similar to: Emergency calls - forcing through on channel

Hi, This is the first attempt to add a new qemu nvme backend using in-kernel nvme target. Most code are ported from qemu-nvme and also borrow code from Hannes Reinecke's rts-megasas. It's similar as vhost-scsi, but doesn't use virtio. The advantage is guest can run unmodified NVMe driver. So guest can be any OS that has a NVMe driver. The goal is to get as good performance as

Hi, This is the first attempt to add a new qemu nvme backend using in-kernel nvme target. Most code are ported from qemu-nvme and also borrow code from Hannes Reinecke's rts-megasas. It's similar as vhost-scsi, but doesn't use virtio. The advantage is guest can run unmodified NVMe driver. So guest can be any OS that has a NVMe driver. The goal is to get as good performance as

Hi, I have a SPA-3000 and would like to use the 911 recipe from http://www.voip-info.org/wiki-Asterisk+tips+911. So I took the simple recipe and modified it slightly: exten => 911,1,ChanIsAvail(SIP/potsoutbound) exten => 911,2,Dial(SIP/potsoutbound/911) exten => 911,3,Hangup() exten => 911,102,SoftHangup(SIP/potsoutbound) exten => 911,103,Wait(1) exten => 911,104,Goto(1) Now,

I am trying to use SoftHangup in my dialplan, but it's either not working or I'm not using it correctly. when i'm on the console, i see: pbx1*CLI> core show channels Channel Location State Application(Data) SIP/vgw1-000000a2 2156181505 at inbound:1 Up AppDial((Outgoing Line)) SIP/143-0000009f s at macro-SaferSIPDial Up Dial(SIP/99302156181505 at vgw1,, 2 active

anybody recommend a supplier in the UK for a pri/isdn30 line (other than BT) thanx very much __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

Hi, is it possible to hangup a channel from another channel? I want to finish a call from another channel, but if I put exten => h,n,HangUp(channelname) it doesn't hangup... Is that correct? Thanks, _________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Is it possible to softhangup a channel based on priority. I mean I want to put some calls in higher priority lets say 100. If all channels are busy and somebody wants to dial an extension with priority higher than 100. How can softhangup drop a line which has priority less than 100? I will appreciate your valuable help. Thanks Smir

Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in

I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used line. Would the following work for 911 calls? [e911] exten => 911,1,ChanIsAvail(Zap/1) exten => 911,2,Dial(Zap/1/911) exten => 911,3,Hangup() exten => 911,102,ChanIsAvail(Zap/4) exten => 911,103,Dial(Zap/4/911) exten => 911,104,Hangup() exten => 911,203,ChanIsAvail(Zap/5) exten =>

I have put together this call forwarding and dnd config: I'm sure it can be dome with macro's but I couldn't figure that out... anyone care to input. 74 Turns DND on my phone will not ring, drops caller to voicemail... 73 Turns DND off 72+ext forward your extension to another extension and voicemail is left at the forwarded extension. 71 turns off call forwarding. ; dnd Could

Hi, I'm almost ready to migrate Courier to Dovecot 2.2.31 but I have one question about home and mail directory. It's good idea to have same directory form home and mail ? Which scenerio will be better and why ? For example: A) Home: /home/doamin/user1 Mail: /home/doamin/user1 B) Home: /home/doamin/user1/home Mail: /home/doamin/user1 C) Home: /home/doamin/user1/ Mail:

Hi, I've been following the instructions on how to use the LLVM Gold plugin at http://llvm.org/docs/GoldPlugin.html while building an multiple versions of WebKit. The documentation hasn't been updated since 2010 and hasn't really matched my experiences, so I'd like to ask if I'm doing these steps incorrectly. What I'm trying to do is force all compilation steps to

Hello *'s, I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone integrate it with asterisk if anyone what is the scenerio i have scenerio which is quite simple but i am confused about it whether it is possible or not : I integrate it with asterisk for interanet no PSTN at all just only IPphones and analog phones connected on FXS port.Is it's neccassary to cannect with

Hi, I'm wondering if someone can share their thoughts on how to implement a system that periodically checks active channels which have been up for more than X minutes and plays/injects a sound file. The idea is to simply warn users that they've been on the phone for quite a while and maybe they should consider hanging up. If the call stays up for more than Y minutes, it is dropped

Hi there, here is my attempt to initiate a "restart when convenient" from a software SIP phone. exten => 588,1,Answer exten => 588,2,Wait(1) exten => 588,3,Playback(restart-convenient) exten => 588,4,Wait(1) exten => 588,5,Authenticate(00000) exten => 588,6,System(/usr/sbin/asterisk -rx "restart when convenient") exten => 588,7,Hangup The problem: We

Hi all, I am trying to drop a random channel in asterisk 1.6. The following line in extensions.conf works fine for the first channel exten => 911,4,SoftHangup(DAHDI/1-1) But I need to drop random channel for emergency not any specific one. Can someone show correct syntax for this Thanks smir

When I place an outbound call from asterisk 1.6.1.12 to a FXO port on my Cisco 1760V 12.4, the channel changes - seemingly incrementing: e.g., in the first call, below, the channel name is "SIP/vgw1-00000075" -- the second call (on the same FXO port after a soft hangup on the CLI) is "SIP/vgw1-00000077" How can I extract this information in the dialplan so that I can use

Hi guys, I have 1 zap channel in my house shared among couple people. If someone dials 911, I want that zap channel to be disconnected right away to make way for the 911 call. I dug through voip-info.org and didn't find much. Any hints? kel

when i issue asterisk -vvv command i get this error please help regards Barbra [app_softhangup.so] => (Hangs up the requested channel) == Registered application 'SoftHangup' [codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder) == Registered translator 'lpc10tolin' from format 7 to 6, cost 50 == Registered translator 'lintolpc10' from format 6 to 7,

Hello, Most traditional PBX-es have the ability to use external audio source (e.g. radio tuner) for music on hold. This is also useful because you can let your users listen to radio by dialing some extension. I wanted to achieve the same on asterisk, and chan_alsa seemed the logical choice. I installed ALSA drivers, connected the radio to line-in and added the folowing to extensions.conf: exten