similar to: 1.2.12.1 crashing

On Fri, 2006-10-13 at 10:50 -0600, Joseph wrote: On Fri, 2006-10-13 at 07:27 +0200, Remco Barendse wrote: > > On Thu, 12 Oct 2006, Eric "ManxPower" Wieling wrote: > > > > > Matt Florell wrote: > > > > If you downgrade, let us know if it fixes things for you. > > > > > > > > It's strange that there were so many changes in the

We are using SPA-2002s and PAP2Ts to service our VoIP customers. It seems that one of our customers (female) has a voice that is just right that it generates DTMF tones when she talks... I know I've seen this sporatically, but this seems to happen often on her line, and I'm curious if there are any settings on the device to alter this behaviour?

I setup two asterisk servers with identical settings (same extensions, same queues, etc). Each one is connected to the same amount of incoming/outgoing links (1 PRI, 4 BRI, 1 IAX friend, etc, on each box). Most extensions are sip and they register via DNS SRV and other methods so that the two servers are load balanced. Incoming PSTN calls (BRI) reach 50% each server so that's load balanced

Hi Everybody, as far as I see, I installed 1.2.12.1 on a 1.2.0 box. Everything ran OK, but VoiceMail application stated that there were no entries in voicemail.conf, so it didn't work. Installed again 1.2.0 and voil? the VoiceMail app. was working again. I asked to the group, but it seems I'm the only one with this issue! In Italy we say: "Chi lascia la via vecchia per la nuova, sa

Dovecot version 2.1.7 Ubuntu 12.04.3 LTS Kernel 3.2.0-35-generic x86_64 I'm not sure exactly when this started occurring, but sporatically users report issues receiving email, having email saved to "Sent," etc. Looking in dovecot.log, I see the following errors: 2013-10-16 09:53:20 imap-login: Info: Login: user=<user1>, method=PLAIN, rip=127.0.0.1, lip=127.0.0.1, mpid=27434,

So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry

i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be more stable than Fedora core Linux or it makes no significant difference _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ -------------- next part --------------

I have some general questions about marketing. Lot's of technical info but I was wondering how people are getting the business to begin with. I'm from the IT end of things but Telco is quite a bit different. Is cold calling still the way to go or networking? General stuff like that. Are there any resources on the web I can search for? Any suggestions would be appreciated.

Hello, Don't know if this is related but I just got a segmentation fault today while trying to register my new SNOM200 phone: *CLI> *CLI> NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '<sip:mattf2@10.10.10.15>' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from

I just upgraded to Asterisk 1.2.7 from 1.2.5. Page() is behaving differently. I'm getting an error - Incomplete destination '' supplied. -- Executing Page("SIP/2944093-5999", "SIP/3254107&SIP/3254105|") in new stack Apr 13 11:06:11 WARNING[9294]: app_page.c:193 page_exec: Incomplete destination '' supplied. -- Playing 'beep' (language

This is a global message to all to announce our callcenter / dialer / predictive dialer / vicidial program is now open. Codecs: G711, GSM, G729, G723 Protocols: SIP Duration Rate : 30/6 (6/6 with monthly minutes over 100,000) Channels : 100 to start with , more on demand. We are predictive dialer friendly , your account will not be shut off. Contact us to do a test run. Mike

Hi All, Eagerly waiting for v1.4.x to mature a bit before getting serious about it. Is it ready for production yet? If that's too general, where is it in terms of stability compared to where 1.2.x is now. Anyone running it successfully in production environment and if so what sort of config do you have?

I am looking at ways to harden my asterisk install to prevent computer related issues from happening. I am concerned about about disk write cache. That seems to be a major source of hard drive corruption on power failure. Hard Drive corruption is simply unacceptable for the 99.999% uptime requirements of my Asterisk install that needs to be as reliable as a proprietary PBX. Of course I will be

Hi there, I am getting ready to set up a production Asterisk system. It needs to be stable. Upgrading, patching, rebooting, troubleshooting etc. are pretty much NOT an option once this thing is deployed. Like any phone system, it is expected to just work. Having said that, which is the best version and subversion of Asterisk to use? I was leaning towards 1.2 but it appears there are some

Looks like I am going to be doing my first serious commercial install of FreePBX. I DO mean serious. They are willing to put up with a few glitches initially which is why I have decided they will be a good first client. I have turned down several over the past couple years because I just did not feel comfortable with the state of software/hardware. It seems to work much better now. I was

Hello, Now that we've had our new Digium TE406P card in production for 4 days we have discovered audio drop problems that happen randomly across all channels. Here's more about our setup: P4-3.2GHz 2GB ram Slackware Linux 10.1 with custom kernel 2.4.29 Asterisk 1.2beta1 Digium TE406P quad T1 card with the following attached: - 2 x RBS D4/AMI 24 channel T1s - 1 x RBS B8ZS/ESF 24 channel

Hi there, Here is my dilema. I have a new small business customer that wants me to put in a VoIP phone system for them. Based on their requirements, I have determined that it needs to be a "set it and forget it" type of thing like a lot of small business proprietary systems. At the same time they would like to be able to do minor dial plan changes themselves so I have determine

Just wondering what web GUI people like for asterisk. I installed asterisk from source and I was looking at possibly installing web GUI for system management. So far freepbx.org looks promising anybody else have any suggestions. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 roy

Is there a way to change the return path sendmail uses when sending out voicemail to email? Currently the voicemails my asterisk system emails out have a return path of "asterisk at mydomain.com" I would like the return path to be "noreply at mydomain.com" I cannot find any place where I can change that. I tried adding a sendmail alias to send "asterisk" to

So what are peoples thoughts about the new Digium software which appears to combines Asterisk, FreePBX(?), and Linux into one release to eliminate inter dependency issues and emphasizes stability. Seems like a much more professional way to go compared to Trixbox. Anyone using/testing it. I am curious what sort of opinions people have about it. http://www.rpath.org/rbuilder/project/asterisk