
	fxoks=78
	fxoks=79
	fxoks=80
	fxoks=81
	fxoks=82
	fxsks=83
	fxsks=84
	fxsks=85
	fxsks=86

	# Global data 

	loadzone        = us
	defaultzone     = us

	Regards,

	vidura

	--
	Thanks & Regards,
	Vidura B. Senadeera.

	--
	Thanks & Regards,
	Vidura B. Senadeera. 

	--
	Thanks & Regards,
	Vidura B. Senadeera.

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	Message: 17
	Date: Wed, 7 Mar 2007 11:17:07 +0100
	From: "Thomas Deillon" < Thomas.Deillon at smart-telecom.ch>
	Subject: [asterisk-users] Asterisk 1.4.1 - Calling problem 
	To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
	       <asterisk-users at lists.digium.com >
	Message-ID:
	       <
86918CDC1242004D8B0563A43D1E2F0C027E2D09 at exch-pul-01.interne.smart-telec
om.ch
<mailto:86918CDC1242004D8B0563A43D1E2F0C027E2D09 at exch-pul-01.interne.sma
rt-telecom.ch> >

	Content-Type: text/plain; charset="us-ascii"

	Hi all,

	I install the Asterisk 1.4.1 in order to use the T.38
pass-through, but
	for the moment, I cannot even make call ....

	I have this WARNING:

	[Mar  7 11:32:09] WARNING[13395]: chan_sip.c:12290
handle_response:
	Remote host can't match request BYE to call 
	' 5759b80c119e1d51679dc66b519c6eac at 194.148.41.50
<mailto:5759b80c119e1d51679dc66b519c6eac at 194.148.41.50> '. Giving up.

	Do you know what is this error and what can I do to solve it ?

	Thanks a lot for your help,

	Thomas

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-- 
Thanks & Regards,
Vidura B. Senadeera. 

-- 
Thanks & Regards,
Vidura B. Senadeera. 

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