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...Hi Gareth Blades & Doug, > > Thanks so much for for the feedback. I have searched on lot of documents > but couldn't able to find clear answer regarding it. > > I hope you guys replies are very much help all in aterisk community. > > > Thanks & Regards, > > Vidura Senadeera, > > Network Engineer, > > Debug Solutions > > Sri Lanka . > > Tel - +94114520036 > > Mobile - +94777766596 > > Web - www.debug.lk > > ------------------------------ > > Message: 14 > Date: Thu, 15 Mar 2007 15:39:07 +0530 > From: &quo...

Hi All, Thanks for every one who helped me on this regard. I think i was able to rictify the problem. what i did is remove callprogress=yes usecallinpres=yes and restart asterisk. Today i didn't report any drop calls. Many thanks for Eric. :) I hope this situation will continue. Regards, Vidura. On 3/8/07, Vidura Senadeera <vidurased@gmail.com> wrote: > > Hi, > > Opps ...there are some more attachments i missed to send you. Please > refer. sorry for the inconvenience occured. > > > Thanks & Regards, > > Vidura Senadeera, > > Network Enginee...

...ms 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@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] 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 <vidurased@gmail.com> Date: Mar 8, 2007 11:27...

...dware 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@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Vidura Senadeera Sent: Wednesday, March 07, 2007 2:15 AM To: support@digium.com 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 g...

> > Dear all, Thanks for the greate explanation regaing Software/H/W Raid. This details better but on voip-info.org/wiki pages. Thanks lot agian. Regs, Vidura Senadeera. ====================================== Dear All, > > > > I would like to get community's feedback with regard to RAID1 ( Software > or > > Hardware) implementations with asterisk. > > > > This is my setup > > > > Motherboard with SATA...

...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@asteriskhelpdesk.com> > Subject: RE: [asterisk-users] Back to back...

...at I have to face with regard to the call flow ) 2. If I use Hardware based RAID 1, what would be the impact to the system? 3. According to your practical experiance what is the ideal solution among both options? I will be highly appreciate your feedback on this regard. -- Thanks & Regards, Vidura Senadeera, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070821/1294b9de/attachment.htm

...e E1 is not getting up. "cas signalling on span1 conflicts with HDLC with FCS on channel 16" The switchtype and signalling im using is national, pri_cpe I'm attaching the avaya config details for more information. Please help me to sorted out this problem. - Thanks & Regards, Vidura Senadeera, Sri Lanka. Tel - +94114520036 Mobile - +94777766596 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070905/929ee2a6/attachment.htm -------------- next part -------------- An embedded and chars...

..., Thanks for your reply. I can remove dchan=16 from zaptel.conf But according to the documentation of Digium and sangoma they mentioning to use dchan=16. Are there any specific reason you have experiance regarding this and I am confusing that what this is included to the documentations. Regards, Vidura. On Wed, 2007-09-05 at 20:26 +0530, Vidura Senadeera wrote: > Dear All, > > I'm integrating avaya commuication manager difinity ver 1.0 with > asterisk using B2B E1. following are the details of my H/W, > zaptel configs and software installed. > > Digium TE110p > aster...

...y I have configured asterisk server and E1 card to ISP. then other side I am having ATA to PC for connecting internet through DialUP connection. is it possible and please send me the procedure how I can do it ?? * ISP< <-> Asterisk <-> ATA <-> DIALUP -- Thanks & Regards, Vidura Senadeera, Sri Lanka. msn/yahoo/skype Ids - vidurased -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090629/516b73bc/attachment.htm

...stn' > > > > Try changing this to 'context=from-zaptel' > > > > Robert Jenkins. > > > > > > > > _____ > > > > From: asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Vidura > > Senadeera > > Sent: 19 January 2007 15:19 > > To: asterisk-users@lists.digium.com > > Cc: support@digium.com > > Subject: [asterisk-users] Integrating asterisk with Toshiba Astrata > > DK380 > > > > > > > > Deat all, > > > &gt...

...ABX. All previous DID's which configured with the traditional PABX will be configured in asterisk. Actually I am not sure that i will be able to achive this migration , but i am trying to acomplish. It is very much appreciate that if anyone can guide me on this regard. Thanks & Regards, Vidura Senadeera. Sri Lanka. -------------- next part -------------- A non-text attachment was scrubbed... Name: legacy pbx to asterisk migration.JPG Type: image/jpeg Size: 15175 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060426/16b33b0d/legacypbxtoaste...

...> s,n,Set(CHAN=${CHANNEL:4}) exten => s,n,Set(CHAN=${CUT(CHAN,-,1)}) exten => s,n,Macro(from-zaptel-${CHAN},${DID},1) ; If nothing there, then treat it as a DID exten => s,n,NoOp(Returned from Macro from-zaptel-${CHAN}) exten => s,n,Goto(ext-did,${DID},1) -- Thanks & Regards, Vidura B. Senadeera. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070119/01153415/attachment.htm

Dear Andrew, Thanks for your kind responce. Regards, vidura. ============================= > Motherboard with SATA RAID1 support That's a mulit-port SATA controller with RAID in the driver (software). > 256 MB RAM Use a little more RAM. > digium PRI/E1 card Is there any reason you aren't using Sangoma cards? > 1. If I use Softwa...

...ar All, I am experiance a issue with my IAX clients. I have upgradeed Asterisk to 1.4.28 After then IAX clients are not working and It's not registering even. Please help. Asterisk previous version - 1.4.26.1 ( for this worked fine) FreePBX version - freepbx-2.5.2 -- Thanks & Regards, Vidura Senadeera, Sri Lanka. msn/yahoo/skype Ids - vidurased -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100716/a20da240/attachment.htm

...n between CCM and asterisk, you have to configure dialplan on CCM to pass the calls to asterisk. One the caller id comes to Asterisk you have to use extension.conf to route the calls. You can also try with freepbx GUI to configure inbound route, it makes your life easy. -- Thanks & Regards, Vidura Senadeera, Sri Lanka. msn/yahoo/skype Ids - vidurased > ====================================== > Message: 16 > Date: Fri, 10 Apr 2009 00:06:50 -0600 > From: Shocky <shocky1 at users.sourceforge.net> > Subject: [asterisk-users] Can Asterisk bridge between a SIP client and >...

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...ve url and have some idea. My PRI E1 timer is t203, what is the best vale that i have to use for as counter. default is 10000ms, If i changed it to some big amount, like 60000 what will happen ???? T203: Layer 2 max time without frames being exchanged (default 10000 ms) -- Thanks & Regards, Vidura B. Senadeera. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070308/34e320e0/attachment.htm

...ve to face with regard to the call flow ) None. > 2. If I use Hardware based RAID 1, what would be the impact to the system? A PCI slot. > 3. According to your practical experiance what is the ideal solution among > both options? Software RAID works fine. -- Thanks & Regards, Vidura Senadeera, Network Engineer, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +94777766596 Web - www.debug.lk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070828/ee00f97e/attachment.htm

...ing to setup E1 link with Nortel DMS 100, which is resides at one of telco provider in Sri Lanka. But we don't know what is the exact protocol varient to use. Is anyone help us out on this reagard. How do we know the exact details of the protocol varient we have to use?. Thanks & Regards, Vidura Senadeera, Senior solutions Specialist, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +94777766596 Web - www.debug.lk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071015/18dcaf72/attachment.ht...