similar to: target number is busy after some calls

Hello, I have a question about "directmedia" or "canreinvite", I have experience that whatever I set directmedia=yes or no. After I run sip show settings. all settings looks the same. My question is how I could make sure from "sip show settings" that my "directmedia" configuration is applied. Thanks

Hi all, Anyone ever used GoIP GSM SIP Gateways ? If yes, what was your experience with those ? I'm looking at this: http://www.ebay.com/itm/HOT-GSM-VOIP-GoIP-Gateway-SIP-Trunk-to-Asterisk-iP-PBX-/280736774012?pt=US_VoIP_Business_Phones_IP_PBX&hash=item415d37377c If anyone has any (good) experience with another brand, I'll take the names and models. Thanks

Hi, I write some php code in AMI to working with asterisk command. I don't know exactly what is the different between AMI and AGI and witch one is better for my planning. Im planning to call party users that their number is is my panel on web. We have some operator and they can call party users via client softphone by clicking on their number, so they have to limited to call just listed

Hi All; How to let the virualbox (ubuntu OS) to be able to see the digium card? Because when I install elastix or asterisk with dahdi, it is not able to see the digium card if the installation though the virualbox .. What is the solution? Regards Bilal

I am using asterisk 1.4.21 with outgoing call files. I am call a line that is busy as you can see below. How can my AGI ask what the status of the last call was so I can tell if there was NO ANSWER or it was BUSY? Thanks, Jerry -- Attempting call on SIP/401 for smvoice_callprogress at smvoice-dialout:1 (Retry 1) -- Got SIP response 486 "Busy" back from 192.168.1.161

Hello; I am looking for SIM adaptor to be connected with Asterisk to be able to send and receive calls from the mobile operator and if possible the same adapter to be used for SMS "sending and receiving". But what if anyone called this SIM card that is connected to this adapter and no one relied his call, how this miss call can reach for the use at the asterisk PBX? Regards Bilal

Hi this message give me when I calling a number than actually not busy: "Dial failed due to trunk reporting BUSY - giving up" max channel is unlimited and sometimes it dial number ok but most of the time it gives me this error. Please inform me how can solve this problem. thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am really a beginner of PRI ISDN board, I am wondering if there is a "quick start" chan_dahdi.conf configuration I could use. I tried to install two "FreePBX" boxes follow the instructions from "http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html" connected them between PRIs, It worked. And now if I refer the FreePBX "chan_dahdi.conf" it looks

UPS: PowerWalker VFI 3000RT LCD CentOS 6.5, nut installed from RPM package: nut-2.6.5-2.el6.x86_64 nut-client-2.6.5-2.el6.x86_64 Driver: blazer_usb Problem description: USB device "loops" when starting blazer_usb. USB device permissions are ok. Receives some information. Error messages: "failed to claim USB device: could not claim interface 0: Device or resource busy"

Hi I just bought some digium analog cards and I would like to build an IVR system for my customers. However I am googling and googling , I didn't find any blog and instruction for beginners like me. So I come here for help. Any tips or blogs will help. Regards, Hua Jie -------------- next part -------------- An HTML attachment was scrubbed... URL:

https://bugs.freedesktop.org/show_bug.cgi?id=60704 Priority: medium Bug ID: 60704 Assignee: nouveau at lists.freedesktop.org Summary: [nouveau, git regression] - I2C PWM fan control broken on nv50 adt7475 on kernels 3.3.x+ QA Contact: xorg-team at lists.x.org Severity: major Classification: Unclassified

> > I have written them by hand in the .s file in the same way of the IR > > reference (http://llvm.org/docs/LangRef.html#named-metadata) : > > > > define i32 @function(i32 %argInt, metadata !3) nounwind { > > This seems wrong. "metadata" is a type (like i32), and the exclamation > mark is only used to refer to metadata nodes, not to declare functions

Hi all! I can't use the TSE licence server in Windows 2008 server. This Server is member of my Samba Domain. My TSE licence server is actived and my licences added, but when i want configure the TSE service and launch the Licence diagnostic the diagnostic failed. I think my problem is due to my Windows Server is not an Active Directory controller. What are the solutions : quit the

That's metadata for arguments to calling a function, you tried to attach metadata to the arguments of a declaration of a function. On 11 February 2013 22:58, Niko Zarzani <koni10 at hotmail.it> wrote: > > > I have written them by hand in the .s file in the same way of the IR > > > reference (http://llvm.org/docs/LangRef.html#named-metadata) : > > > >

Dear all, i have a problem with a system running - Ubuntu 10.04 ( all updates done ) - ii asterisk 1:1.8.5.0-1digium1~lucid Open Source Private Branch Exchange (PBX) - ii asterisk-dahdi 1:1.8.5.0-1digium1~lucid DAHDI devices support for the Asterisk PBX I also use freepbx 2.9 for the configuration. Hardware is a Dell R410 and a Digium

Jonas, At this point, the DFA packetizer models a simple VLIW architecture and does not accommodate multiple stages. That's the reason for the behavior you're seeing. -Anshu --- Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation *From:*llvmdev-bounces at cs.uiuc.edu [mailto:llvmdev-bounces at cs.uiuc.edu] *On Behalf Of *Jonas

I just signed up for Voicepulse with a DID. I can register with Voicepulse and dialout just fine. Only problem is that when I dial my DID from my POTS line I just get a fast busy and nothing in the console. Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040104/d5bd5bd3/attachment.htm

Hi, I've built an Asterisk HA cluster by means of heartbeat and drbd. The following folders are stored on shared storage and referred to by means of symbolic links: /etc/asterisk /var/lib/asterisk /usr/lib/asterisk /var/spool/asterisk /var/log/asterisk I was under the impression that phone registrations were stored in /var/lib/asterisk/astdb and as such preserved when failing over. But

Hi all I would like to install CentOS on this: http://www.lannerinc.com/x86_Network_Appliances/FW-7520 - No VGA/DVI - Only a "console" port (the old 9600 baudrate one) - Boot on USB actived by default Would you know a tutorial helping on installing CentOS on this? I guess I have to - download the minimal iso, - 'dd'-copy it to a flash USB drive - setup some files to use the

Hello There, I would like to play a busy tone (ie BEEP BEEP BEEP) when the underline carrier sends back 486 Busy Here. Looking at Dial parameters ( http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial), it mentioned something about the r parameter as not being very professional or something like that... Then there was: U(x): Executes, via gosub, routine x on the called channel. This is similar