similar to: SIP phone w/ XML browser

Drive space for voice mail I've looked in the dimensioning information on voip-info.org but can't find any hard information on the amount of drive space the various codecs use. Since we would eventually like to support web-based voice mail retrieval, I'm thinking of the wav format. I've specced out 2x160GB drives in RAID-1 (software RAID via Linux) for the box. It will be

Hi, I have a bunch of NetPhones that I have bought from MeritCall some time ago for their service. How can I use this phones (supposed SIP phones) to integrate with a Asterisk Setup. I have seen a manual for a similar one but I don?t know If mine are hardcoded in some way. This devices are used by MeritCall , MamaKall, Vivophone etc. I found the manual for an exact same type here:

Has anyone been able to get the minibrowser on the Polycom SoundPoint IP 500/600 phones working? If so could you share the relevant sections of your config with me?

I'm trying to develop a company phone list accessible via the minibrowser feature on the phone. The pertinent section of ipmid.cfg is as follows: <microbrowser mb.proxy=""> <idleDisplay mb.idleDisplay.home="http://server/polycom/index.html" mb.idleDisplay.refresh="300"/> <main

I've configured SIP softphone to work with asterisk n it's working fine..... but i m unable to connect my netphone IP phone.... I've connected my phone to LAN and assigned an IP address to it.... but how can i make call... plz tell me step wise..... _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today

Hello to all, I'm trying to build a PBX using Asterisk. I have a single BRI ISDN line and I need to connect 4 internal normal phones and a couple of softphones on PC. I have bought a single port Billion S0 card and a TDM400 with 4 FXS modules for the intenal phones. ISDN lines here in Greece terminate to an NT1 terminal called NetMod

Hi all, we are using samba 2.2.6 with windows xp SP1, and roaming profiles. We would like to have email settings (servers/usernames etc) with the roaming profiles, but outlook (either microsoft, or express) store their settings in a path that is not included in the profiles we do not care about the actual emails, since the mailserver is IMAP, we only care about settings Has anybody

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

Hi all, I have 'Voipac NetPhone 210' phone apparatus with

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

i read that asterisk supports iax,sip and h323 protocols.... i've used sip & iax softphones ... now i've a hardphone... an IP phone (Netphone) that supports h323 ..... i've compiled pwlib ,oh323 and asterisk -oh323 successfully ... but i m unable to place calls to/by my phone... i m confused whether to use h323.conf or oh323.conf and how ? i think it's different from iax.conf

Hi all, I am trying to run some performance tests using libvirt 1.2.17. For these tests it is necessary to set a specific affinity for the VCPUs of a running VM. The problem is, that I noticed a significant performance drop in my metrics when using 'vcpupin' from the virsh console instead of running 'taskset' natively. Specifically the methods I am comparing are 1.

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

Hi Folks, I have posted a couple of message to the list and do see them, even after waitin for long time (2 days). Can someone please point me to the rules for posting to this list? I think it had to do with the subject that I was looking for. I was looking for IAX terminiation service that can handle high volumes. Thanks John. --------------------------------- Yahoo!

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

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

I'm currently testing these phones and so far I'm very pleased with them. I'm using the SIP image (Version: 1.0.0.23 Release: 0032-01-A4). There are a few issues with the speaker not working properly when you dial out or pick a call but this is a known issue and will be hashed out in the next release. I've been working with the developers at Sayson and they have been very