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

Wed Mar 7 22:57:02 MST 2007

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

Message: 16
> Date: Wed, 7 Mar 2007 05:05:36 -0500
> From: "Steve Totaro" <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.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
>
>
> _____
>
> From: 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-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'. 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.
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