similar to: video call doesn work

could anyone help explaining what does this error mean? i get this error when make a video/ audio call from X-lite to Bria prof. phone rtp.c:1739 ast_rtp_read: Unknown RTP codec 126 received from '10.0.0.26' Gres -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does anybody know if i can make (chan_mobile) module to be installed and work with Asterisk 1.4.19 ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080513/57510b8a/attachment.htm

please can anyone help me knowing if i can install Linux and Asterisk on HP servers _________________________________________________________________ Put your friends on the big screen with Windows Vista? + Windows Live?. http://www.microsoft.com/windows/shop/specialoffers.mspx?ocid=TXT_TAGLM_CPC_MediaCtr_bigscreen_012008 -------------- next part -------------- An HTML attachment was

i am asked to implement a call center of 50 seats for my company , and i was wondering if Asterisk can fit this as a relaibale and low price system is it mature enough for this task?? best regards Gers -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090712/b5bab36d/attachment.htm

Hi all, I want to use a cell phone as my FXO line to Asterisk Box ,did anyone try this and configured it and how to physically connect it to Asterisk server? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080506/6fafcfba/attachment.htm

does Asterisk suppoet CSTA protocol for CTI applications? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090722/a2149cc5/attachment.htm

does anyone use OpenSBC , or know if it is mature stable opensource for a production enviroment http://rpm.pbone.net/index.php3/stat/4/idpl/10795970/com/opensbc-1.1.4-3.el5.pp.i386.rpm.html -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091201/67c95aac/attachment.htm

i have 2 asterisk servers one on CentOS and one on Fedora , i configured IAX trunking between the 2 servers so that i dial -say from a sip extension 2000 on fedora server to a sip extension 3000 on CentOS server the call seems to be established but hangup automatically after very short time and here is the iax2 set debug command result on centos server and also my iax.conf and extension.conf and

Can I get some opinions on which free DB to use with rails for a virtual server that will eventually be doing lots of heavy database lifting? Not a lot of heavy duty querying per se. Think multiple simultaneous forums and blog sites and the like. I know each of the different DB possibilities such as MySQL and PostgreSQL has plus points and minus points. I''m particularly interested

Hi Bill, With the latest llvm version I'm getting the following error: opt -instnamer < core-device.c.bc > core-device.o.bc Attribute 'nobuiltin' only applies to functions! void (%struct.fw_csr_iterator*, i32*)* @fw_csr_iterator_init Broken module found, compilation aborted! 0 opt 0x0000000001914bf1 llvm::sys::PrintStackTrace(_IO_FILE*) + 38 1 opt

can i use MP3 files as an IVR prompts directly without converting to .gsm format? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091010/67205e07/attachment.htm

Is there a way to ensure that the source IP address from witch the SIP user register is not tampred with , is there a feild in the SIP register message header can be used to achive this security ? i have an asterisk server in witch SIP users register through an SBC(session border controller) , i wanna make sure that those users are really registering from the IP they are claimming they are

I install CentOS 3.4 but don't found the databases mysql and postgres where are ? Thanks

Hi, I am setting up crashkernel and was wondering where I get the kernel-debug rpm for centosplus kernels. Thanks, -- Stephen Clark *NetWolves Managed Services, LLC.* Director of Technology Phone: 813-579-3200 Fax: 813-882-0209 Email: steve.clark at netwolves.com http://www.netwolves.com

Dear R-community, I'm analysing some noise using the nlme-package. I'm writing in order to get my usage of lme verified. In practise, a number of samples have been processed by a machine measuring the same signal at four different channels. I want to model the noise. I have taken the noise (the signal is from position 1 to 3500, and after that there is only noise). My data looks like

I have port 143 open so that I can get my mail when away from home. Occasionally, though, my router reports things like Thu, 2008-03-27 02:00:11 - TCP Packet - Source:200.122.134.9,3821 Destination:88.97.17.41,143 - [IMAP rule match] Thu, 2008-03-27 05:39:49 - TCP Packet - Source:140.127.181.141,3461 Destination:88.97.17.41,143 - [IMAP rule match] Thu, 2008-03-27 16:10:03 - TCP Packet -

Hello, What is the set of configuration that should be done in the Asterisk 1.0.8 using FreePBX that can allow a simple video call between two extensions? Thanks in advance. [http://www.ericsson.com/shared/images/Email_line.gif] JULIO ARAUJO TE ENGINEER MS Ericsson ITTE & Test Environment S?o Jose dos Campos, Brazil Phone +551239084121 SMS/MMS +551281150089 julio.araujo at

Hi, I've created a new partition on /dev/sda on my CentOS machine after which fdisk -l gives output as: Device Boot Start End Blocks Id System /dev/sda1 * 1 13 104391 83 Linux /dev/sda2 14 3500 28009327+ 83 Linux /dev/sda3 3501 6527 24314377+ 83 Linux Now, when I run partprobe I get the

Hi! I guess I just overread the part I need, I checked the FAQ and online-documentation, but could not find the solution for my problem: I have a server with 1 external IP and a LAN behind, where I need ports to be forwarded like this: external 1.2.3.4:81 forwarded to internal 192.168.1.1:80 external 1.2.3.4:82 forwarded to internal 192.168.1.2:80 and so on .. (for configuring/accessing the

Hi all, I've got one of those cool free incoming IPKall phone numbers from www.ipkall.com. These numbers just connect to the SIP proxy of your choice, they default to Frreworld Dialup. You can use them with your own sip proxy on asterisk. My config for this is below. The trouble I'm having is the incoming calls do not seem to hit the section in sip.conf for the call. With sip