similar to: calls get stuck in asterisk

Hello Is het possible to assign blf to a misdn channel? I want to watch the status of my external misdn channels on a linksys 962, e.g. green = available , red = in use and as an extra I want when I press the blf use the external line or when busy i want to barge in that call. Did somebody do this before?

Hello all, I am using asterisk 1.4.17 together with misdn, once in a while: -when a call was put on hold -the operator tries to call a internal party for transfering the call -the internal party doesn't answer the phone -the operator wants to get the external line backup again by putting the call "off hold" And then the external line is disconnected. an exact log of events is

*Postdoc Position Computational Systems Biology 'Systems Bioinformatics: Computational Modelling Methods' f/m * * VU University Amsterdam, the Netherlands * *Research project * The goal of the project is to develop and implement computer methods for computational modelling of biological systems. This encompasses methods for computational model assembly, execution, analysis and

Hello All, can anybody tell me how can i integrate asterisk and skype users so that skype users can dial my asterisk number or dial internal dialplan form skype regars Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090708/cccd4587/attachment.htm

Hi there. My problem, I can't figure out how to ask this question. So, hopefully someone out here can point me to the FM on this. I would like to have either a web page or an application that I can view that whenever a call arrives on the Asterisk server the application will display the callerid information. I've found quite a few examples of the reverse of this. To where a script is

Hi, Here's my smb.conf file: [global] workgroup = MYGROUP server string = Samba Server hosts allow = 192.168.0. 127. load printers = yes log file = /var/log/log.%m max log size = 50 security = user socket options = TCP_NODELAY dns proxy = no [homes] comment = Home Directories browseable = no writeable = yes [printers] comment = All Printers path =

I have setup a few Asterisk systems for customers using Digium TDM400 cards and Aastra phones. No problems with sound quality at all except at this one site. Every time I try their system I don't hear any problems but they tell me that it is really bad. They describe it a a loud scratching sound. Are there any tests that can be done to pinpoint the problem? Has anyone seen this before? Are

I'm running AsteriskNow 1.7.1 with a OpenVox 2/FXO/2FXS card, a Linksys SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the call and when Asterisk connects the call at my end. There's enough delay time that I hear an additional ring after the PSTN

Hi, I still see problems when std::map's are used. Is this expected or are map's still not supported? Cheers, Fons. -- Org: CERN, European Laboratory for Particle Physics. Mail: 1211 Geneve 23, Switzerland E-Mail: Fons.Rademakers at cern.ch Phone: +41 22 7679248 WWW: http://fons.rademakers.org Fax: +41 22 7669640

Hi, what would be needed to make a C/C++ interpreter using the LLVM libraries. We have in our project (http://root.cern.ch) a C/C++ interpreter (http://root.cern.ch/twiki/bin/view/ROOT/CINT), but it has some limitations (the biggest being maintenance). I see there is a libLLVMInterpreter that can interpret the LLVM IR. Could this be used to interpret, or a starting point, to an real

Hi, I am playing around with PCI passthrough and came across some posts that said you could not do PCI passthrough unless you had IOMMU hardware. which it would appear I don't Is this the case? Am I flogging a dead horse? Thanks, Marco -- Marco van Beek ========================================== Supporting Role Ltd. Grove Park Studios, 188-192 Sutton Court Rd, London, W4 3HR

Hi Chris, it would be good if "the powers that be" could be made to understand that if only Apple employees are not allowed to put up their slides while Apple is employing the core of the LLVM foundation, that this sends the absolute wrong message on openness of the project. It is also totally incompatible with the fantastic work you guys are doing and your tremendous presence

Hello again, Because the LLVM in HPC workshop will be in Austin on Nov. 15th (http://llvm-hpc2-workshop.github.io/), we'll have an anomalously-high density of LLVM developers in Austin that day. I think it would be a great evening to have an LLVM social! I'm not familair with the Austin area, but I've cc'd some folks who are (or at least were) in Austin, so hopefully we can get

Dear list, I need to buy a phone which could monitor the state of the maximun number of sip extensions about 200. It is for an attendant. I just saw Snom 370 with keypad and Linksys 962 but they do not let me to monitor 200 extensions states adding keypads. Do you know any kind of phone that let me do that? Which is the maximun number of extensions your phones can monitor and which models phones

Hi, I've noticed that if I have a multi-line linksys (942 or 962) phone with the same sip registration mapped to each line key, that if all the lines are full the phone will accept another call. I would expect the phone to respond with "busy" so the call would to directly to voicemail. Has anyone else experienced this and know of a workaround? I know it seems like an

Let me add another variable into the mix...what about the Linksys SPA-962? Good, bad or otherwise? The 32 button sidecar seems like a deal at $80 street price. Michael On Fri, 22 Feb 2008 08:28:37 -0500, Matthew Brothers wrote: > >Michael Graves wrote: >> I need to add a few phones to an existing installation. They have a >> dozen IP430 at the moment. Does anyone feel that

On Sat, Jun 18, 2011 at 6:40 AM, Larry Moore <lmoore at starwon.com.au> wrote: > On 18/06/2011 5:36 AM, Matteo Campana wrote: > >> >> Inviato da iPhone >> >> Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling<EWieling at nyigc.com> >> ha scritto: >> >> We experience the same thing. The solution we use is to not change >>>

Hello All! I have a problem with my calls, that drops after 20 - 30 seconds. I got a piece of PAP2-NA log and Asterisk log and there's an error 603 - call declived, as showed. Thanks for any help. McCoy *********** PAP2-NA LOG *********** Feb 9 09:00:56 192.168.4.205 Feb 9 09:01:11 192.168.4.205 [0:5060]->192.168.3.14:5060 Feb 9 09:01:11 192.168.4.205 [0:5060]->192.168.3.14:5060

Hi, I am trying to get a USB device working on a Windows 7 virtual machine, and it needs USB2 to work, so the USB passthrough does not work. I have searched for an answer and either the answer is so obvious I am being thick, or too complicated for anyone to address it. Can I simply do a PCI passthrough of a USB host controller? Would that then give me a whole load of USB2 ports I could

Hello, Yesterday night we have upgraded our Asterisk from 1.2.32 to 1.4.24.1 with lipbri 1.4.10, dahdi-linux-2.2.0-rc4 and dahdi-tools-2.2.0-rc2. Libpri and dahdi is only for dahdi dummy cause of the meetme function. After the upgrade we had the problem that some Linksys spa941 phone at one location could not dial out. incoming calls to the phones works without any problem, but outbound the