search for: anlog

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

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: http://lists.digium.com/pipermail/asterisk-users/attachments/20030905/61af5090/attachment.htm

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 =============> used to transport ss7 signaling. - FXS interface in PBX2 -----...

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 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, Firstly, I'm new to Asterisk and am a system admin rather than a phone engineer. I've googled and read around but haven't been able to answer my questions sufficiently to buy hardware and get this thing set up. Secondly, if I've missed vital information from what is below, please let me know what. Onward...: So, what I'm trying to solve us remote working. We're a

...gt; audio sine wave into the soundcard, and then use the > default VBR to encode the stream to OggV and stream that up > to the icecast server? As Mohamed already pointed out it all depdends on what is before your encoder and is not strongly related to icecast. If you are encoding from an anlog source (without some post processing on your system) your stream will likely drop in used *bitrate* but not to zero as the codec will encode the noise from that input. This is true and should work fine with icecast (no strange timeouts and stuff) with Ogg Vorbis which I strongly recommand for most...

Hi all, Sorry if this is a little off topic, but I just want to know a thing here. What system is used for sending out the caller's number in the US? Here in Sweden we use DTMF to send the number out. I just need to know what is used in the US since I don't think I will be able to use an American caller ID device over here. Many thanks for any info, Christian

...self to no avail. Here is my setup. Asterisk Box Digium T1 card connected to an adtran Total Access 624. I need to pass ANI and DNIS information from asterisk to the analog devices hanging off the adtran. The digits are announced to the fxs devices between ring 1 and ring 2 (If I pick up the anlog phone after ring 1, I can hear them so I feel good about this stage). Problem is that my analog device "once it is finished with the call" needs to be able to "FLASHHOOK"<exten> to transfer the call to a SIP Handset. Problem is that as soon as the FLASH (which is 700mil...

I am running a very small dovecot installation with only one user (me). I use the Solr indexer for indexing. Due to complicated reasons, I was forced to remove all the indexes and need to re-index everything. All the files in my Maildir are owned by md5i:mail (I am md5i), and have 660 permissions. All directories have the same user:group permissions, and 770 with the setguid bit set. (That

i am just curious how many * systems are in the real world with more than one user. do you run a certain version? you dont update CVS do you? any admins running a system of over twenty? over fifty? over one-hundred? i deal with 3com and nec systems all day (i am cerified in the 3com nbx advanced network telephony, elite voice mail, CCNA, A+, nec ipk, and soon to be "asterisk school of

I have 8 lines coming into an existing PBX system and am looking for a cost effective way to replace the existing system with Asterisk. We need some of the features in Asterisk, including its ability to support remote offices (long distance savings). At first glance this appears to require a T100P card and a channel bank, but that seems rather expensive. My estimated price on that would be

Hi List Is someone out there using one or more GSMgateway(s) from CyberTelecom ? Me and some friends are interested in buying some of them, but before we would like to ask, how the experiences are others have made. e.g. How easy to setup ? How reliable ? How's the voice quality ? etc. Any input/feedback is welcome. Greets Adibar --