similar to: native bridge

Hey, For the bridge issue, take a look at 'notransfer=yes' option in your iax.conf. It'll force * to stay in the path http://www.mail-archive.com/asterisk-users@lists.digium.com/msg42262.html

I think what i did is the same with you except i use make rather than make -sj8, and I did as root. And I do as you wrote again: root at hurricanetong-VirtualBox:/home/hurricanetong/xapian-1.2.17/xapian-core-1.2.17# ./configure [...] root at hurricanetong-VirtualBox:/home/hurricanetong/xapian-1.2.17/xapian-core-1.2.17# make -sj8 Making all in . Making all in docs Making all in tests root at

I uninstall xapian1.3 and install xapian-1.2.17 but i still failed hurricanetong at hurricanetong-VirtualBox:~/workspace$ g++ `xapian-config --cxxflags --libs` demo2.cc /tmp/cc2wsfDJ.o: In function `main': demo2.cc:(.text+0x4a): undefined reference to `Xapian::WritableDatabase::WritableDatabase(std::basic_string<char, std::char_traits<char>, std::allocator<char> >

Hi, I have create one vm by libvirt api, And I goto virsh, when I doing save the domain, the following errors occurred: virsh # save demo2 /home/wk/libvirt/demo2.save error: Failed to save domain demo2 to /home/wk/libvirt/demo2.save error: internal error unable to send file handle 'migrate': No file descriptor supplied via SCM_RIGHTS virsh # list Id Name

hi, virt guys, This is Hongqing from Fedroa QA team. I try to forward the guest installation logs to host with virtio. I have used virsh edit <guestName> to add a channel, it works fine. I think it would be better if I can initialize it when I create the guest using virt-install, and virt-install also offers the option, I have tried below, but it does not work. virt-install

So I have a problem. A customer of mine wants a PBX, owns an office building. I want to sell him on asterisk. He has 4 tenants. I am using my asterisk box to simulate it. My asterisk box has a TDM400P card, not a PRI card. Don't know if it makes any difference. Anyway, I want to route incoming phone calls to different contexts based on the phone number being called. Here is my

On 12 Feb 2014, at 14:58, "Hurricane Tong" <zhangshangtong.cpp at qq.com> wrote: > hurricanetong at hurricanetong-VirtualBox:~/workspace$ g++ `xapian-config --cxxflags --libs` BuildIndexDemo.cpp You want to use xapian-config-1.3 here, not xapian-config (which will be the installed version, not the trunk version you built). I've just built trunk (on Mac OS, admittedly),

We encouter a situation where we need to use SIP info to convey infomation for one end point to another endpoint. I use asterisk to do the test and find asterisk does not forward the SIP info to another endpoint, but act as UAS and returns a 4xx error message. I think asterisk is not right to handle this SIP info message. In RFC 3261 Page 70 "This protocol is designed to be extended.

Hello all, Not sure if this mail belongs to this users or dev list. Sorry about that. We have the following scenario: PhoneA OpenSER Asterisk PhoneB PhoneC | | | | | | | | | | | | | |

Hi I have noticed that asterisk returns 'SIP 603' when the called party does not answer. My test setup is simple: two SIP phones (extensions: 100 and 111) registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30 seconds. When 100 calls 111 and after 30 seconds, asterisk sends a CANCEL request to 111 (expected) and a '603 Decline' response to 100 (unexpected &

Hi, again. Are there any updates to the development of Theora? Like a version 1.2? I ask this because I want to be re-assured Theora is not just sitting while people are choosing H.264... -- Daniel Hendrycks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/theora/attachments/20100320/3c18a22c/attachment.htm

I am having a problem registering my cisco phones which is exactly like that described in http://lists.digium.com/pipermail/asterisk-users/2011-May/262306.html except that I am on Asterisk 1.8.3.3 and using sip level POS3-07-4-00 The symptoms are: o 7960 lines show [X] o Outbound calls can be made from the phone, including call pickup of inbound calls, but not to it. o Trace shows REGISTER

Guys.. I have a problem with 2 asterisks connected together but each time one tries to call another I get this: -- Executing Dial("SIP/fozy-dfbc", "IAX2/voip-gw/201|60|mwtWT") in new stack -- Called voip-gw/201 -- Started music on hold, class 'default', on SIP/fozy-dfbc -- Stopped music on hold on SIP/fozy-dfbc -- Call accepted by 10.0.101.1 (format

Hello, I saw a behavior on shares, where "msdfs root = yes" is set and wanted to know if this is a bug or expected. The following I've done on a 4.1.9 member server: * Create a share: [demo] path = /srv/samba/ msdfs root = yes browseable = yes * Create a subfolder "demo" somewhere on the shared directory on Linux and make sure, that a group you

I recently purchased a Sipura 2000 and connected a phone to it which is connected to my asterisk box via sip. Calls to the Sipura 2000 work fine from another sip device connected through *, from either an fxo or fxs (via adtran channel bank connected to a T400P card) port. However, when a call comes in from the phone company over a T1 with em_w trunks, the phone on the Sipura will ring but I

It got planned, and then canceled. I assume you guys are going to try again? I was looking forward to attending.

Hi! Is thear any tool dedicated to analyze theora stream, like H264Visa for H.264? Or any other tool helpfull for analyze? I want use it to analyze products of my developing hardware implementation of theora encoder. Thanks all! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/theora/attachments/20110321/8e875089/attachment.htm

Hi list, I just came across this site: http://web.mit.edu/xiphmont/Public/theora/demo.html I was wondering are these issues being addressed or will be addressed before the 1.0 release? Theora is like the only open sourced free code people can use, i will be nice if it performs close to other libs. please keep up the good work! -------------- next part -------------- An HTML attachment was

Hello Where could I find examples on how to work with the time index in a timeseries or zoo series? Let say I've got this series DATA 1990-01-01 10:00:00 0.900 1990-01-01 10:01:00 0.910 1990-01-01 10:03:00 0.905 1990-01-01 10:04:00 0.905 1990-01-01 10:05:00 0.890 ....................... 2000-12-31 20:00:00 0.992 How do I make simple calculations such as ... ? Calculate the

Hello, I have the following setup: (*)<--->IP<--->Micronet 5012 H.323 box <---> POTS <---> PBX (Alcatel OmniPCX) Grand idea is to use the micronet's POTS interfaces to connect SIP phones to the PBX and to the PSTN. I think i even managed my way in the arcane and cryptic management interface of that appliance, but I am stuck against theese messages: -- Executing