search for: recenly

hi friends, maybe you could help me with following problem: We have one SuperMicro dual xenon machine running Centos 4.3 ia32 version and we have problem with reboting this machine remotely (command reboot from ssh connection). After upgrade to kernel versions 2.6.9-42.0.2.ELsmp and 2.6.9-34.0.2.ELsmp the machine hangs right after the: md: md0 switched to read-only mode. message and does not

Hi, We are using VOIP-SIP gateway to route outbound PSTN calls. Recently, I am getting == No one is available to answer at this time message, after making 5 SIP attempts (Retransmitting #5 (no NAT):), and the calls are going out through alternate Zap-trunk. I do not see any hit (sip-debug traffic) on the voip-gateway for the failed calls. Strange thing is that this is happening randomly,

Hello All, I've recenly installed RedHat Linux 9 in my Celvin EasyPC from Fujitsu Siemens. I've installed the minimal instalation. Video card, mouse, keyboard and everithing was detected and configured properly (I could install the packages in graphical mode). The problem came when I tried to ping another machine in...

Hello! I wanted to make sure that, in addition to my complaints, I make it very clear: Digium's support is excellent. The jury is still out on the usefulness of the TDM products. However, Digium has worked very hard to make sure that this issue is resolved. I actually got an e-mail from someone at Digium actually asking what they could do to make me happy! She even gave me

Here is the output. These are Panasonic KX-TG2564's. Does something need to be set for the phones? I can call out fine, but all of the extensions seem to be busy. Starting simple switch on 'Zap/5-1' -- Executing Macro("Zap/5-1", "exten-vm|200@default|200") in new stack -- Executing SetVar("Zap/5-1", "FROMCONTEXT=exten-vm") in new stack

Hi! I have configured a SIP trunk with a dialing rule. The trunk behaves normally for incoming calls but when in used for outgoing call a strange thing happens. When I place a call I cannot hear the tone confirming that the remote phone is ringing. I simply hear the voice as soon as the party picks up. When the remote phone start ringing Asterisk receives a SIP packet stating that the call is

...9;' The problem seems to be occuring in lib/mocha/test_case_adapter.rb which was updated by revision 68. Wondering if anyone else is seeing this problem and if not where I should start to look in my code for the incompatibility. I''ve looked and looked to see if something I changed recenly could possibly have caused this problem (even canceled posting to this list once because I thought I''d found it), but I can''t see what I could be doing wrong. Thanks for the help, Greg http://www.urbanhonking.com/ideasfordozens http://www.virtuous.com

OK, i've got asterisk @ home 1.3 up and running with Broadvoice. BUT I have one nagging problem to sort out. When you call my BV # the calling party gets no ring indication, just silence until either I answer the phone, or the call bounces over to voicemail. below is the console output when a call is recieved. what am i missing here? thanks Bernie -- Executing

I don't succed in getting callerId on incoming calls on a zap trunk. I am using a zaphfc card When a call is received, one line in asterisk pbx says -- Executing SetCIDName("Zap/32-1", "") in new stack second parameters should be the caller ID, but it is not set The callerid is not hidden at source, so I think that is some kind of setting in zapata.conf I am using

Hello all, I have been posting some questions about this problems that I cannot yet solve, but I think I have a better diagostic, so maybe someone can give me a clue why it is happenning. I have Asterisk + AMP configured as a PBX with a Customer Center Queue with 4 agents that login/logout dinamically. If there are no agents, queue timesout and gets derived to another queue that somebody

Hi, I have some problem to get this setup working. I have a CAC Channel Banl I, with FXO and an Asterisk box ( I am using Asterisk@Home 1.2) and I have a TE110p installed in this box. What I want to do is, just to be able to dial one of those lines that already are connected to the channel bank, and transfer that call through TE110p and Asterisk to a user agent somewhere through Internet.

Hi All, I am a new bee to *. I just installed Asterisk@home on FC3. I hv a FXO card. I hv configured two extensions one x-lite and other iaxComm. I configured * using AMP. The following setup works - x-lite (x 200) to iaxComm (x 201) - PSTN to x-lite - PSTN to iaxComm Voice mail, weather etc work fine. When i try to make an external call i am getting message "All routes are busy". In

Hi, We recenly upgraded to verion 1.9.18p4 on Digital Unix 4.0b (I believe our previous smb version was 1.9.16). Occasionally we have reports of one of our users having their w95 logon script freeze at the point where it tries to mount their samba drives. I have found I can recreate this problem by: - log...

Hello, reading the wiki I found http://wiki.dovecot.org/PasswordDatabase/ExtraFields/Proxy?highlight=(no DNS resolving) I have a setup where the destination of a proxyconnection is an DNS name. Is there a technical reason which currently require host= to be an IP address ? -- Andreas Schulze Internetdienste | P532 DATEV eG 90329 N?rnberg | Telefon +49 911 319-0 | Telefax +49 911 319-3196

The 2 errors: pam_setcred: error Permission denied Cannot delete credentials[7]: Permission denied Looks to be a major bug in the PAM module for Solaris-2.8/2.7/2.6. Has anyone from the list (developers of OpenSSH, endusers, hackers, etc.) came up w/ a solution? Even a temporary one? When authenticating yourself on the same system that worked, but when authenticating to another system failed. I

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

Hi - I've recently bought a mediatrix 1204 and have had a complete nightmare getting it up and running with an asterisk@home setup. I know this isn't a mediatrix list but I'm at my wits end and the support with this product is atrocious. (mine was even shipped with firmware that was incompatible with the win32 software it came with so I wasted a day trying to work out why the SNMP

I currently have: Telco-PRI ---- Panasonic DBS576 PBX ---- E&M wink T1 ---- Asterisk. I have configured the Panasonic to forward my Asterisk DIDs to the Asterisk extensions over the T1. I do not get DID nor CID on the Asterisk, so I want to use PRI between the PBXs. I do not want to pay for another PRI card for the Panasonic. (T1 and PRI are different cards) I see this as my least

Hi Is there a phone comparison matrix I could consult I have a series of features that I would like to evaluate on the most common phones on the market example: dual-ethernet POE / direct power / both number of lines speed dials programmable buttons BLF LEDS Headset plug conference call built in hands free operation display size codecs communication protocol (SIP, h.323) price availability

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm