similar to: TDM400P FXO and Primus TalkBroadBand

Primus in Canada has launched a SIP-based service to replace your business and residential POTS lines with a VoIP version. It's called TalkBroadband and it looks killer: http://www.primus.ca/en/residential/talkbroadband/index.html Basic service for $20 Cdn a month!! Local number portability!! Cheapo Primus LD rates!! They don't care where geographically you plug it in!! When you sign

I am sure Primus has a SIP platform because we have played with it. We managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2 hard phones. Their PC-Phone app is also a SIP soft phone. If you are registering to sip.iprimus.net then it is definitely their SIP platyform not MGCP. David >>> asterisk-users@eol.ca 1/21/2004 6:39:34 AM >>> I'm not sure Primus

If you look at the specs on the Dlink box that Primus gives you, you will see that it is SIP. I am sure Primus has a SIP platform because we have played with it. We managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2 hard phones. Their PC-Phone app is also a SIP soft phone. If you are registering to sip.iprimus.net then it is definitely their SIP platyform not MGCP.

Hey, I've checked all over and can't find what I need to know, so I'm posting here. I want to use Asterisk with my Primus VoIP service but it seems I need a username and password to authenticate with at Primus. Has anyone had any experience with this? How did you get it? Is it stored somewhere in the D-Link gateway they gave me? Thanks in advance and sorry if this makes no sense.

Hi all, I curious to know if other UIP200 users have this same issue: You flash (XFER button) to consult-transfer a caller to another extension. If the transfer target party is unavailable (ie: voicemail), there appears to be no way to get the original caller back. If it's a known limitation, has anyone come up with a functional work around? Thank -- ..................................

I couldn't be happier with the simplicity of this - but it's a hack! Hi all, I'm currently using a SIP client (BT101) to connect via DSL to a remote instance of Asterisk. - Asterisk has a private IP behind my OFFICE router. - The SIP client has a private IP behind my HOME router. I'm doing this _without_ the use of STUN or proxy servers. Here's how it works: -

Hi all, We have an analog line from telco, on which 3-way calling is subscribed to. This line is plugged into an FXO module on a tdm400p. If an incoming call comes in on this line, can */zaptel send Flash to telco via the FXO module? If it could, then an incoming call could be 'transfered' to a cell-phone, for example, with a single analog line. (where 'transfer' is really

Hello, Over the past several weeks, we have been having an intermittant problem with our deployment of a TDM400P card (4 fxo). ?We have tried many things, and the problem still re-occurs. The Problem: Occasionally (every 48 hours), the TDM400p card will stop answering incoming calls on ALL fxo ports. ?Attempts to send outbound calls on any Zap channel will result in hearing a loud

Hi, We've recently deployed 6 Uniden UIP200 phones (running firmware 4.54). We've been having some serious problems: 1) All the phones randomly reboot themselves. Typically when trying to answer or initiate a call. 2) All the phones will disconnect from a calls with the PSTN after 2-3 minutes. 3) The phones are unable to interact with a remote IVR (digit presses are not received at

Hi, We are using Vicidial and sometime even when agent disconnects, outgoing call originated by dialer is still active. Since call was initiated by dialer and then bought into meetme conference of agent and we can't corelate this call to any agent channel. When agents are dialing, channels doesn't show calls vicidial2*CLI> show channels Channel Location

I easily get confused when try to undertstand FXO & FXS ports. Is it possible to use an ATA to connect to a TDM400 card. If so, would I use FXO modules or FXS modules ? My goal is to connect my asterisk server to Vonage (via the ATA they send me) so I can use thier standard plan and do with out the Softphone account feature that only allows a few hundred minutes talk time. Thank you, Steve

Hello all, The "Secret Agent" final release of the Asterisk Management Portal is now available for download: http://amp.coalescentsystems.ca/ This exciting new release adds a great deal of functionality and flexibility. Thank you for all the contributions and feedback! 1.10.007 - Added AMP Users (multi-department, basic multi-tenant) - Added incremental upgrade script

Hi all, I've run into 2 separate problems relating to fax: 1) Using tdm400p + fxo, Asterisk is unable to detect the fax from some fax machines (from others it can). Using zap barge, I can confirm that these troublesome calling fax machines _do_ send the fax tone loud and clear. Are there certain circumstances where I should expect a Digium card to fail in detecting a fax? 2) Using

Hi all, Several months ago we built an * box with a quad-FXO tdm400p (REV e/f). >From the get-go, there has been a problem where occasionally (2-3 times a week) zaptel/* will not detect the ringing on a line. (The call will ring through to telco voicemail). The problem is not specific to a single line or FXO port on the tdm400p. I have 2 theories: #1 - the ring voltage for some calls is

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I am new to this list. Pardon my dropping by like this. I regularly use VoIP devices for residential/small office use, so far involving small setups (1-3 devices) like the ATA-286 or S2K. I also have Primus TalkBroadband service and have been experimenting/evaluating offering some options to my customers. I'd like to discuss a project for

Hello all, Asterisk Management Portal 1.10.009 has now been released. This exciting new version has several notable additions (listed below). The AMP homepage is http://amp.coalescentsystems.ca. Here you'll find links to the download, install guide, and documentation wiki. As usual, please use amportal-users mailing list for discussions about AMP:

Hi list, For a client I am setting up a system which will use T1 PRI from Primus, who offer only NI-1 and NI-2 protocols for D-Channels. Previousely I have only used switchtypes euroISDN and National. Although the documentation says Asterisk does support NI-1 ans NI-2, but wanted to get your opinion if you have used these protocols on an Asterisk box and if there were any things to consider. If

Ryan, from the console what does "zap show channel 1" or 2/3/4 in your case say. I have X100P's and I seem to be having similar sounding problems. I noticed that the above command shows the channel to be off-hook at all times when a phone line is plugged in. I don't know why or if it is a bug in the application reporting the status. dbc. Ryan Courtnage wrote: > On July 8,

Hello, We are using a TDM400p with 4 FXOs and SIP phones in a high call-volume environment. At least twice a day there are complaints of 'dropped calls'. Examining the debug logs, I see that in each case, an "on hook" event is detected, followed by the zap channel being hung-up and * saying "BYE" to the sip phone: Jun 23 14:17:22 DEBUG[2441232]: Exception on

Mon Dec 25 20:19:05 GMT 2000 Greetings. I noticed that in OpenSSH_2.2.0, DSA keys were allowed to be added to ssh-agent, however the ability for allowing ForwardAgent does not yet seem in place for protocol-2. I've noticed that when using protocol-2, no socket is created in /tmp/ssh-*/, and consequently SSH_AUTH_SOCK is not being set. Hence the ability to ssh to another machine (using