search for: vidurased

Displaying 20 results from an estimated 28 matches for "vidurased".

2009 Jun 29
1
ISP< ->Asterisk <-> ATA <->DIALUP
...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
2007 Mar 09
0
Re: Back to back E1 - asterisk <=> toshiba pbx - Call droping issue [ ref:00D36mPe.50032wycQ:ref ]
...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 Engineer, > > Debug Solutions > > Sri Lan...
2007 Aug 21
6
Saftware RAID1 or Hardware RAID1 with Asterisk
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 RAID1 support CENT OS 4.4 Asterisk 1.2.19 Libpri/zaptel latest release 2.8 Ghz Intel processor 2 80 GB SATA Hard disks 256 MB RAM digium PRI/E1 card Following are the concerns I am having I'm planing to put this asterisk
2009 Apr 16
1
Can Asterisk bridge between a SIP client and a Cisco Call
...ialplan 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 > a Cisco Call Manager server? > To: as...
2007 Mar 08
0
Fwd: Back to back E1 - asterisk <=> toshiba pbx -Call droping
...s-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 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 Than...
2007 Dec 03
1
Subject: Newb Question
Hi, Use orecx, voip call recording and monitoring. www.orecx.com Thanks & Regards, Vidura Senadeera, Sri Lanka. Tel - +94114520001 Mobile - +94777766596 yahoo/skype Ids - vidurased > ------------------------------ > > Message: 17 > Date: Fri, 30 Nov 2007 08:58:41 +0530 > From: ram <talk2ram at gmail.com> > Subject: Re: [asterisk-users] Newb Question > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-u...
2010 Jul 16
1
IAX endpoints not Registering after upgrage from Asterisk ver 1.4.26.1
...s. 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
2007 Dec 13
0
Didnt get a frame from Channel and call gets
...ation about your setup. Hardware/software details details such as. server configuration PSTN cards you are using?? ( E1 or FXO card) sip.conf, zapata.cons, zaptel.conf config details?? Thanks & Regards, Vidura Senadeera, Sri Lanka. Tel - +94114520001 Mobile - +94777766596 yahoo/skype Ids - vidurased ================== Message: 5 Date: Mon, 10 Dec 2007 15:26:52 -0800 From: "Jai Rangi" <jprangi at gmail.com> Subject: [asterisk-users] Didnt get a frame from Channel and call gets disconnected To: asterisk-users at lists.digium.com Message-ID: <eb007ec0712101526x4...
2007 Mar 07
0
Back to back E1 - asterisk <=> toshiba pbx - Calldroping issue
...ple 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@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 p...
2006 Apr 26
1
Intergrate Asterisk IP PBX with Legacy PBX, continuing existing funtionality of legacy pbx
Hi All, I would like to explain the layout that i am trying to achive. I am so helpless on this regard. So here is the story ........ " This is with regard to the setup which you can find at the "Asterisk The Future of Telephony" , chapter 11, page # 196-197, I am attaching the picture for your information. Now I am taking a challenging step to of integrate IP PBX with our
2007 Mar 15
0
Re: busy/hangup/answer detection in PRI E1 channels (Vidura Senadeera)
...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: "Vidura Senadeera" <vidurased@gmail.com> > Subject: [asterisk-users] busy/hangup/answer detection in PRI E1 > channels > To: asterisk-users@lists.digium.com > Message-ID: > <afa14ef00703150309k51b4bbd8ka239ca7e2ba08fb@mail.gmail.com > > Content-Type: text/plain; charset="iso-8859-1&...
2007 Sep 05
4
ztcfg error : TE110p error with " CAS signalling on span 1 conflicts with HDLC with ...
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 asterisk 1.2.19 cent OS 4.4 zaptel 1.2.18 libpri 1.2.4 etc/zaptel.conf span=1,0,0,cas,hdb3 bchan=1-15,17-31 dchan=16 when i ztcfg -vvv im having this error message and the E1 is not getting up. "cas
2007 Nov 27
10
Asterisk behind a PIX firewall?
Is there anything special that anyone here has had to do to get an Aastra phone (on the Internet) to talk to Asterisk behind a PIX firewall? Ports 10000-20000 UDP are open on the PIX and forwarding to the Asterisk server. The Asterisk server's RTP.CONF is set to use 10000-20000. The phone registers, and will place AND receive calls, however, no audio is passed. The phone is an Aastra
2007 Mar 07
1
Re: Back to back E1 - asterisk <=> toshiba pbx - Call droping
...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@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 tel...
2007 Aug 23
0
asterisk-users Digest, Vol 37, Issue 88
...8859-1" For RAID1, I am not sure. But for RAID 5, You should always use hardware RAID. If you use software RAID and your CPU spikes for too long, you can corrupt your disks. I have seen this several times. -- -- Steven http://www.glimasoutheast.org "Vidura Senadeera" <vidurased at gmail.com> wrote in message news:afa14ef00708201903v209f8034lca006c8093251a2a at mail.gmail.com... 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 RAID1 suppor...
2007 Mar 07
1
Back to back E1 - asterisk <=> toshiba pbx - Call droping issue
Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: zapata.conf Type: application/octet-stream Size: 2410 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070307/871ac980/zapata.obj
2007 Mar 08
0
"pritimer" parameter in zapata.conf
Hi all, Please discribe me more about pritimer parameter in zapata.conf http://lists.digium.com/pipermail/asterisk-commits/2006-July/005824.html I found above 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
2007 Aug 28
0
(no subject)
> 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 Software RAID, what would be the impact to my deployment? ( > problems that I have to face with regard to the call flow ) None. > 2.
2007 Aug 28
0
Saftware RAID1 or Hardware RAID1 with Asterisk (Andrew Joakimsen)
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 Software RAID, what would be the impact to my
2007 Sep 06
1
14. Re: ztcfg error : TE110p error with " CAS signalling on span1 conflicts with HDLC with ... (Carlos Chavez)
Hi Carlos/All, 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,