similar to: Question about "directmedia" or "canreinvite" in sip.conf

Hey all For some reason the mailing list is sending all messages from the sending party. This makes it less than ideal when responding; as selecting reply goes to the person and not the list. Can we have it set back to the old way please? Thanks Andrew for pointing this out to me. Bryant -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 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

I've got several SIP clients with dynamic IP addresses Asterisk has one public and one private IP address SIP clients might connect to Asterisk from either the internet or the private network (192.168.1.255) - they're portable By default, directmedia/canreinvite is enabled and Asterisk sets up direct media connections between clients. In this case clients on the internet can make calls

Hi, when using some non dynamic host eg. host=192.168.111.153 in sip.conf asterisk is not considering specific peer options eg. directmedia=off, transport=tcp if I set host=dynamic and register the sip phone it works as expected. Is this a bug or feature - I wanna disable the usage of directmedia for some peers with fixed ip but wanna allow it in general. Same with transport=tcp. [97]

Hi, all, Is the Xen currently compatible with the InfiniBand? I found some information about the Smart I/O module, but it was posted in 2006. Is the module still maintained? Or, are there any up-to-date alternatives for that? Many thanks, Chiu _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

Hello, I have an Asterisk 1.8.11 installation. When I built up this Asterisk, I didn't install DAHDI channel, if I issue command connect*CLI> core show channeltypes I would have response like: connect*CLI> core show channeltypes Type Description Devicestate Indications Transfer ---------- -----------

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

Hello all, I'm setting up a couple of test boxes and I'm running into a problem. What I need help with is determining whether I'm going something wrong or if I need to post a bug report. I have two asterisk 13.0-beta 3 machines set up with extensions connected to each as such: 3700 ----> AST-A <------> AST-B <---- 3800 & 3801 When I place a call from 3800 to

I have one thing not clear to me. If the llvm diect generate object code. Then how about the llvm assembler processing llvm code with inline assembly? Indeed, it's seems now llvm is direct using gas syntax in the assembly code. And because it's generate .S files, so it can be manipulated by binutils. But there is one day, no binutils any further, how to deal with these inline assembly.

Hello List, I have some doubt about direct media settings. I have an asterisk 1.8.14 instance running on 172.20.255.50, a soft phone on IP 10.100.210.51 and a gateway at 10.100.210.254 I have set both gateway and peer to "directmedia=yes" but still on gateway I see RTP from asterisk's IP, have tried setting nat=yes/no and also specifying localnet values but not sure where I am

Hello, I have installed the Xen 4.1.1 and a working kernel 2.6.32.43 which supports Xen on a Ubuntu Linux 10.04 LTS. However, I have never successfully booted the Hypervisor and Dom0. The screen is always black after some kernel messages rapidly go by. I think that I may pass the wrong parameters to the kernel or Hypervisor with GRUB2. Can any one share his/her working grub.cfg for GRUB2 with

Hi, I used Asterisk 1.8 and I have a gsm modem with 8 port. When I called target number, gsm modem and asterisk show me one of these ports active. after hangup, the actived port is going to idl status and ready to use. but after some call from extension, when I want to call another number, asterisk gives me Busy status, however all ports are idle and ready to use. I think asterisk have to

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

On Windows XP, I played the AAF demo (svn://projects.jkraemer.net/acts_as_ferret/trunk/demo) that works nicely with English content. However, if the keyword is non-English (no matter whether there is any content in db), the server immediately causes memory leak over 1GB and stops responding. The languages I tried include: French (utf-8) German (utf-8) Spanish (utf-8) Chinese (utf-8) Japanese

On Jan 28, 2010, at 11:51 AM, Dustin Laurence wrote: > On 01/28/2010 11:41 AM, Anton Korobeynikov wrote: >> >> The required efforts equal to ones required to write new assembler. >> "Too weak to be usable" means "it's not possible to represent many >> important constructs with masm/nasm/fasm". > > Wow. It's perhaps too much of a

Hi, If anyone is interested, I have ported app_rtsp.c to Asterisk 11.x. I have tested it with GStreamer RTSP server and a C920 webcam streaming H264 SVC video from one machine to another machine running Linphone. Contact me at this e-mail address robkrakora at messagenetsystems.com for source code. Best Regards, -- Rob Krakora MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032

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:

On 01/28/2010 11:41 AM, Anton Korobeynikov wrote: > > The required efforts equal to ones required to write new assembler. > "Too weak to be usable" means "it's not possible to represent many > important constructs with masm/nasm/fasm". Wow. It's perhaps too much of a distraction, but I'm curious about the details of this. It's probably because

Hi all: I am a user of "JM" package. Here's the problem of "sample size". The warning is: Error in jointModel(fitLME, fitSURV_death, timeVar = "time", method = "piecewise-PH-GH") : sample sizes in the longitudinal and event processes differ. According to the suggestion of "missing data",I use the same data set(data_JM) without any