similar to: Gateway VIP450FO and VIP 400FO

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 all, I am having problem in using SIP gateway in a very high volume environment (4+ concurrent dial-in) and phone calls in every seconds. There are totally 12 phone lines. Currently I am using 6x VIP-450 for 12 FXS and 12 FXO. However, in this high volume environment, problems encountered: 1) After the gateway running for about 30 minutes, the VIP-450 becomes very unstable. When I try to

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

Hi, I have a new phone in our IP phone network: Planet VIP-101T. When calling from that Planet phone to anybody, everthing is fine. But when calling from anybody to that Planet phone, I get a mashine gun noise and the following msg in asterisk log: NOTICE[262161]: File channel.c, Line 1091 (ast_read): Dropping incompatible voice frame on H323:0 of format SLINR since our native format has

I have some Grandstream BT101 SIP phones. Work great (so far). I have some "Planet VIP-101T" H323 phones... how do I make them look/feel/act like a SIP phone ???? I can dial to them from both Trunk + SIP's (ie - I've added 'oh323' libraries) What config do I add so that if I dial the * IP - they then at least act as an extension? Ideally I'd like to just pick up

-----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)

I have on Chan_h323 with G729 and X100P trying to connect to a Planet VOIP400 gateway box(http://www.planet.com.tw) I uncommented g729 in the Makefile and I'm setting g729 in h323.conf I'm receving in my side: 1:20.906 H225 Caller:810f070 h323ep.cxx(1537) H323 Clearing connection ip$localhost/4112 reason=EndedByRemoteUser and the other side(Planet) says: 15- RADH 2

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 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 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

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

A few months ago I discovered *. Recently have taken more interest in it, since I have to install phones into my wife's small office. Hope that my questions weren't discussed recently and are not to dumb. The current requirements for * consist of a single incoming ISDN BRI line (currently have 5 MSN allocated), 3 internal extensions and a single analog fax. 1. I plan to buy some el

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>'

I'm attempting to do a first-time Asterisk install at home, firstly for use by my self and my family, and secondly as a learning experience. I've got a new house, and the previous owners removed all but one (1) phone jack. So I figured I might as well build a PBX. Functional goals include station-to-station calling, rudimentary auto attendant/voice mail, and perhaps tieing into the

Hi all, I've a Planet VIP-150T VoIP Hardphone connected to Asterisk. When I'm in a call and i press Hold button, the other party starts listening Music on Hold but then when i press the button again to get the call back it doesn't work! I've checked asterisk CLI: -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 --

Hi pros, Newbie to asterisk, need some help. My existing senerio is we have 6 analog quintums and 1 digital H323, and our gatekeeper is gnugk openh323 located in US. Our business is Call Center and our method of dial is using prefix and gateway IP provided my Carrier. I also brought two AudioCodes MP108 8 FXS gateways, as our gateway runs on h323 my friend suggested to go for Asterisk. If

Hi, I don't know quite how to ask this question, because my knowledge is so limited at this time. I have an h323 phone that I am trying to use to do VOIP to phones on the PSTN. I want to sign up for a service and not have it go out my POTS line. I do have a Quicknet Line jack in my RH 9 box and it is fully confiugred. I have downloaded the latest drive from openh323.org and installed it

Hi, I am familiar with asterisk, though never actually tinkered with one myself ... so i don't know the full extent of its capabilities. I am facing a request to bridge a sip network and an h323 network. I would like to operate the sip with ser as the proxy and some gatekeeper on the h323 side (not required though). Actually, i have a few more points that may make it simpler - i do not need