similar to: no delivery from queue on IAX2 extension

I have 4 7960's that I am trying to get working but 2 of them will not update to the SIP image on my tftp server like the first ones did. i keep getting the error on the phone 'Defaulting CM to TFTP server' like it isn't seeing the *.bin on the server. are you supposed to have on of those for each phone? would be like cisco et al to do something like that. TIA Jason Kawakami

so i have been running v1-0 on all of my test boxes for about a month now testing iax/sip/res_xxx. I decided to put it into production so I updated a box that was running 0.9.? that had been working perfectly for months and low and behold the inbound line from telco now intermittantly doesn't clear and none of the other channels can dial out on that line. I have tested the line in this

-----Original Message----- <snip> Is this possible with asterisk? Anyone have a sample dialplan? -other than the problem outlined below I would try something like S,1,wait(20) S,2,voicemail(uwhatever) S,3,hangup That should ignore the call for 20 seconds and then leave a message in the unavailable greeting for 'whatever' then hangup That leaves another problem -

-----Original Message----- My question concern outgoing calls. How can I configure my extensions.conf to get a PSTN line on my TDM04B card in the following order : first trying on the channel 4 then if 4 is busy then switch to 3 if 3 is busy then switch to 2 and if 2 is busy then say there's no more line available. I don't want to dial on the first channel as it's my main number

Hello all- So I have * up and running and connected to a legacy system via em_w lines and have no trouble dialing from * through the tie line but from the PBX across the tie line I am having intermittant receipt of the DTMF. T-Berd testing is showing that the digits are coming across but * is either missing the first digit consistantly. This seems to me to have something to do with start timers

So I have a route with [1-9][0-9][0-9][1-9][0-9]* as a base route that should match NXXNX. Right? I built another route 01144[0-9]* that I thought would match 01144X. and send the call to the UK but the script is matching 01144207108???? With the first route. Can someone smarter than me help with some samples? Please? If I can get one for 1NXXN. and 01144. I should be able to figure the rest

----- Original Message ----- > Subject: Re: [Asterisk-Users] Analog Phones with Status Light Indicators > From: Adam Goryachev <mailinglists@websitemanagers.com.au> > To: asterisk-users@lists.digium.com > Organization: Website Managers > Date: Thu, 12 Aug 2004 14:53:02 +1000 > Reply-To: asterisk-users@lists.digium.com > > On Wed, 2004-08-11 at 20:42, Steven

hello all- has anyone had any problems with the handsets on BT100's. Just picked one up for my lab and the speakerphone works great but I am only getting one way audio (incoming) from the handset. Since the speakerphone works fine, I can't think of any config. reasons why the handset wouldn't other than a faulty handset. Any thoughts or experiences? Jason Kawakami Technical

is there something that needs to be set up to make the 'called' and 'callers' buttons work on this phone? all i get is the backlight to switch on and off. Jason Kawakami

I'm thinking about using asterisk as an IVR system with an existing avaya index system. I've got 2x PRI 30 lines coming in to the Index, and I have 4 spare PRI cards in the Index. I was thinking about using a QUAD PRI card from Digium and having the calls come into the Index then transfer to Asterisk for IVR then back to the Index. That way if we get 60 inbound calls we'd in

----- Original Message ----- <snip> > That's what I was afraid of - we have more extensions than channels ... the number of extensions isn't the issue. it is the signal the meridian broadcasts when your digital telephone on your desk is on/off hook, or ringing or whatever. if you could tap into that information and monitor it with something similar to *'s manager interface

What experience can be shared about installing and running the OrderlyQ application? I have a bunch of queues set up and want to start adding some additional apps and this one looked promising but I have very little java experience and it doesn't seem to be running properly. Jason Kawakami -------------- next part -------------- An HTML attachment was scrubbed... URL:

After our department went to using *, I've had several inquiries about doing VoIP for my entire organization (Small county). We have ~10 locations with various links in between (Mostly p2p T1s, some Frame (1.544mbps commit), some ISDN, some VPN over 768kbit internet) Right now we're using several NEC Electra Elite systems, and 2 Nortel Meridian systems. In one of the main locations we have

Signed-off-by: George Dunlap <george.dunlap@eu.citrix.com> diff --git a/xen/arch/x86/mm/paging.c b/xen/arch/x86/mm/paging.c --- a/xen/arch/x86/mm/paging.c +++ b/xen/arch/x86/mm/paging.c @@ -722,6 +722,10 @@ int paging_domctl(struct domain *d, xen_ /* Call when destroying a domain */ void paging_teardown(struct domain *d) { + /* Make sure log-dirty is turned off before trying to

Dear Friends Greetings, i am CentOS User for some years now, have installed and configured squirrelmail number of times without issues. but this time it is on CentOS 6.2 x64 - i see very ugly login interface. of squirrelmail, i wish to mention that the package was installed from epelrepo becuse it is not available on centos or rpmforge repo either. i can login also, after login this is how i

This is very off topic but, I have no idea where else to ask. We obtained some used Leveno CTO7483 desktop units for experimentation. I had intended to install CentOS-7 on one of them. As they arrived with a vendor upgraded Windows 7ProSP1 install without media I decided to pull the disk drive and install onto a spare drive that I installed. Those of you with any experience with this model

Hello, I've been testing llvm-exegesis on X86. Latency and inverse_throughput modes work fine but when I run uops I get an error: event not found - cannot create event uops_dispatched_port:port_0 LLVM ERROR: invalid perf event 'uops_dispatched_port:port_0' I'm running this on a i7-4790K. Am I missing something on my computer or is this not yet fully implemented? This also

I'm trying to clean up some notices/warnings that are repeatedly logged in *.Any Help would be appreciated as I'm not sure of the cause /solution. Here are the errors: Nov 17 15:53:38 WARNING[1217602880]: File chan_zap.c, Line 1321 (zt_call): cidspill already exists?? +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ /* Don't send audio while on hook, until the call

Jean-Marc, What exactly does the speex code do to calculate the granulepos? With vorbis it's the 'count of decodable samples including this packet'. So if you had a packet that allowed you to decode 1024 samples since the beginning of playback, the granulepos is 1024, not for example 1023 which would be the *index* of the last sample assuming C-style array indicies starting at

I need to be able to dial a long distance provider that uses an access number and an auth code. I would like to be able to program this so that the user can dial 8 and then the long distance number, asterisk will hopefully do everything in the middle. The sequence to accessing the provider is on my traditional phone speed dial as: * Dial local access number * Wait 5 seconds * Dial the auth