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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vidura Senadeera Sent: Thursday, March 08, 2007 1:01 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Fwd: Back to back E1 - asterisk <=> toshiba pbx -Call droping ---------- Forwarded message ---------- From: Vidura Senadeera <[EMAIL PROTECTED]> Date: Mar 8, 2007 11:27 AM Subject: Re: Back to back E1 - asterisk <=> toshiba pbx - Call droping To: asterisk-users@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" < [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> > Subject: RE: [asterisk-users] Back to back E1 - asterisk <=> toshiba pbx - Calldroping issue To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: < [EMAIL PROTECTED] m <mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> ] On Behalf Of Vidura Senadeera Sent: Wednesday, March 07, 2007 2:15 AM To: [EMAIL PROTECTED] Cc: asterisk-users@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 <[EMAIL PROTECTED] > 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070307/62 a82c69/attachment-0001.htm ------------------------------ Message: 17 Date: Wed, 7 Mar 2007 11:17:07 +0100 From: "Thomas Deillon" < [EMAIL PROTECTED]> Subject: [asterisk-users] Asterisk 1.4.1 - Calling problem To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com > Message-ID: < [EMAIL PROTECTED] om.ch <mailto:[EMAIL PROTECTED] 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 ' [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> '. Giving up. Do you know what is this error and what can I do to solve it ? Thanks a lot for your help, Thomas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070307/67 40a4e6/attachment.htm ------------------------------ _______________________________________________ --Bandwidth and Colocation provided by Easynews.com <http://easynews.com/> -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <http://lists.digium.com/mailman/listinfo/asterisk-users> End of asterisk-users Digest, Vol 32, Issue 22 ********************************************** -- Thanks & Regards, Vidura B. Senadeera. -- Thanks & Regards, Vidura B. Senadeera.
_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
