similar to: Sip phone with push display?

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

Hello - I have (finally) obtained a fair amount of success in connecting my Intertel Axxess PBX to an asterisk box via a T1 PRI. I can place calls from the Intertel side through the T1, out to an IAX2 softphone and the calls get routed correctly and all of the CID information stays intact. However, when I call from the IAX side to an extension which should route back through to the Intertel

I've downloaded the entire archive of articles and searched through them for an answer on this, but I haven't come across one yet. I'm looking to replace a small phone system in my church with Asterisk, and I'm stuck looking for phones. I know that the staff are going to want a button for their commonly-called extensions, but I'm having trouble finding phones that have, say, 10

Have the Sayson IP phon started to deliver yet? I'm thinking about two new phones for my office and considering the Sayson 480i and Zultys 4x4. Would also consider the Virbiage phone if it becomes available. I have Snom 200s and a Pingtel phone at the moment. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist

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

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

I have a strange problem with my Sayson 480i IP phone. If I press the Transfer button and then dial extension 200 to try to transfer the call, the Sayson apparently is treating the 200 as the last part of an IP address, and the call fails. As soon as I enter 200 and press Dial, I see an IP address of the form x.y.z.200 show up on the display. How do I get the phone to treat the dial string as an

We have a customer with a handful of Sayson/Aastra 480i phones behind a Juniper Networks Netscreen firewall registering with our hosted PBX service. The Netscreen monitors the REGISTER messages and only keeps the reverse mapping open for the duration of the registration period. It appears that every so often the Sayson does not send out another REGISTER message after the registration has expired

For those that are interested, here is my report back to Sayson on the 480i ---------- Forwarded message ---------- Date: Sat, 31 Jul 2004 22:03:31 -0400 (EDT) From: xxx@bgcfreedom.com To: xxx@sayson.com Subject: 480i User Feedback With Asterisk Seshu, I am using a 480i, and I am impressed with the phone on a whole, but obviously the firmware is lacking. Details follow. Hold button works, but

There isn't much documentation on adsi, but I called NETXUSA (the vendor of my 480e) and they helped me along. My experience: 1. I really had no experience with ADSI so I had (probably still have) some misconceptions on how the configuration is loaded onto the phone. 2. I set the following in my /etc/asterisk/asterisk.adsi (most of this is the stock asterisk.adsi script): ;

I am currently a new asterisk user I have worked with the old rolm systems in the past. I have been asked to look around and find out how to do a few things in asterisk, either in asterisk itself or with third party software. The features that I am looking for are: 1. A good management application for setup and edit the .conf files for both the exten & voicemail. 2. A receptionist software

Does anyone have any recent news on the Virbiage FT201 IAX Hardphone? I'd *really really* like to deploy these phones instead of SIP hardphones, and I can't help but wonder if I'm going to shoot myself in the foot (or another sensitive area) by deploying a ton of SIP phones just to find the IAX Hardphones were released a week later... Thanks, Brian D'Arcy

I have a couple of Polycom phones, bootrom 2.5.0, SIP 1.3.1.0056. Works great with Asterisk when I power on the phone. However, after some time, say an hour, I cannot receive calls on this phone. On Asterisk, when I do "database show", it does show the phone in there, but it cannot reach the phone. Is there a way to keep alive the connection with Polycom? Any suggestions? Thanks, --

From: "Guillermo Salas M." <gsalas at manta.telconet.net> > http://www.gizmo5.com/opensky Free calls are available up to 5 > minutes. If you need longer calls there's a commercial service you can > purchase. > Can be used to receive calls from skype? Yes it can. For example anyone who calls me now on Skype at michaelGizmo5 it will ring the IP phone connected to

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

Does anyone have success in upgrading Aastra/Sayson 480i CT firmware? All I get, no matter what I've tried is "Unable to upgrade firmware". tftpd is working because the dialplan freshens, and I have aastra.cfg whatevermacaddressfile.cfg and firmware.st in /tftpboot Am I missing something stupid? Is there another way to upgrade it? Chris Coulthurst chris@shuksan.com

Hi all, I'm experiencing a problem with SIP channels going ZOmBIE after the following sequence of events: - IAX2 client calls SIP client - SIP client consultive transfers (using sip REFER) the call to a MeetMe extension, and hangs up. At this point, the IAX2 client will indeed be in the meetme room, but a 'show channels' at the * CLI reveals that the SIP channels that were involved

Hi all, I want to setup a meetme application in the following maner: One operator is connected to a room. The operator hears and can talk to all the participants, but one participant can only hear/talk to the operator, not others. The operator is using one phone. To be more explicit, this means that every new person etering the room has a one2one conversation with the operator only, and the

I'm looking at the Sayson 480i or the Cisco CP-7960. Which one would you suggest and why please ? Best Regards, Alex Brecher Visit us at http://www.Successfulhosting.com <http://www.successfulhosting.com/> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041128/8e282c51/attachment.htm

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