similar to: how to decrease answer time !

Buen día a todos, Alguien conoce si existe la manera de determinar cuellos de botella pasados o actuales en poblaciones naturales remanentes con información genética (microsatélites) en R. Gracias por su valiosa ayuda de antemano. Saludos, Diego ________________ Diego P. Vélez Mora Departamento de Ciencias Biológicas UNIVERSIDAD TÉCNICA PARTICULAR DE LOJA San Cayetano Alto s/n Loja, Ecuador

Dear all, Anyone can point me how to soft hangup all channels using single command ? I am using Asterisk 1.4.15. thanks Salaque

(clean install) >gem install rails Attempting local installation of ''rails'' Local gem file not found: rails*.gem Attempting remote installation of ''rails'' Install required dependency activesupport? [Yn] y Install required dependency activerecord? [Yn] y Install required dependency actionpack? [Yn] y Install required dependency actionmailer? [Yn] y

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I''m using xen to run real linux system. I''ve configured partition for my system. When I''m installing Debian Etch everything is fine but when I''m starting installed system I get a hang. My xen disk conf is: disk = [ ''phy:/dev/root/node1,ioemu:hda,w'' ] device_model =

Dear List, I am new in this world (Asterisk) and facing a problem . i want to make some group, base on extensions .so that certain extensions could call to certain predefine number only. let me give u all a short example extensions 1,2,3,4 will be group A , extensions 5,6 will be group B . so group A only allowed to call one or certain predefine number but group B could call anywhere.

Dear List, I am facing another strange problem . some of my envisions like to use other prepaid card (whatever they found in market) but when they dial that access number (phone number to put the pin) they get IVR (Please provide your pin number ) but when my user press pin its not going through, that IVR even can't get wrong pin number . just get disconnected as no pin number provided. what

Anyone suffering an outage with them right now? I am getting the following from my box when I try to dial using them.... == No one is available to answer at this time W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050309/e1096325/attachment.htm

I just downloaded the zip file from grandstreams website to upgrade my gxp2000 firmware from 1.0.0.3 to the latest but seems there are some files missing on the zip file... Anybody been able to upgrade their firmware? My website shows this files as missing: 201.133.125.152 - - [11/May/2005:16:47:16 -0500] "GET /firmware/ring1.bin HTTP/1.0" 200 12737 "-" "Grandstream

I'm trying to use a Xorcom Astribank wth Ubuntu 8.04, but got no success. I can see the channel bank with lsusb, but when I tried to use zaptel_hardware, or when I try the /etc/init.d/script, they don't see my Channel Bank. I compiled the latest Zaptel 1.4.10, with Astribank's dependecies, fxload and libusb-dev. Anyone have a similiar experience ? Best Regards, -- Guilherme Loch G?es

Hey, For the bridge issue, take a look at 'notransfer=yes' option in your iax.conf. It'll force * to stay in the path http://www.mail-archive.com/asterisk-users@lists.digium.com/msg42262.html

Sipdiscount has replaced their asterisk servers for another thing. Then, no more iax. Ok, but I can't make calls using sip also... I'm getting a "forbidden" error when using sip1.sipdiscount.com. Anybody got it working? -- Alejandro Vargas

Hi folks, Just a quick question. Does the GrandStream GXP-2000 phone support presence (hints)? Cheers, Richard. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060109/26c4a63c/attachment.htm

Well, this is weird. After receiving a sip subscribe message from peer 2529266, here's what Asterisk responds with: -- (14 headers 0 lines)--- Found user '2529266' Looking for 2943110 in bell_CallStart (domain ua2.ipt.xxx.com) Dec 26 10:19:34 NOTICE[27345]: pbx.c:1741 pbx_extension_helper: Cannot find extension context 'bell_CallStart' Transmitting (no NAT) to

Hi List I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Thanks -- Chris Blunt Entropy IT Ltd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061220/4919f3cb/attachment.htm

I see that 1.0.5 is out. I thought that if I am tracking cvs v1.0.x I would always get the newest releases. However, I just did a fresh update and install from cvs stable and it reports as only being v1.0.3. Should I just be using the tarballs rather than the cvs -r 1_0? Or maybe my initial cvs was incorrect? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in

Hey, this is happening to anyone who I try this with. We get into the DISA, then hear the dial tone. Dial 1 then start dialing the number, and it hangs up. I thought adding a wait time after the DISA may help, I was wrong. Here is what I have thus far in the DISA extentions. [DISA] exten => 7,1,DISA(no-password||"Scheda" <565> 455-1337) exten => 7,2,Wait(45) exten =>

hello I am using ser with asterisk asterisk on 5070 (on back end) ser on 5060 (on front end) i am getting all requests at asterisk. i tried by changing asterisk port bindport=5090 but still getting all requests from sjphone at asterisk. can any one tell what is the reason regrads Kamran __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on

The person on 617 is unavailable --- Why???? *CLI> -- SIP Seeding peers from Astdb: '617' at 617@192.168.250.107:6990 for 3600 -- Executing Dial("SIP/601-f18a", "SIP/617|60|tr") in new stack -- Called 617 -- Got SIP response 480 "Temporarily Unavailable" back from 192.168.250.107 -- SIP/617-602e is circuit-busy *CLI> sip show

Hello, If asterisk.org can't provide you documentations have a look here : http://www.digium.com/index.php?menu=product_detail&category=software&product=ABE I do hope some people understand my posts. Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger

i dont how to edit the time for ringing "30000ms" to "40000ms" when it displayed on console "Nobody picked up in 30000 ms" and its very short time for ringing . please if anyone can help me do it please. ____________________________________________________ Sell on Yahoo! Auctions ? no fees. Bid on great items. http://auctions.yahoo.com/