similar to: VIP 30 phone

Hello again, I'm not attempting an install of Samba 3.0.4 from source. I want to specifically compile in ads and winbindd support. So, I already compiled and installed OpenLDAP 2.2.13. Now the configure script is hung on krb5 dependancy: checking for krb5.h... no configure: error: Active Directory cannot be supported without krb5.h So I downloaded and attempted compile of krb5 1.3.4, but

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I've got my developers kit from telappliant and got a machine up and running to become the house phone system. Most things are working now, such as incoming calls, call transfer, call parking, voicemail, etc. The one thing I can't do is make my analog phone ring! I can see the call coming in on the asterisk console and can then pick

I have put my phpconfig stuff out into the Digium CVS tree. Project name is phpconfig. see it at http://rads.netcom.utah.edu/phpconfig/phpconfig.php Lemme know if you have any patches or add on's are welcome Dave Packham aka p0lar

Here's a bit of a twist to the common UK Caller ID question... (Which I've got working nicely thanks to some slight changes in Jonathan McHarg's scripts off the asterisk-dev mailing list, and a Pace modem from ebay!) Can a standard BT phone that supports CID (Such as a BT Decor 310) pick up the CID information that asterisk passes out to analog lines or would I have to get an

I'm looking at using a modem to provide caller ID info on my NTL line following the steps in the article posted by Darren Poulson: http://www.22balmoralroad.net/modules.php?name=Sections&op=viewarticle&artid=1 I was wondering if anybody had experience of using any modems (can be pci/isa/external) with an ntl line (in an ex. Cable & Wireless area if it makes a difference) and

Hi, I've got a recent copy of CVS (09/08/04) with the UK CID patch on and working, but I'm having trouble receiving SMS messages. I can send messages without any problems but whenever an incoming message is detected, nothing is left in the /var/spool/asterisk/sms directory. Can anyone help? Thanks, Darren. extensions.conf: [default] exten = s/08005875290,1,SMS(testq,a) exten =

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

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, 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

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

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 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, 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

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'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:.

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 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,

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

Hello all, Forgive me as this is probably a common question. However, I could not find an answer while searching. I have a few Linux servers running Samba 2.2.x that are hosting open shares with guest read/write access. I would like to have these become a member of the active directory and allow pass-thru authentication and authorizations to the shares hosted. I find where I can join the