similar to: Asterisk with Primus Talkbroadband

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

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

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.

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

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 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

Hey all, Wondering if this is possible.. Incoming call is answered through X100P, then an extension is dialed using the same X100P card. Basically I want to dial in, enter 9 + <phone#> and have it do a flash then have it dial *08 <the same phone number> + # on the same PSTN line to have it transfer my call to another phone number. I realize this isn't very safe, but I would

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

Yes and No The D500 is a terrible thing First problem: it sends some horrible DTMF, so if your voicemail is configured to send #H and *H, it will not work, configure it to send numbers, like 8H and 9H (H is a placeholder for the extension). I also managed to use the MWI (message light), it's a perl script that is in voip-info.org, but with a little correction because the wiki distorted it. If

Hey all, I'm trying to get my Asterisk server up and running on fwd.pulver.com just to get the hang of it until I get my FXO card in a couple of days. It seems to connect but that's about it. If I try to dial into it from another fwd # it says user is not online. In sip.conf I have the following added: register => xxxxxx:xxxxxx@fwd.pulver.com/489125 [fwd.pulver.com] type=friend

According to this: http://www.redhat.com/support/wpapers/redhat/ext3/why.html ext3 is forward and backward compatible with ext2... Any user who wishes to un-journal a file system can do so easily... I am assuming un-journalling is the equivalent of converting it to ext2. How do you do this? I haven't been able to find anything. The reason I want to do this is so I can modify my

Hi, does anybody know if primus isdn cards - they support capi under linux, provided by own driver are usable with asterisk together with capi channel driver ? http://www.primuxisdn.de/primux/index.htm regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/

hi Can anyone recommend a good DID provider offering numbers in Toronto ? ( 1 very stable 2 support porting numbers from Bell, primus, telus.. ) Mario -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070702/cc39db9d/attachment.htm

Lamar Owen wrote: > On 11/28/2017 12:04 PM, Valeri Galtsev wrote: >> Thanks, Lamar! that is very instructive. > You're welcome. > >> I was always unimpressed with >> persistence of attempts to make more secure (less pickable) cylinder >> cased >> locks (precision, multi-level, pins at a weird locations/angles). > > The best way to make an

I have read over the archives and am still a little confused about mgcp support in asterisk. I realize the mgcp channel is server side only (but jump in and correct me if I got this wrong.) I have seen a few references to sip<>mgcp interteroperability in patches, so now I am wondering what the case is with support in the code. What I actually want to do is have asterisk act as the client

Hello, I have 2 Dell 1955 blade servers, running RHEL5-Xen. I''m testing the migrate functionality from one blade to another. I can start the domain, move it to one blade (minor delay/packet loss) and everything is fine. When I try to move it back to the original blade the migration fails and the DomU crashes c1b1 = Blade 1 (192.168.131.201) c1b2 = Blade 2

I am trying to install Rome: Total War (which should work fine it has golden mark on WineHQ) But when it starts to prepare for installation i get error 1603 and says that i have to consult windows installer helper or MSDN. I had similar problem before but then it was because I wanted to instal on unmounted drive noe this is not the case, so I really don't know how to resolve this.

-----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

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

Hi All I’m facing problem in bridging two SIP channels . I’m having SIP Trunk with Service Provider. Whenever I make any international call it get through and after some seconds it give error “Bridge stops bridging channels SIP/XXXX-3be3 and SI P/primus-9381” & call drops. May 13 17:25:19 VERBOSE[7491]: -- SIP/primus-9381 is making progress passing it to SIP/7125-3be3 May 13