similar to: Which ATA adapter to use with an analog fax maschine?

Hi, I saw your posting about the cisco ata186 I2 vs I1 and the simple vs complex impedance. I ordered a cisco ata186 i2 for use in Canada by mistake, didn't know that I needed the I1 version. Will the I2 version work in Canada with regular anlog phones, or will I need to change it. Thanks for your answer. Samy -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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

Hi Does anybody if the X100P works in Switzerland? We can't get a line to PSTN. When I run zttool it shows me always a red alert. I can make and receive calls with an anlog phone plugged in the phone connector. I've compiled and configured the card according to the wiki. Everything seemed to be ok. Is there a way to debug this? Regards Reto

Hello, Is it possible to transfer ss7 signaling over an FXO interface. I need to setup an ss7 test system composed by two Asterisk based IP-PBX systems with anlog interfaces only (FXO and FXS). I want to know if it is possible to connect the two IP-PBX as following: - FXS interface in PBX1 -----------------> connected to -----------------> FXO interface in PBX2 =============>

I've gotten nothing from http://bbs.atcom.cn on this so far. Perhaps someone on the list has experience with this. Is there a way to get MWI support for PA168V-based ATAs? Apparently some IP phones based on the PA168V chip has this support already (Atcom AT-320 for example) by configuring Asterisk with 'mailboxdetails=yes' in iax.conf. On my ATA, however, it does nothing. Any

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:

we have performance problems with printing with cups/samba we use samba 2.2.1a on this (print)server (sideproblem: the authentication to another samba 2.2.0a (file)server seems to be broken) first the printer's print postscript and we left it to cups (or ghostscript) to render the appropriate printerdriver. but this really torn down the maschine. now we have the drivers installed on the

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

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.

Hi List, what do you think is the best way to migrate (to a new maschine) round about 30000 mboxes (Qpopper) with an amount of 43Gigs data, to maildir format (Dovecot). I think there are two ways: 1. - stop services (smtp and pop3) on the old maschine - copy the mboxes to the new maschine - run a conversion script (for excample: "Perfect_maildir" http://perfectmaildir.home-dn.net/)

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

Hello experts, I've migrated our NT4 domain to sambe 3.0.20b/ldap backend with "net rpc vampire", and nearly everything works as expected. But one big problem remains: it's not possible to login to the domains member maschines now, because "the domain is not available at the moment" (translated from german). After the maschine rejoined the samba domain, login

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. >>>

hello friends, I want to install Asterisk on a Debian machine. I need to download the sources or just with apt-get install is enought??? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070320/227a3b32/attachment.htm

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

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,

On Mon, 14 Oct 2013, Basil Mohamed Gohar wrote: > To: icecast at xiph.org > From: Basil Mohamed Gohar <basilgohar at librevideo.org> > Subject: Re: [Icecast] Bandwidth Usage > > On 10/14/2013 12:42 PM, Keith Roberts wrote: >> If there is no sound input on the client audio stream being >> sent to the icecast server does this mean there is no >> bandwidth

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