similar to: POP 23 problem

Hi there I am using Dovecot on Ubuntu 6 LTS forPOP3 services. The user account created at installation works fine and Iam able to log to POP3 services by telnetr. When I created a another user and tried tologin via telnet this is what happens telnet XXX.XXX.XXX.XXX 10 + OK Dovecot ready user XXXX + OK pass XXX +OK logged in Connection tohost lost and session terminates. Pls advise what

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:

Hi All, We successfully installed MFC/R2, chan_unicall.so with asterisk ver 1.2.6. asterisk is loading properly and we can see US show channels working fine. We are using digium Te120P card. Now we are trying 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

Hi, I have a rsync server set up. Can i push data from the server to another machine instead of pulling data from that machine. when i try to do that i get this error. mkdir tohost:/tmp ; No such file or directory. But the directory exists. This is the command i executed. rsync -avz --rsync-path=/usr/bin/rsync fromhost::test tohost:/tmp This works fine if iam logged onto "tohost"

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

Am I reading the data below incorrectly, or does it appear that even though I have the directive canreinvite=no set for the two asterisk boxes, they are trying to do a reinvite (which fails) anyway? Is this expected behaviour in this situation? If so, how can I prevent this? ---- Lots of output ---- Using CVS Head from 2005-08-28, I have two asterisk boxen, one (box A) has a sip ua (2608)

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

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

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.

We're running a setup with M3W on a PC in Sri Lanka, encoding to a server in the UK. What factors govern the time delay between things being said to the microphone in SL and heard by a client connecting to the server, and how can we reduce it as much as possible? Currently we're experiencing 20-30 secs or more.

Hi, Does LLVM support decimal precision numbers supported? Also does it have Fixed point arithmetic? S -- Suminda Sirinath Salpitikorala Dharmasena, B.Sc. Comp. & I.S. (Hon.) Lond., P.G.Dip. Ind. Maths. J'Pura, MIEEE, MACM, CEO Sakrīō! ▣ *Address*: 6G • 1st Lane • Pagoda Road • Nugegoda 10250 • Sri Lanka. ▣ *Mobile* : +94-(0)711007945 ▣ *Tele*: +94-(0)11-5 864614 / 5 875614 / 2 825908 ▣

Hi everyone, I'm Shiluka Dharmasena, 3rd year computer science and engineering undergraduate from University of Moratuwa, Sri Lanka. I'm interested to participate as a gsoc student for Xiph.Org Foundation's Icecast project. It's a great opportunity for me to get involved in an open source project and I'm keen to contribute as a developer. I went through the project ideas

Hello community, I've been running Asterisk on an embedded device for about six months, and my operation has been largely trouble-free. I'm hoping I could get some help with a minor problem: Every week or three, my PBX gets stuck in a state where it can receive calls, but it becomes completely unable to originate outgoing calls until I do a "sip reload". After doing the SIP

Hi gurus!!! I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn Every minute asterisk sends an OPTION Request, i beleived that it's related to qualify functions. The every minute annoyng answer of the pstn is "403 Forbidden". Some people told that asterisk is not sending the username in the OPTION, required by the pstn. Taking a look of the example of rfc3261.txt

Hi I'm using samba 3 I want to hide dot files I used hide files = .* hide dot files = yes and even veto file no success at all, help would be greatly appreciate. cheers chaminda ---------------------------------------------------------------------- This e-mail message is confidential and for use by the addressee only. If the message is received by anyone other than the addressee,

Hi all, i have asterisk 1.2.17 with sip tcp support and i am trying to connect asterisk with HiPath 4000 V.3.0 using SIP. I can see the registration from the HG3540. But when i try to place a call from Asterisk to HiPath, the call fails with SIP/2.0 603 Declined. The strange thing is that the first INVITE uses tcp and the response is a 100 TRYING, the next 7 INVITE are using udp and the respose

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

Hi, to register my Asterisk with a SIP provider I use the following syntax, as shown in the default sip.conf: register => 2345:password@sip_proxy where [sip_proxy] type=peer context=from-messagenet host=sip.messagenet.it port=5061 <------------- please note this one!!! 5061 is provider's port I have to register to. This also would work for me: register =>