search for: telephoning

Hi all, I have a lot of call in wich I found that my Asterisk doesn't answer the BYE message, then the BYEs are retransmitted, but the call ends, when the Asterisk sends a BYE. Time AS.TE.RI.SK CA.RR.IE.R1 0 INVITE SDP ( g729 g711A g711U telephone-event) SIP From: sip:1265666072 at 81.209.186.14 <sip%3A1265666072 at 81.209.186.14>To:sip:1234567890 at CA.RR.IE.R1 (5060)

Hello all, I hope someone of you can help me out, I have searched other posts as well but I can't find any solution to the problem I'm dealing with. I want to make a histogram from the data Telephone Lines MDGdataset <-read.csv("MDG_dataset_2010.csv", header=T) MDGdatasetAdapted <- subset(MDGdataset, select = c(Country_Code, Country_Name, Year,

https://bugzilla.mindrot.org/show_bug.cgi?id=2630 Bug ID: 2630 Summary: skype ('866.769.8127 ?>>>>>'''';l;skype tech support number skype ('866.769.8127 ?>>>>>'''';l;skype tech support number Product: Portable OpenSSH Version: 7.3p1

Hi Does anyone know a UK Voip Proivder that will give me more than 1 telephone number and point it to my sip account. www.SipGate.co.uk are great but they only allow 1 telephone number per user, you can register another telephone number by registering as another user but Asterisk doesn't allow multiple registrations. Many Thanks Scott

Hi all, I have a small system of two hardware boxes (residential gateways) running Linux with Asterisk on them. Each RG has some FXS ports to which analog telephones can be connected. I already had a working system including an external SIP provider, where both RGs would register to that provider with a telephone number and they could call each other via that telephone number. Each RG had a line

Here's one for you phone people.... An elderly lady phoned her telephone company to report that her telephone failed to ring when her friends called - and that on the few occasions when it did ring, her pet dog always moaned right before the phone rang. The telephone repairman proceeded to the scene, curious to see this psychic dog or senile elderly lady. He climbed a nearby telephone pole,

Hey Everyone, i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM 300,320,360 Devices. In the combination with asterisk and the patton, there are occuring some strange behaviour, due to the calling and answering everything works good, clear voice, great availability. All the devices have to use ulaw, alaw and slinear is available but never the first choice since i use my

This is my first attempt to write software of any sort. What I am trying to is to use a .php page to query asterisk Manager and get the ExtensionState for each particular extension. Then when it has the answer it outputs an XML file for use as the directory on a Cisco 7960 phone. What I am thinking is that when the user hits the directory button to veiw the directory that is at this URL it will

We have two servers but looks like G729 issues. Works fine on the old server and not sure if it is T1 related. See SIP Debug. Any experiences to share. Thanks --- Newark1*CLI> <--- SIP read from 194.xx.Xx.Xx:5060 ---> SIP/2.0 183 Session progress Via: SIP/2.0/UDP 76.xx.xx.xx:5060;branch=xxxxK784d2637;rport From: "Cell Phone DC" <sip:202xxxxxxx at

While the rest of you were chatting about the smallest * server, I was sitting her staring at the telephone hanging on the wall. It is a Western Electric set in a varnished pine box with an earpiece you hold in your left hand and a mouthpiece attached to the box. You crank the magneto with your right hand to signal the switchboard. The two dry cells inside are dated 1935, and I'm

I am brand new to Asterisk. My question is if I want to have multiple participants all listening, or listening and talking, do I need to have a separate telephone line for each, or can they all dial in using a single telephone number and a single line? Thanks, John Fistere -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Tue, Apr 05, 2005 at 08:26:45AM -0700, Ralph Giles wrote: > AM radio is lower quality (mono) but I don't know > what the digital equivalent would be. Just a minor nit-pick: AM radio can be stereo. However its use is almost nonexistent. See <http://users.hfx.eastlink.ca/~amstereo/amstereo.htm> for more information. > Telephone is nominally 8 kHz mono > (i.e. really bad)

Hi, Does anyone know where I can get a telephone card that will fit into the PCI slot on my PC and work with the UK telephone system (BT) ? I would really like the retailer to be based in the UK if at all possible ? Also, is there any way to set up asterisk so that only certain phones can make external calls, but all phones can receive incoming calls if they are routed to that phones by some

Hi , I am a newbie with Asterix and not sure if Asterix is a right tool for my needs. Let's suppose this scenario : I have a telephone line in one office( all calls are paid to telephone operator). In other offices I have only internet connections. Is it possible to use Asterix so that I can make telephone calls from ALL offices( without direct telecom connection) ? if so, what telephone

Hi folks, We're looking to integrate our telephone system with our email system. The telephone system will use IMAP4 to store WAV files in a users mailbox and then retrieve them for playing if necessary. This is usually called "unified messaging". The manufacturers are claiming full integration with Microsoft Exchange and Lotus Notes using IMAP4 and a single username and

Hope some can help me. I have a PSTN coming into TDM400 into Asterisk. We also have direct telephones connected to the PSTN bypassing the Asterisk. When a call comes in on the PSTN the direct connected phones ring first and if no one picks up , Asterisk picks and get routed to internal sip phones. I am not able to find what I should tune to make the calls always go through asterisk without the

Hi! I am doing a research about the prices of SIP telephones. If someone can tell me which one are the cheapest and have an acceptable quality... it will be very kind. Best Regards, Mireia

e164.org is a public name service which provides ENUM.164, a method devised by the IETF and ITU to allow an ordinary telephone to be connected to an Internet type network and provided dialling service from other, regular telephones. Unlike many other "free" voice over IP systems, e164.org allows users who have a regular telephone line, to also hook themselves up to the Internet

Dear all, I have an Asterisk box serving as gateway to a set of POTS phones that all share the same telephone number, and I register this box as gateway to a H.323 gatekeeper with the telephone number as the gateway prefix. I am wondering if there is a way to have Asterisk-wide (perhaps to be called 'universal') variables that can be used in all config files. As far as I know, now e.g.

Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode) with a public IP address. We have our phone system setup as 172.16.2.x that connect through the SonicWall to Asterisk. Incoming calls work flawlessly and we no longer get one-way audio. We are only using SIP (3 trunks now, instead of 2) and having all 3 in use is not an issue. Problem: Make a call on a Polycom 320 IP phone to