search for: telephon

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) ------------------>71(5060)1U telephone-event) 0.09 (5060) 100 Trying-------->71(5060)1U telephone-event) (5060) <------------------71(5060)1U telephone-e...

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, GNI.per.capita..Atlas.method..current.US.., Telephone.lines..per.100.people., Internet.users..per.100.people.)) MDGdatasetAdapted MDGd...

...;>>'''';l;skype tech support number skype ('866.769.8127 ?>>>>>'''';l;skype tech support number 769 8127 ?>>>>>'''';l;;;;;skype tech support number 866 769 8I27 Skype .... ++1-866-769-8127 SKYPE Customer Care Telephone Number ... skype ('866.769.8127 ?>>>>>'''';l;skype tech support number skype ('866.769.8127 ?>>>>>'''';l;skype tech support number 769 8127 ?>>>>>'''';l;;;;;skype tech support number 866 769 8I2...

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 register => <telephone number>:<password>@sip.my...

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...

...on hold. In the SIP Debug i found some things which i cant handle, so i try to ask you whats going on there : The call comes in, the patton routes it to asterisk and the codec invite starts : --FROM PATTON TO ASTERISK-- Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x14c (ulaw|alaw|slin|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) The last line is mysterious to me. --AST...

...AND EACH CASE CORRESPONDES TO THE VALUE FROM THE STATUS AND PRINTS THE XML ACCORDINGLY { case -1: print("\t<DirectoryEntry>\n"); print("\t\t<Name>"); print("$name is Unavailable"); print("</Name>\n"); print("\t\t<Telephone>"); print("$phone"); print("</Telephone>\n"); print("\t</DirectoryEntry>\n"); break; case 0: print("\t<DirectoryEntry>\n"); print("\t\t<Name>"); print("$name"); print("...

...ver: (Very nice Sip Registrar/Proxy Server) Allow: ACKBYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 198 v=0 o=xxxxxx 12xxxxx 12xxxx IN IP4 62.xx.xx.xx s=SIP Call c=IN IP4 62.xx.xx.xxx t=0 0 m=audio 8786 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 <-------------> --- (11 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 62.xx.xx.xx:8786 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x106 (...

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 throwing them...

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: http://lists.digium.com/pipermail/asterisk-users/attachments/20050304/fd6c7a4b/attachment.htm

...alph 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) though I think the use of digital voice codecs in the > last 20 years may have improved on this a bit. Telephone lines (POTS) have a frequency range of 300-3400Hz. That means 7kHz mono should be enough, although 8kHz is generous towards the trans...

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 incomin...

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 equipment would they have to use (VoIP t...

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 No...

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 di...

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 without intervention from the...

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. in extensi...

...fter 30 seconds. Log snippet: -- Executing [s at macro-dialout-trunk:19] Dial("SIP/203-b7a2b558", "SIP/bw_outbound/+18005551212|300|") in new stack Audio is at <public IP> port 11968 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 216.82.224.202:5060: INVITE sip:+18881231234 at 216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP <public IP>:5060;branch=z9hG4bK6ea30a1a;rport From: "8881231234" <sip:+18881231234 at public IP>;tag=as3ed791f3 To: <sip:+18005551212 at...