similar to: Maximum cable length for analog phone from FXS port

Hi. I'd like to get access to the CDR's generated by Asterisk (1.4) in real-time from a remote connection coming in on TCP. Basically what I have is a Windows application that is used to process incoming, outgoing and missed call records putting them into a database for some analysing etc. This app can connect to a TCP server and read from this connection the CDR's as they are

Greetings. I wonder what is the best way in your opinion to share real-time extension state with applications outside of asterisk? What I'm after is the best way to have Asterisk update a central repository with the state of each extension configured in the local Asterisk setup. To try and explain what I am trying to achieve, Imagine for example if asterisk would call a url like this:

Good question. I'm coming to the conclusion that using plain UDP and "home-grown" packet construction for transmitting the speex data (with timestamp/sequence counter) and implementing jitter control on the receiver end is an adequate implementation for a VoIP application. Assuming of course that I don't care about any interoperability issues with other applications etc. I was

Hi. Calling from Ericsson MD110 via H.323 trunk to an asterisk 1.4.4 I get the call established but no sound heard on either end. What is the best/correct way to try and see what codecs Asterisk is using on each end of the call as it passes through Asterisk? And is there any way to see that voice is in fact being passed through Asterisk during the call (some counters etc.)? Thank you

I have a call coming in from Asterisk-A going to Asterisk-B where it's determined that the called party is in fact yet another number in Asterisk-A so a new call is created from B to A and the two calls bridged (by Asterisk) at Asterisk-B. Originating Caller ==> Asterisk-A ==> Asterisk-B ==> Asterisk-A Now, what happens is that in my case both A and B are on the same network

Heya list, I need some advice/experience. Some of our customers are asking us about DECT solutions for their asterisk install. Some others will not go to asterisk if there won't be a DECT solution. They now have a Siemens or a Samsung PBX. Those PBX-es come with a DECT basestation and optionally repeaters etc. All those basestations speak some own protocol to the PBX, so we cannot use them

While I'm asking about the Polycom ip500, the answers for all phones where mic/handset/headset levels are adjustable would be of interest to many I'm sure. For the ip500, the default value for the handset seems to be voice.gain.tx.analog.handset="3" I've noticed that echo all but goes away when one reduces the mic volume on almost any phone. My question is, for you users

I've looked around and not found details on the expected frequency range the Speex codec can be expected to carry. Is there any documentation available or a table of some sort that has been compiled which would give an indication of the frequency range based on the various compression options in speex? Best regards, Baldvin Hansson Reykjavik, Iceland baldvin@baldvin.com -------------- next

Hello, I'm preparing a setup before installing it within the next few days. In this setup, I'm using a SPA112 as an ATA for an analog phone. The target phone is a Gigaset A400 DECT handset. In my lab, I've got another A400 handset and an old Matracom 46 handset. When I connect my Matracom 46 handset to my SPA112, I can send and receive calls. When I connect my A400 handset to the

hello all...I have been trying to get a handle on CallerPres with an analog handset. I have usecallingpres=yes in my chan_dahdi.conf file and when I dial *67 on my analog handset I see Disabling Caller*ID on DAHDI/4-1 but when the call is then forwarded to my outbound SIP provider the RPID header is not correct privacy=off;screen=no instead of full and yes how can I correct this?

Is there a way to initiate a transfer using an analog handset? For instance I'm looking for a way to do something like the following: External call comes in and is answered by user A. After talking to the caller they determine that the caller really needs to speak to user B. Is there any way for user A to initiate a transfer to user B, using only their analog handset? Now to make

I have a Digium TDM400p (1 FXO, 1 FXS) with the FXS module connected to a standard analog handset with caller id display (US caller ID). Although it appears that caller id information is coming into asterisk (it shows up in voicemail), I can not get it to display on the analog handset. Is there anything special I need to do to send the caller id info out the FXS port? I've tried a few

Hi, I have had this issues for ages but have ignored it, but my handsets get faint background crackle on the FXS port. When I connect the handset directly into my PSTN, it goes away complete, so I am generating this from somewhere in my asterisk box. I get it immediately when the phone goes off-hook and it stays through the dialing and call progress. I can't think of any other devices

Is there a way to hide the callerid on analog line on outgoing calls. Any ideas whether it could be done through configuration, a script or hardware. Thanks; ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com

We're having issues with the FXO/FXS ports on our Digium TDM cards sporadically. I'm wondering if anyone else has had these problems, or if anyone can provide guidance diagnosing or fixing the issue? The symptoms are that the FXO and FXS ports "stop working", usually after 2-4 weeks of server uptime. When this happens, sending a (SIP) call to an analog phone on an FXS port

Not strictly asterisk related, however... Here's an odd one for you.. I got a Nokia E90 and setup it's SIP client which runs via Wi-Fi (anyone know if I can make it work via GPRS/3G?) Anyway, in a fit of idleness, I thought I'd see what codecs it supports, as I couldn't find it in the manual... And it supports: ilbc g729 ulaw/alaw No GSM! How odd is that, given

On Tue, Jul 14, 2009 at 9:48 AM, Adam Rosi-Kessel<adam at rosi-kessel.org> wrote: > The only issue I'm noticing is ogginfo reports: > > Warning: sequence number gap in stream 1. Got page 14 when expecting > page 2. Indicates missing data. > Warning: discontinuity in stream (1) I'd guess this is flagging the data that was overwritten by the bad tagging code. Some

Hi. I have installed B410P in Europe and the cards works more or less ok. My question is what color should the LED's on the back of the card be when connected to the PSTN NT box? Is there anywhere some information on the expected LED color in any given state (idle, call active, cord unplugged etc.)? On my card the lights are shining Red(orange-ish) but flashing to green every now and

Hi. Using Asterisk 1.4.13 running on Ubuntu 7.04 with Intel CPU: 1) Is it possible/supported to install two or more B410P Digium cards in one computer (single Asterisk installation)? 2) Do they need to be hard-wired together with a PCM cable like I've seen explained in some beronet manuals (although that was specifically geared towards their cards, I must say)? Thank you for your time and

Hi. Using Asterisk 1.4.13 running on Ubuntu 7.04 with Intel CPU: 1) Not being able to build mISDN on Ubuntu using "make b410p" I have used mISDN-1_1_7 which seems to work ok. QUESTION: Should I expect this version of mISDN to work ok with these cards? Or is there a way to build using "make b410P" on Ubuntu? (make force does not help at all) 2) In some of our installations