similar to: Inter-Tel Eclipse2 (IP PhonePlus)

I've been trying to figure out if it's possible to connect Asterisk to a parent Inter Tel Axxess system through the MGCP protocol. The archives for this list aren't searchable and I'm wondering if anyone has a simple answer... Dustin Moore

All, I am trying to configure hdlc support on T100P Digium card - everything seems ok... but it just does not work at all. I was able to compile all drivers, the light on the card becomes green when I plug T1 link.... but I even can't ping default router IP... there is no data coming back to me, so I am confused completely now. Also I tried both protocols - raw hdlc and cisco hdlc. The

Has anyone successfully integrated Asterisk with an Inter-tel Axxess phone system via a SIP trunk using the IPRC card? -- Marvin Horst -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101006/5dbe435a/attachment.htm

Hi all, My office recently purchased an InterTel Axxess system with the IPRC card for VoIP. To our suprise, this card allows the InterTel endpoints and MGCP endpoints to work, but not SIP clients. I was really expecting to get a SIP softphone working with this setup, but that appears to require our vendor to sell us a SIP gateway and licenses at a not yet determined price. With this

-----Original Message----- > -this is very true, however, the current version of the Axxess software > (9.0) supports SIP trunking natively on the IPRC. I just got my Axxess > upgraded and am salivating to get * connected to it. Hmm, so 9.0 is out and it supports SIP natively. How did you plan to integrate the 2? -The Axxess will see the * as it would see an IP service provider.

Hello all, We have a TE405P set up with span 1 running to a channel bank, a PRI running into span 2, and a PPP internet T1 running into span 3. We have the first 2 spans up and running without a problem. We have hdlc compiled into the kernel and after making the appropriate changes to zaptel.conf and loading the zaptel, wct4xxp, and hldc modules we can bring up the third span with the internet

When I start * I get 6 mpg123 processes start as well. Is this normal? Often after a couple of days these mpg123 processes start to consume cpu and I have to kill them off. I do not have a sound card in the server and I have noload => chan_oss.so Simon

On Wed, Aug 03, 2005 at 11:28:19AM -0500, asterisk-users-request@lists.digium.com wrote: > 10. Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary > D-channel of span 1 (Gavin Hamill) > Date: Wed, 3 Aug 2005 15:32:48 +0100 > From: Gavin Hamill <gdh@laterooms.com> > Subject: [Asterisk-Users] Inter-Tel AXXESS failure: HDLC Bad FCS (8) > on Primary D-channel of span

Yep, another list posting on this topic :) All the messages I've read on this are from people experiencing these errors in quiet times - I get them as soon as I plug a port on our TE410P to an Inter-Tel AXXESS PBX.. and I get them continuously... I'm just sticking an * box in between ISDN30e (we're in the UK so euroisdn) and the PBX.. and whilst the telco ISDN30e side works like

On 12/12/2018 03:32 PM, Steve Clark wrote: > On 12/12/2018 03:28 PM, Gary Braatz wrote: >> Thanks for responding so quickly! No but I will try. Are you saying the >> first vendor connection worked because id_rsa and id_rsa.pub are the >> defaults if not specified? (I didn't use the -i flag for the first vendor.) >> >> >> -----Original Message-----

On Wednesday 03 August 2005 17:33, Jens von B?low wrote: > Gavin, > > >> Any ideas/advice would be warmly received right now! > > You are not going to like my response... Erk :) > The only way I could get this to work (luckily I had 2 identical sites and > was busy with the upgrade to the gen2 card) was to downgrade to zaptel > 1.0.7. Alas no - just moved down to

Is it possible to act as a middle man on a T1 line? My installation currently has an aging Inter-Tel Axxess box with a T1 coming in (16 in, 8 out). Rather than adding and replacing phones and cards as they die, I would like to slowly migrate to a asterisk SIP installation. I want to take the incoming T1 line, use any available outgoing lines for outgoing SIP, intercept any incoming lines and

Thanks for responding so quickly! No but I will try. Are you saying the first vendor connection worked because id_rsa and id_rsa.pub are the defaults if not specified? (I didn't use the -i flag for the first vendor.) -----Original Message----- From: CentOS [mailto:centos-bounces at centos.org] On Behalf Of Steve Clark Sent: Wednesday, December 12, 2018 2:23 PM To: CentOS mailing list

Hi everybody, Well, I've finally got asterisk to to talk nicely with my Intertel pbx. Currently there is a outside T1 line (e&m wink start, esf, b8zs) connected to asterisk, and then asterisk connected similarly to my Intertel pbx. For right now all asterisk is doing is passing calls between the two. When I call out from the pbx, I can connect perfectly to the outside world. When I

I'm new to SFTP and using this mailing list was able to successfully create my first Private/Public keyset for a vendor hosting the SFTP server (I'm the client). I created the keyset by typing this: # ssh-keygen -t rsa When asked for the password/passphrase I hit <Enter> and afterwards "id_rsa" and "id_rsa.pub" were created in "/root/.ssh/". I

Any Recommendations on a "Good" Wireless Voip Keyset that works well with Asterisk? I would prefer one that is IAX2 as it works better behind a Nat'd Firewall.. But I am reaching out to you guys as you all would know what would work the best :-) ________________________________________________________________ Sent via the WebMail system at kotbh.net

Greetings all - I'm interested in using an asterisk box to supplement and add VoIP capabilities to our legacy InterTel Axxess PBX. After searching through the list archives and through google, it seems that the best way to go about this is to connect the two systems via a T1. Is this correct? The PBX currently doesn't have any VoIP capabilities, so that's not an option for

We're setting up a connection with Level3's voip system and when we use Asterisk or make or recv calls we get an initial pulsing noise. Level3's Interop team explains that's their RTP extension headers and Asterisk apparently doesn't know what to do with it. He said we need to either ignore or of course let the traffic pass. Has anyone heard of this before? I understand the

Additionally to the problems I reported earlier, I'Ve discovered another problem with my server/client setup. find reports find: WARNING: Hard link count is wrong for ./foo: this may be a bug in your filesystem driver. Automatically turning on find's -noleaf option. Earlier results may have failed to include directories that should have been searched. in one directory and if I

Hi, this email is sent to flac-dev AT xiph DOT org, copied to matroska-devel AT lists DOT matroska DOT org, and David Bryant, the author of Wavpack4. We are about to implement the DV ( Digital Video ) video compression standard into matroska, to be able to offer an intelligent alternative to AVI for DV handling in editor tools. DV is using a high quality, high bitrate lossy video compression