similar to: Cisco 30 VIP

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

I was able to get Asterisk working with the demo on FreeBSD 5.3 without crashing, but not the music on hold, so I just have that disabled for now, but I'm ready to get some IP hardware working. So I picked up a Cisco 12sp+ IP phone (mistake?) and am having difficulty finding any truly helpful instructions / troubleshooting to get this configured to work with asterisk. If I could just get

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)

Hey all, I have set up my cisco 30vip using chan_skinny because chan_sccp wont register. The problem I am having is, everytime a call is sent to the phone Skinny/200@jason it rings once, then asterisk segfaults. heres the output -- Executing Answer("SIP/4437821638-7588", "") in new stack -- Executing Dial("SIP/4437821638-7588",

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

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

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

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

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 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

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, 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 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 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 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,

Hi list, first of all, I'm new to all this so please excuse me for asking stupid questions :) I've setup an asterisk server. I connected a Cisco sp12+ phone to it, that works. Now I have a windows SIP client, SJphone, and I want to call the cisco phone. In my skinny.conf: [fabje] device=SEP000196c0066c version=F2.02 context=did line => 101 in my sip.conf: [test] type=friend

Lacy's response in the thread 'Why does everyone seem to dislike *now?', has a small bit that caught my eye. Chan_Skinny made a lot of progress between 1.2 and 1.4, and even more in the later 1.4.X releases. I am curious as to which features/functions that chan_skinny might be lacking compared to chan_sccp. We (the community) now have a small, but active, group of volunteers

Hi Steve, I am having this problem in which RxFax is still holding the file after receiving a complete fax. Somehow the zap channel is still active but on the fax client it was sent successfully. If you call the line it is still busy. Changed from phase 3 to 4 >>> MCF: 8c HDLC underflow in state 8 Changed from phase 4 to 3 Slow carrier up <<< DCN: fb DCN with final frame tag

Hi, We try ti hide the caller id at calls trought E1 in EuroISDN (Spain) using restrictcid=yes and doesn?t work. What can I do, thaks Pedro -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: mi?rcoles, 31 de marzo de 2004 12:00 Para: asterisk-users@lists.digium.com