search for: eky

On 12 Eki 2012, at 20:00, Jim Grosbach wrote: > > On Oct 12, 2012, at 7:07 AM, Baris Aktemur <baris.aktemur at ozyegin.edu.tr> wrote: > >> Dear Tim, >> >>> >>> The JIT sounds like it does almost exactly what you want. LLVM's JIT >>> isn't a classical lightweight, dynamic one like you'd see for >>> JavaScript or Java.

S.Çağlar Onur wrote: > 17 Eki 2006 Sal 01:21 tarihinde, S.Çağlar Onur şunları yazmıştı: > >> 17 Eki 2006 Sal 01:17 tarihinde, Zachary Amsden şunları yazmıştı: >> >>> My nasty quick patch might not apply - the only tree I've got is a very >>> hacked 2.6.18-rc6-mm1+local-patches thing, but the fix should be obvious >>> enough. >>>

S.Çağlar Onur wrote: > 17 Eki 2006 Sal 01:21 tarihinde, S.Çağlar Onur şunları yazmıştı: > >> 17 Eki 2006 Sal 01:17 tarihinde, Zachary Amsden şunları yazmıştı: >> >>> My nasty quick patch might not apply - the only tree I've got is a very >>> hacked 2.6.18-rc6-mm1+local-patches thing, but the fix should be obvious >>> enough. >>>

Hi, I configured cdr_mysql (addons 1.0.9) to write the cdr records to the mysql db. The problem is that no records are written to the db. Why? I can import the csv-file to the db. so i assume the db is setup correct. Is there any chance to get debug from cdr_mysql to find his problem? This is my cdr_mysql.conf file: [global] hostname=localhost dbname=cdr password=passw0rd user=root ;port=3306

Dear Jim, On 12 Eki 2012, at 21:17, Jim Grosbach wrote: > > On Oct 12, 2012, at 11:14 AM, Baris Aktemur <baris.aktemur at ozyegin.edu.tr> wrote: > >> >> On 12 Eki 2012, at 20:00, Jim Grosbach wrote: >> >>> >>> On Oct 12, 2012, at 7:07 AM, Baris Aktemur <baris.aktemur at ozyegin.edu.tr> wrote: >>> >>>> Dear Tim,

Hi, we successfull managed to bridge a PSTN (E1) switch over the TE405P card to our old PBX. So now we could migrate to the * server. But, there are two things we can't live with: 1. A call from the outside to the old PBX is missing a leading 0 before the number. Ex: caller has number 0123456 -> * routes to old pbx -> old pbx sees 123456 as caller number. 2. A call made from a SIP

On Oct 12, 2012, at 11:14 AM, Baris Aktemur <baris.aktemur at ozyegin.edu.tr> wrote: > > On 12 Eki 2012, at 20:00, Jim Grosbach wrote: > >> >> On Oct 12, 2012, at 7:07 AM, Baris Aktemur <baris.aktemur at ozyegin.edu.tr> wrote: >> >>> Dear Tim, >>> >>>> >>>> The JIT sounds like it does almost exactly what you

Hi, i am running * 1.0.9 with a newer version of the TE405P. Modprobe wct4xxp and ztcfg are OK. zap show channels only shows me the following. my zapata.conf: [pstn] switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown usecallingpres=yes busydetect=no ; not need on pri callprogress=no ; was yes but wiki says experimatley could be produce hangups callwaitingcallerid=yes ; show

Hi, i need an recommandation for an ATA adapter to use with an anlog fax maschine. I would appreciate any hints. Regards!

>From searching the web, I found that many people have encountered this problem. The fixes the suggested don't seem to work for me... My smb.conf file looks like this [global] workgroup = OURDOMAIN security = ADS realm = OURDOMAIN.com password server = OURSERVER encrypt passwords = yes add user script = /usr/sbin/useradd %u hosts allow = 192.168.X. 127. winbind uid = 10000-20000 winbind

On Fri, 5 Apr 2024 16:43:42 +0300 Michael Tokarev via samba <samba at lists.samba.org> wrote: > Hi! > > We had stand-alone anonymous samba server serving a read-only share > as guest account. It worked well but had a few strange issues (like > lots of noise in logs about bad smb2 signature). > > Its been suggested to switch to a domain member server. I didn't

Hi, at the moment we have in Avaya Integral PBX with german pri (30 lines). We want to smouthly migrate to an Asterisk server. For this reason: Is it possible to route the external german pri (E1) through Asterisk server to that Avaya PBX? I think at first we need a Digium e1 card 4-Port. But how do we have to configure the routing of the whole PRI? I really would appreciate any sample

Hello everybody, I installed eucalyptus on ubuntu10, basic configuration . Everything is ok untill I start the instances, the disk is being created but it terminates soon. Environment: NFS , KVM, ubuntu10 x64, Euca on 2 machines (wc sc cc clc on one machine, and nc on another). libvirtd (libvirt) 0.7.5 QEMU PC emulator version 0.12.3 (qemu-kvm-0.12.3), Copyright (c) 2003-2008 Fabrice Bellard.

Hi *, I want to integrate the Eicon Diva 4Bri Card to Asterisk. Eicon drivers and capi is installed. I use the latest dev version from eicon compiled and installed for my fedora 2 system. I found the chan_capi for asterisk from www.junghanns.net. Also loaded the patch and applied to the chan_capi source tree. I changed the Makefile to include the capi20.h from eicon:

On Oct 12, 2012, at 7:07 AM, Baris Aktemur <baris.aktemur at ozyegin.edu.tr> wrote: > Dear Tim, > >> >> The JIT sounds like it does almost exactly what you want. LLVM's JIT >> isn't a classical lightweight, dynamic one like you'd see for >> JavaScript or Java. All it really does is produce a native .o file in >> memory, take care of the

I know i''m new with ruby and i''ve been kind of confused with how rails manages all the details. I want to store the key-values in different columns on my database so that when I enter a eky, it gies me the corresponding value. this is what i have: my view: <h1>Importing Strings<h1> <%=error_messages_for("string_record")%> <%= start_form_tag({:action => ''save''}, :multipart => true)%> <p> <b>Lenguage:&l...

Hi, when i use the *8 for the call pickup the call i fetch is directly connected and i can't see the callers number. What i want is that the call in the first only rings at my phone and in the second i can see the callers number before i am connected. I am using a polycom 500 ip phone. Is this a special polycom problem? Regards, Kib

Hi, the following from extension.conf does not work correctly: exten => 301, 1, Dial(SIP/455&SIP/456, 15) That is the console output: -- Executing Dial("mISDN/1/105", "SIP/455&SIP/456&SIP/456| 10") in new stack -- Called 455 -- Called 456 -- SIP/455-46a8 is ringing == Spawn extension (incoming, 301, 1) exited non-zero on

Hi, the option c for the announce of the user count does not work me in * 1.0.9. exten => 9999,1,Wait(1) exten => 9999,2,MeetMe(|Mdcs) And how to handel the marked mode with option A? I can't find any sample config for this. Regards

Hi, I don't if was yet an issue. It really would be nice if each user is able to active/deactivate the mail forwarding of his voicemail via the VoiceMailMenu. Regard