similar to: GS 101/102 Reboot

I just got an email from SIPphone advising that there have been problems with the above firmware and advising to reload from their server. This does in fact reload 1.0.4.35 into the phone. And now voicemail has gone AWOL again. -- Dave Cotton <dcotton@linuxautrement.com>

As far as I have been able to figure out, dovecot auth always works over a Unix domain socket. I believe it is not currently possible to operate dovecot auth over an Internet domain (TCP) socket. Am I correct? I want to call dovecot's exported authentication from a Java application. Java doesn't natively know how to talk to a Unix domain socket, so there are inconveniences. There

Guys.. I just saw this for the first time... I did some google and wiki without any luck.. what does a red or yellow alarm mean in zaptel? Feb 23 02:54:16 WARNING[16890]: chan_zap.c:5865 handle_init_event: Detected alarm on channel 2: Red Alarm Feb 23 02:54:24 NOTICE[16890]: chan_zap.c:5860 handle_init_event: Alarm cleared on channel 2 This just happened by itself..

I've already sunk several hours into this without any real progress, so I'd really appreciate any help My task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569

Hello, I am trying to change over from bind9 to internal dns. For some reason I keep getting the following error: Error parsing DNS data from '/opt/samba/private/dns/TH01.INET.zone' () I have inspected TH01.INET.ZONE but I could not find an error. I do not understand this error message. I am hoping someone could help me with this issue please. Your help will be very much appreciated.

Hi all, Now i want to check a number (channel) online, offline or unreachable on asterisk but i don`t know to do. Can anyone help me to solve this issue. Thanks and best regard! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110109/c193b48d/attachment.html>

Hi all Do any of you know i can force asterisk to lookup ip addresses for peers and trunks everytime it tries to make a call. One of the peers has a dynamic ip and is using DynDNS to register host. Now i need to reload asterisk everytime i want to call it thanks liaan __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection

Hi, and sorry to bug you all for something so simple: The project in which I am working would like to use CentOS, but it has very strict requirements regarding licensing. The item that I have had trouble with is centos-indexhtml-6-1.el6.centos.src.rpm. Within it, the License is only identified as 'distributable', which is not precise enough. Is there any way of contacting the author of

If someone could point me in the right direction I would much appreciate it. Here is my problem: My directions for my sip phone says to dial an ip address 12*34*65*78#. When I dial that into my phone my asterisk server is only picking up some of the numbers in the above example it would pick up 6578. Then of course not find it and ring busy on the phone. The same is true for dialing a regular

Can anyone tell me what is causing htis by chance? I know it is the application. What's weird is i "fixed' it last time by reformatting the machine to stabilize this one app. Now after a month it crashes again.... Well the steam hlds has started randomaly crashing again. ---------------------------------------------- ---------------------------------------------- CRASH: Tue

Hi, I was wondering, if someone could point me into the right direction, towards building just llc and opt (potentially 4.0 + patches) to distribute them in the interim with compiler? The idea is that we’d like to pin our llvm ir code generator to a specific set of llc and opt tools, with potential patches applied, until we manage to get all those patches upstreamed. I’d like to find answers to

hey * folk, am trying to configure a WellGate 3504A FXS SIP ATA (http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set up two SIP clients in sip.conf as follows: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls [1235] host = dynamic secret = somepass context = default type

Hello everyone, I''ve been working on an internal best practices document that also tackled giving proper terms to things that were at the time uncoined. But looks like we have a bit of a merge problem as other terms have been floating around that seem to conflict with some terms we''ve been using internally at my organization. I''m hoping we can discuss and

Hi, I am trying to forward calls to another * server with IAX Here is What I want to Do 1- Call SERVER1, let say at 51412345678 2- SERVER1 should transfer the call to SERVER2 in a remote location 3- SERVER2 Receive the call and transfer it to the PSTN number. I have one X100P card on each machine. What is happening is that when the remote party picks up the phone, all he can hear is a weird

Has anyone come across any interesting ways to make rails apps easily distributable to run on a local machine. I''d love to be able to "bundle" my rails app into a single file that would contain a simple webserver, database and my rails app which a non-experienced user could then download install on their machine, go to a web browser and run the app locally. I envision it

I appologize if this was already answered somwhere on http://www.voip-info.org/wiki-Asterisk, I'm sure it probably is. And if you wish to just point me to a link that would be appreciated. I am very new to asterisk and unix all around, so these questions may sound rather ignorant. First being, how do I setup asterisk to point to another asterisk server and make all the lines which should

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, 2 weks ago I asked questions about PJSIP and T.38 but got no replies. I upgraded Asterisk to git as of yesterday (309dd2a), and I'm still having the same issues. In the trace below, I'm sending a fax from Hylafax server through iaxmodem on Asterisk-13 (tiare) to a second Asterisk (11.18.0 t0gw) connected to the PSTN via ISDN; the

Mandrake 9.1 Samba 3.0, windows2000 pro sp4. I finally got Samba 3.0 working thanks to G?mes G?za. Now I am having another issue as outlined below: 1)There are 2 users on the Linux pc, markw (regular user) and root. Both have valid samba passwords. 2)I have a share called: [data] path = /test valid users = markw, root read only = No browseable = yes 3)I am able to map the share with the user

Hi, I needed to create config files for downloading to Grandstream devices and made a little script for it. It seems to work fine for the HT486. The script needs a config file (cfg.in) which is in this format: P2 = blah P10 = hrm ...etc... The configfile may contain comments (starting with '#') and empty lines. Mind that the 'gnkey=0b82' shouldn't be in the configfile, as

Not sure what this is. When I call my own ext the call will ring for 10 sec and goto the voicemail. However the phone will keep ringing and I see this on the asterisk CLI Jul 27 11:17:48 WARNING[3563]: chan_sip.c:8666 handle_response: Host '192.168.0.200' does not implement 'PUBLISH' Have no idea what this is talking about 192.168.0.200 is a cisco 7960G