similar to: Problems on SIP gateway (especially Planet VIP-450)

Hello, I just received what seems to be a nice SIP<->DECT gateway but can't make it work with asterisk. The manual is very unclear (written in "chinese" english) and the web configurator is ambiguous as well. Has anyone succeeded in making one of these babies work with * ? info: http://www.planet.com.tw/product/product_dm.php?product_id=367&menu_id=3

Hi everyone! I want to know if anyone has the sip gateway VIP-450FO from Planet (www.planet.com.tw). I?m looking for his firmware because I would like to transform my VIP-400FO (H323) in a VIP-450FO (SIP). Does anyone has this firmware to send to me? Thanks, MCelo.

Hi All, I'm having problem with this phone. Problems are regarding voicemail message alert on the phone. --- handle_response: Host 'xxx.xxx.xxx.xxx' does not implement 'NOTIFY' --- Can somebody help? On the phone manual, is written that it can acept MWI, but... not mine!!! Thanks! -- .:FaberK:.

Hi, I want to extend my asterisk stuff and buy some Planet devices, to be certain I'm going to buy PLANET VIP-050 with FXO and FXS modules. Has anyone heard about it. Is it compatible with Asterisk, or it would cause a lot of problems. Dose anyone have some experience with it?? All the best Andrutto ---------------------------------------------------------------------- Oferty sprzedazy

Does anyone know how to connect the planet VIP-152T phone to asterisk because I can't get it to work at all. What configuration is needed? I have a asterisk@home setup and works perfect with the soft phones but not with the planet... I 'm a newby so help me pleaseeeeeee ____________________________________________________ Start your day with Yahoo! - make it your home page

I am attempting to get a planet VIP-150T to register with asterisk 1.2.4. After searching google I've found what appear to be instructions in German, Russian and Spanish. Has anyone perhaps seen this before? Asterisk is kicking back the following error: Feb 14 09:59:32 NOTICE[21765]: chan_sip.c:10851 handle_request_register: Registration from '<sip:101@192.168.100.240>'

Hi, right now I am using only one external server as a gateway for the internal servers. I would like to enable a fail-over on a second server. To implement the floating VIP, should I use heartbeat+pacemaker? Or is there something more "lightweight"? Basically, I just need server2 up the VIP when server1 is down, and server2 down the VIP when server1 is back up (or server 1 does not up

Hello everyone! I am working on implementing my first Gluster/Ganesha NFS setup and I am flowing this guide: http://blog.gluster.org/2015/10/linux-scale-out-nfsv4-using-nfs-ganesha-and-glusterfs-one-step-at-a-time Everything is working fine. I've got Gluster and Ganesha NFS working and I have VIPs on each node and it is failing over fine, if a little slowly. However, the VIPs don't

I have gotten some cisco VIP 12 and VIP 30 IP phones that I would like to use with asterisk, I have set them up using chan_skinny. The phones work well, except the only problem is that it is like the cisco phones are muted. When I talk on the cisco phones I can hear my self through the ear peice, but the person who I am calling can not hear me at all. I have tried various cisco phones from various

Hi, I am using a 30 VIP phone and a 12 SP+ phone with Asterisk. When I complete a call outside through the ZAP device, the phone does not go back to dial tone, even after I hang up. The line gets disconnected as per Asterisk console. But the phone stays in the same state like it is connected. The ZAP line is freed up and I could make calls from other phones. Only this phone just remained in

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm just learning about VoIP and Asterisk. I've got a developers kit on its way and I've managed to get hold of a couple of cheap Cisco VIP 30 phones. I've trawled the web and found a few snippets of information on these phones but I still can't get them to work. Does anyone have any config files or any idea on how (if I can)

Hello all, I post this issue thinking too that could help other people on an asterisk deployment over distributed offices considering both quality, prices, devices and so. Well, i am working on a deployment of a telephony system based in asterisk. My company have a central office with seven remote offices connected all through a VPN. To reduce and evaluate costs i consider solutions like:

Does anyone know a good (and stable) voip gateway product with 4 ports (2 fxo and 2 fxs), with the following requirements: * being able to connect analog phones to the FXS ports, and communicate over SIP with an REGISTRAR/PROXY server (SER in our case). * being able to connect the FXO port to local office PSTN network, and dial to that office pstn number and getting an internal dialtone, or

Some of the Red Hat Enterprise Linux servers in our environment sometimes get assigned virtual IP addresses (eth0:1, eth0:2, etc). Puppet Dashboard''s parameters seem like an ideal way to define and provision virtual IPs on these servers. One could create a "vip1" parameter on a node, and define a value of the IP address to use for that vip. A manifest could then be written to

Are virtual IPs supported on eth interfaces in domU(s)? Thanks _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

Hi all, thank you very much for support! I filed the bug: https://bugzilla.redhat.com/show_bug.cgi?id=1457724 I'll try to test it again to get some errors / warnings from log. Best regards, Jan On Wed, May 31, 2017 at 12:25 PM, Kaleb S. KEITHLEY <kkeithle at redhat.com> wrote: > On 05/31/2017 07:03 AM, Soumya Koduri wrote: > > +Andrew and Ken > > > > On

Hi there, I know this was discussed many times already, but after trying to get my 30VIP to work for a week, I think I might be doing something wrong. I'm trying to set it up with chan_skinny. Haven't tried chan_sccp yet, but since there are many of you already doing it with skinny, it should be working also in my case. I read and used all the hints and samples I was able to find on the

Hi all, I love this project, Gluster and Ganesha are amazing. Thank you for this great work! The only thing that I miss is IPv6 support. I know that there are some challenges and that?s OK. For me it?s not important whether Gluster servers use IPv4 or IPv6 to speak each other and replicate data. The only thing that I?d like to have is a floating IPv6 for clients when I use Ganesha (just IPv6,

faced this exact same problem a few times on more than one servers and it was 1) dialplan issue which was not hanging up the zap channels correctly 2) using more than 8 spans on a server. Asterisk can't handle more than 96 zap channels on T1s. FXO/FXS combinations can vary the number of spans but if you know what I mean by spans, in production don't use more than 6 spans. On 2010-03-17

The CentOS team is pleased to announce the Planet CentOS website. Are you wondering what is cooking in the kitchens of the CentOS developers? Wait no longer, Planet CentOS aggregates the blogs of CentOS contributors, merging them into one feed. You can read Planet CentOS with your favorite RSS/Atom reader, or through the Planet CentOS website. The Planet CentOS website, with links to RSS and Atom