search for: telcos

I use a TDM400 card to interface with my telco. I used asterisk voice mail. However, if I'm on the telco line while another call comes in, obviously it cannot go to asterisk voice mail but instead bounces to the Telco voice mail. Is there any means by which I can get asterisk detect the Message Waiting Indicator (MWI) from the telco? Is there some application or variable which can be

Hi, I'm using Asterisk and a TE110P E1 PRI in Chile. When I call to a disconnected number or any not operational number, the telco sends the Hangupcause disconnection code and an audio message notifying the disconnection cause to the user. Asterisk does not allow the user to hear the audio message form the telco, instead it cuts the call. Any other legacies PRI PBX I've tested allow

I get an echo when going from a SIP phone to a PRI trunk. I hear the echo on the SIP phone. From reading some other post I think that I need to tell me phone company to turn on echo canceling. If the echo was on the other end than it would be my problem? Is this right? What exactly should I say to my phone company so they know exactly what I'm talking about? -- Michael Sampson

...nute. If this sort of stupid nonsense is allowed to start it will never end. What has become a world without borders, the connected will fast become the disconnected. Mr. Telco the Internet happened while all of you were asleep at the "switch" so wipe that evil smile off your faces. To me Telcos are nothing more than OED's ( One Eyed Droids )

Hello, I have two FXS port on my TDM card. channel 4 is attached with a telco line that I use frequently. And channel 3 have another telco line. but I dont publish that number to my friends. If I receive a call through channel 4, how can I handover that call to channel 3 ..so that I can keep channel 4 open for incoming call. Thanks,

Hello Everyone, Unlike going through DIDx, DIDLogic etc.., we have an option of getting DIDs directly from local telco Bell Canada. Currently our SIP Trunk provider assigned a DID to us, and as you know, they just redirect requests it to our PBX. However, when dealing directly with a telco, what equipment will we need? Basically giving us the same capability as a DID provider. If someone can

Hello users, I have a Digium Te235 and asterisk 13 which have worked well with 1 carrier but we have failed to add a 2nd carrier. The second telco brings their E1 line over finer, terminated in a RAD modem and they give me ethernet to the E1 card. It's the first time i am having install such a solution, which ideally would be not a big problem. However, The physical layer has failed to

Hello, I new to this, but with the help of mailing lists archives and IRC I am able to build my PBX. Thanks to all who had help me to reach till here. I am stuck at a point where I can't find the solution on mailing lists or even on IRC. I have individual 4 DID (Direct Inward Line) coming from Telco and terminating into TA 750 to FXS card. Many of them told that Phone instrument terminates

Are phone numbers portable in other countries? Are the same rules and conditions that exist here in the States mirrored elsewhere? How does a person in Europe go fully VoIP and still keep the main number? Do they use call forwarding? Is their another way to use an origination carrier without loosing your number? Alex -------------- next part -------------- An HTML

Hi all, We have been fighting with telco for a entire week. Today they came here with a LITE3000 to analyze what is going on. When I configure zaptel with no external clock, E1 gets aligned/synchronized with bit rate in 2048000 bps, both me and telco. span=4,0,0,ccs,hdb3,crc4 But when I configure span4 to get clock source from telco they become unsynchronized. TElco bit rate stays in

The Telco in Canada is been real painful. I was wondering if anyone has installed a Digium TE1X0P card in Canada and if their Telco was so difficult. The Telco will not provide us a service until they see a FCC or DOC number for the equipment ware are connecting to their service. If have found "FCC Part 68, ANSI/ITA-968-A, Including Amendment A1 and A2 Industry Canada CS-03"

Hi, I have a B400P BRI card with point-to-point connection (signalling: bri_cpe) with this dmesg: http://pastebin.com/sXrRt1yM When i restart asterisk server, the card cannot connect to the telco, the control led flashes red. If I unplug the cable between the ISDN nt and the card and wait 40 sec, the card can connect and works properly. The telco says the asterisk crashes the connection

Hi guys, I spent a couple of hours in Google, but the problem appears to be uncommon, so I'd like to ask about it here. The problem is exactly the opposite to "Asterisk does not detect FXO hangup". In my case it's the Telco who does not appear to be detecting Asterisk's hangups. Telco is Telus in Vancouver, Canada. The setup is very simple - Telco -> FXO/TDM400p

Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks to discovered this configuration and restricted them due to legal conventions, and stated that in order to continue doing this, they would have to talk

Hi I have a system (S) that has a PSTN gateway to accept incoming calls and setup outgoing calls from/to Telco network. In the other hand I have a distant Asterisk box (A) that I would like to connect to (S) using the PRI interface. I understand that the proper way is to order to my Telco two PRI lines one for (S) and another for (A), and configure (S) and (A) to call each other numbers when

Howdy I have a question regarding support for picking up when the telco sends a MWI message. My client's setup is a small office with three incoming lines on a TDM400P with iaxy's and a Grandstream as extensions. I am using CVS Head from yesterday. (I was resolving a different issue.) Since they only have two voice lines, with the third as a fax, I am using voicemail from the telco.

Hello, I just installed a PRI and when I make a local (seven digit) call, I get Code 28 back from the telco, (I believe code 28 means "Invalid Number") and I hear a fast busy on the phone. Here is the output: -- Executing Dial("SIP/marke-17b1", "ZAP/G1/4967171") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called G1/4967171

Hi all, Our asterisk PBX, randomly restarts all the channels of the E1 connection. It sends this message "There is no D-Channel, using channel 16 anyway".Then the asteisk recive (or it thinks it recives) yellow alarms at all the B-channels, after that it restart all the channels. When restarting the B-channels it cut all the conversations that is handling at that moment. Does anyone

Hi, I would like to know if one can set the outgoing caller ID within Asterisk when calls are going out through: 1) an analog POTS line (I suppose not) 2) a telco BRI line (I don't think so) 3) a telco PRI line (maybe) 4) a voip provider (surely) Thanks, Vieri ____________________________________________________________________________________ Moody friends. Drama queens. Your

I am trying to turn up DID trunks with our local phone company but do not know the correct format of extensions.conf to do this.