similar to: MFC/R2 protocol varient - sri lanka/Nortel DMS 100

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, > >

> 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 .

> 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.

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

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:

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

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,

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:

Hi, Let us know more information 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

Hellow, * I have a problem with dial up signalling. currently 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.

Hello: As someone making the move from STATA to R, I'm finding it difficult at times to perform basic tasks in R, so forgive me if I've missed an obvious and easily obtained solution to my problem. I've searched the help guides and the archives and have not been able to find a solution that works. I have a data frame with thousands of observations that looks something like this:

I currrently have Asterisk 1.2.8 with a TE110P Zaptel card tied to our Nortal DMS-10 switch via T1. This T1 trunk is configured for Feature Group D MF. The purpose of this is for LD delivery to a VoIP LD provider as a LD choice for our customers off of the DMS-10. When I sieze a channel on the zaptel, I get a spill of digits from the switch that is the called number, and I can see that in

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

Dear 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

Hi, I am a Computer Science and Engineering student at University of Moratuwa, Sri Lanka. I went through the project proposals of Xapian at Gsoc 2014. I found the project "c# binding improvement", very attractive for me to start up with committing to open-source community. And then found that is is not taken by any student. Please, some one can help me to start on this. I have 3 years

Hi everybody, I have been searching around for days on how to record calls between SIP Doesn't seem to work during a call. Thanks Ishanka -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned for viruses and dangerous content by ITABS Lanka Mail Scanner, and is believed to be

> > Hi, You can achieve this by integrate CCM and asterisk using SIP trunk. In CCM you can create SIP trunk, After creating SIP trunk in 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

Dear All, Please provide me access to create wiki pages for my howtos Regards, Chinthaka -- Chinthaka Deshapriya (RHCE) Director/CTO Centre of Open Source Excellence Fossmart Private Limited. 163/1, Dutugemunu Street, Kohuwala, Colombo Sri Lanka http://www.fossmart.net Tel: +94-114-962-600 Fax: +94-112-823-429 Mob: +94-716-903-433 -------------- next part -------------- An HTML attachment

Hi, Is to possible to have a language construction and IDE integration kit for LLVM so that LLVM / Clang can be used for Meta Programming and defining new languages and DSLs. The definition of grammar, parsing, debugging, lint checking and IDE integration should be seamless and easy to use for a novice. Suminda -- Suminda Sirinath Salpitikorala Dharmasena, B.Sc. Comp. & I.S. (Hon.) Lond.,

Hi All, I am Ananda student of university of Moratuwa Sri lanka(www.mrt.ac.lk). I have worked with SAHNA open source community and have a experience with open source software as well. I am familiar with C language base soft ware development through the my university master degree and out source, so that I have decided to apply for GSOC in this year through your organisation. I am going to apply