similar to: DID providers in Toronto

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.

Outbound calls on my Telus PRI aren't taking the Name portion of the callerID. I've looked at the logs, and it is being set (see below), but the PRI debug output doesn't show the name being sent anywhere. As a result, received calls always display from Unknown (or just the number). Is there some config that I've missed somewhere? I'm running NI-1 (Telus says NI-2 doesn't

I'm looking at switching VOIP providers, but want to ensure I move to a company with sufficient capacity. Can any Canadian VOIP users post/email me with feedback on their providers? I'll post the results for all to read...... -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hi all, A while back, there was a short thread on using the FXS interface from a Primus TalkBroadBand device (a DLink ATA) as a incoming line for the FXO interface on the TDM400P: Primus <--> DLink ATA FXS <--> TDM400P FXO <--> Asterisk In that thread, a couple of people suggested that this does not work reliabley, and the ATA FXS <--> TDM FXO link 'goes

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 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

Hello All, As you can see below I am having a problem checking for updates. This happens repeatedly. I have to kill the process then rerun. I have tried "yum clean all" but no joy - the process hangs again on "extras" - see second listing below. Suggestions? Dave [root at cserver ~]# yum check-update Loaded plugins: fastestmirror, priorities Loading mirror speeds from

Hi, We ran into a little problem recently with our phone provider (Telus Canada): we are unable to dial numbers outside North America. This is what happens: the phone number 011... is sent out over our T1, Telus sees the correct number on their switch. However the switch thinks it's a North American phone number (and thus has to have 10 digits) and rejects the overseas number as

My apologies for the cross-posting. If you are a business or individual providing Voice over IP services in Canada then we encourage you to read this email carefully otherwise please disregard. ----- As you are most likely aware, the CRTC has undertaken the roll of regulating VoIP services in Canada and is currently conducting hearings with the goal of putting in place regulatory requirements

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

Config: AAH 2.2 Digium TDM card connecting to 3 x Telus POTS lines Polycom 501 phones pretty basic setup, working mostly just fine... When I dial a number such as: 96045551212 Telus automation will sometimes come online and tell me that the number I have dialled cannot be completed as dialled. If I hang up the Polycom 501 and redial the EXACT same number, it will work the second time. I

I just got the list confirmation and noticed it''s text only email so here it is again in plain text. Below is the oringal message. Hi all, I am really struggling with this one, I have built a lot of linux machines using IPSEC tunnels and shorewall gateways. I decied to build a new test machine with Debian running 2.4.25 and Shorewall 2.0.15. I have two subnets on their own switches and

I've been losing patience with my current provider, a small company called Sellvoip. Their termination is good, and they are asterisk based, but they are understaffed and have no concept of customer service. So I'm shopping. I am interested in the opinions of others on the providers they work with. Here are my criteria, roughly in order a) Decent quality, low latency. In

Has anyone has issues with echo using a Wildcard with a PRI from a major Canadian Telco? (Bell, Telus, AllStream, Sprint, Group Telecom). We are using a T1 from GT that is giving use annoying echos whenever a SIP/IAX2 client calls a local analog line. Calling cells phones is no issue since its digital. Regardless, there should be no issue with echo on a PRI at all. NOC at GT is telling us

We are trying to send caller ID NAME information over a Telus PRI in Alberta. The PRI tech says that he sees the NAME information, and for calls over the same network, that NAME info should be reaching the receiving station, but it is not. The technician was stumped. I suspect there's something specific that I need to do to make it work, since many PBXs can do this. The switch is a

Annoying :( S. -----Original Message----- From: Mail Delivery System [mailto:postmaster@bembang.com] Sent: Thursday, September 02, 2004 3:06 AM To: stormp@telus.net Subject: Mail delivery failed: returning message to sender This message was created automatically by mail delivery software (HiveMail). A message that you sent could not be delivered to one of its recipients: sysad The error

Hi, Red Hat 9.0 Asterisk 1.2.7.1 Whenever I start Asterisk, I am unable to call out on my SIP channel: >-- Executing Dial("Zap/1-1", "SIP/6137451576@6477235412||t|") in new stack >Apr 30 11:02:00 WARNING[12814]: chan_sip.c:1973 create_addr: No such host: 6477235412 >Apr 30 11:02:01 NOTICE[12814]: app_dial.c:1029 dial_exec_full: Unable to create >channel of type