similar to: Message waiting light on polycom 301 using asterisk 1.4.14

I am having a issues with adding a analog card to my dell 2800. I already have a t1 card installed and running fine but when I install the analog card asterisk will not start (ztcfg fails). I have determined it is because of a IRQ problem and have decided to get a new server. Can anyone suggest a server grade setup that supports this? I would rather not buy a machine that will be unstable. I

I am trying to tweak my dial plan and I am running into a problem. Sometimes my VoIP out bound calls do not complete on overseas calls(busy or just a hang-up). Is there a way in the dial plan to automatically dial out of my PRI when something like this happens. Either by time limit by a failure event? Any point in the right direction would be great Thanks, CLI output (cleansed to protect the

Hi List!! Thanks for the colaboration, especially to Richard Cavanna who gave me the necessary support. I followed your indications and the comunication was better for the test users. The warning indication is no jumping anymore and the voice is not delayed. This is my sip.conf: [general] context=default ;allowguest=no ;realm=mydomain.tld bindport=5060 bindaddr=0.0.0.0 srvlookup=yes

I think your phone may be trying to register with the username '1234', while your sip configuration is expecting 'luca'. Can you try changing your phone registration credentials to use 'luca'? Can you give us a sip transcript when you try to place a call from it? On 15-05-28 05:09 PM, Luca Bertoncello wrote: > Darryl Moore <darryl at moores.ca> schrieb: >

Hi list! Finally I got my wife's phone working in my Asterisk. Unfortunately I have some problems, too... Current situation: - AsteriskNOW with 4 Accounts (00493511111111, 00493512222222, 00493513333333, 5678). This is "for test" and it will be replaced by "the real world", when I got my Asterisk to work... - A second Asterisk (Ubuntu-PBX) on another VM, logging in

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

Hi, when I'm trying to call the following extension exten => 6002,1,Verbose(1|Extension 6002) exten => 6002,n,Dial(Agent/6002) exten => 6002,n,Hangup() the call is terminated and I get the following warning from asterisk: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Agent' (cause 17 - User busy) When calling the agent with Dial(SIP/6002) no problem

Hi all, I'm confused, in digium website, it says: TDM2400P: It supports a combination of up to 6 FXS and/or FXO modules for a total of 24 lines. 6 plus 6 is 12, how come it's 24? if I have 24 PSTN lines, i'll be needing 24 FXOs. Pls. elaborate. thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am trying to configure MeetMe so that external callers can enter the conference rooms after an IVR menu. I have created Conf rooms for all internal Ext's with a prefix of 8. When I call into the system from my vonage trunck the IVR picks up but will not let me dial a conf room. It tells me it is a invalid extension. Can anyone help with a sample conf on this? Thanks, RC

Hi again, list! I know, I'm really annoying the list... :) Well, maybe I got my Asterisk at home ("wrt" on the previous E-Mails) to accept my mobile phone from Internet. It was a problem with the network and the firewall. Now I can log my mobile phone in my Asterisk in and the phone is REACHABLE. Wow! Got it! If I call a phone at home using my cellphone it works and the

[sorry for my maybe newbie Qs and using this group for my busybox issue but though having visited some forums about this I'm getting nuts] I unslugged a Linksys NSLU2 device with V2.3R63-uNSLUng-6.8-beta as I plan to use it as a DC replacement for my W2k server at home. But I already got stuck before this. Following the unslinging I installed the Samba3 IPKGs for the NSLU guided by the hints

Xapian 1.4.14 can now be downloaded from: https://xapian.org/download This release is mainly composed of optimisations, bug fixes, and test harness improvements. The wiki has the usual summary of the most notable changes: https://trac.xapian.org/wiki/ReleaseOverview/1.4.14 A big thanks to Austin Clements, Ilari Nieminen, Oliver Runge, and Dagobert Michelsen for helping to make this release a

The Asterisk Development Team has released Asterisk version 1.4.14. This is a regular maintenance release that contains numerous bug fixes across the entire code base. A ChangeLog that lists all changes that were made is available with the release. http://svn.digium.com/view/asterisk/tags/1.4.14/README?view=markup The release is available on downloads.digium.com. It is also available as a patch

The Asterisk Development Team has released Asterisk version 1.4.14. This is a regular maintenance release that contains numerous bug fixes across the entire code base. A ChangeLog that lists all changes that were made is available with the release. http://svn.digium.com/view/asterisk/tags/1.4.14/README?view=markup The release is available on downloads.digium.com. It is also available as a patch

Hi! The Netfilter project proudly presents: iptables 1.4.14 This release several bugfixes and support for the new cttimeout infrastructure. This allows you to attach specific timeout policies to flow via iptables CT target. The following example shows the usage of this new infrastructure in a couple of steps: 1) Create a timeout policy with name `custom-tcp-policy1': nfct

I have problems with 1.4.14, it crash every few minutes. The same configuration and machine in Asterisk 1.4.6 it doesn?t happend. Is there anybody with similiar problems? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071129/e1447f73/attachment.htm

I have a Polycom Soundpoint IP 500 that I have been using with Asterisk for a few weeks. It has been working OK, no major problems other than a freeze up every now and then, until today. The power apparently went out last night and for some reason the phone appears to be working but I keep getting the following errors repeating over and over in my Asterisk log file (IP's X'ed out): Aug

We recently deployed 10 SNOMs as part of a PBX hosted solution. We have one phone setup as the receptionist phone, using hints to show busy office lines. This all works as expected. This is a new installation, and people are just starting to setup their phones. For those of you not familiar with SNOM phones, there is a row of keys on the right side of the phone which SNOM calls function keys. In

Hi All, I have an small issue, which is not creating any problem on working syatem but not sure about the problem that is why eager to know about it. I had installed Asterisk realtime with Asterisk 1.4.41. Every thing is working good but getting warning at Asterisk CLI. [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:17:36] WARNING[17811]:

I have just finished setting up a new asterisk system which is basically the same as our first system. We are using 7960 phones and I used the phone config files the first installation with appropriate changes. The problem is that on the new system I get no message lights, I can't figure this out. One thing I do notice is that when I monitor the sip debug on the second system the sip