similar to: VIP-050

Hi Last friday I have upgraded to Asterisk 1.2.7.1 (bristuff-0.3.0-PRE-1p.tar.gz). Since that I have a problem with my Asterisk box. I am receiving these messages: May 24 09:30:11 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 2). May 24 09:30:12 asterisk kernel: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 2).

Hi, I just want to ask if anyone has some experience with Alarmreceiver application in * 1.2? Is this application reliable (according to wiki it isn't)? I managed to communicate Asterisk (Alarmreceiver) with a burglar alarm, but it behaves very strange. Sometimes alarmreceiver is able to get some events but sometimes not. Maybe there are some other non commercial applications which work under

Hi, Does anyone have some experience with Nokia 32 Terminal (it is an analog GSM Gateway)? After a configuration I can make only incoming calls, I'm not able to do any outgoing. Nokia signalize an error (4 short tones), when I try to phone someone. I tried postpaid simcards as well as prepaid simcards with the same result. Does anyone try to connect this gateway to Asterisk PBX if so what

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

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

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)

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

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

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

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, 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'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,

Is H323 built into the current CVS? If so, could someone give me an idea of a simple config? thanks, darran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030429/fffec1d1/attachment.htm

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