similar to: xp100 not hanging up after call disconnect

Wanting to connect my asterisk box off of 2 unused analog extensions on the non* PBX system. Can I bring those lines in to * on XP100 FXO cards? Any special wiring/open loop issues to watch for? The desired config is inbound PRI -> non* PBX, attendant picks up and if caller wants services on * box then xfers call to one of the analog extensions where * picks up and handles call from that

All, I have been thru the archives and all the relevant URL's sent to me. I have sent e-mail to those who have gone before me and are attempting to accomplish the same goal - no one has it working?. Doesn't anyone have a WORKING asterisk pbx that hooks into vonage? Thanks, Jerry Roy 562-305-9545 -------------- next part -------------- An HTML attachment was scrubbed... URL:

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.

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

Has any one found any issues (bug?) with ver 1.2.5 as CallerID generated on an outgoing FXS port (to the handset) fails when UK tones are used, with a message 'Didn't finish Caller-ID spill. Cancelling.' Any tips on getting this running ? Looked at Mantis, but only known bugs seem to relate to XP100 cards - not the TDM card. Thanks

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

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

Has any messed with getting Asterisk to work using the Voice Pulse Open Access plan? I currently have 2 numbers with Voice Pulse, 1 is a number that is assigned to their hardware device (Sipura SPA-2000), the other is a Open Access number that uses SIP from any device (you must authenticate with them). I want to be able to use the Open Access number on my Asterisk server here at home with no FXO

Hi John, I've been using Packet8 via a physical ATA and XP100 card for some time. As far as I know it is not possible to connect to the Packet8 service without the ATA. If this is not the case I would be very interested to hear this. In addition since moving to the USA I now only have a single packet8 line into my asterisk box (I used to have this and a 2nd regular pstn line) I used to be

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 have a snom 200 phone here which works perfectly when using the handset to playback the voicemail messages etc. However when I play back the voice using the speakerphone it sounds choppy. Anyone had this problem before? Regards Mark

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

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

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

Hi All I facing problem in bridging two SIP channels . I 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"#12539;& call drops. May 13 17:25:19 VERBOSE[7491]: -- SIP/primus-9381 is making progress passing it to

hello, i have a config that is with a large domain of rate, from 2kb to 40Mb and i have some problems with i don''t know how to deal with. so here are some classes 2 with q=1000, one with q 200000 and one with q=6400, all have been calculated by htb, in the code i speficien just rate,ceil, no quantum, no r2q, no bursts ($tc class add $IF_INT parent 1:2 classid 1:21 htb rate

I am trying to dial a mgcp extention from my sip phone and i am getting this error message. anyone got any idea? error I> -- Executing Dial("SIP/2204-5dc2", "MGCP/aaln/1@10.0.1.150") in new stack May 19 22:30:01 NOTICE[1251156800]: chan_mgcp.c:1104 find_subchannel: Gateway '10.0.1.150' (and thus its endpoint 'aaln/1') does not exist May 19 22:30:01

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, Does anyone know what this means? I was running asterisk fine. Installed it on a new pc and I am using the g729b. codec that is optional. I ran the install for the codec it went ok but when I run askterisk via asterisk -vvvgc it gives me this error anyone know? I make sure I entered in the correct reg number. I followed the steps correctly. Too Registration error! Please try

Hi all, I just saw this article about this new offer from Lingo.com: http://www.techweb.com/wire/story/TWB20040607S0008 $20 monthly plan with unlimited local and long-distance calling in North America (US & Canada) and Western Europe. Plus first three months free and free equipment. It doesn't say what hardware they send you. Sounds like a very good deal. I searched the list and