similar to: URGENT - micronet & asterisk on h323

hi all, I don't know how to setup asterix to work as PBX. If I want just basic configuration with 2 SIP phones (snom, ata), what all I have to write in the configuration files, or respectively in the configuration of ata and snom ? If there is any good documention available, send me URL too. All (ata, snom) are behind firewall (nat) and astrix is on the public IP, but I can move for

hello, can anybody help me with folloving problem I have asterisk with the public IP and two UAs (snom100, x-lite) in the same private network behind the same FW/NAT. All is working good, but whan I tried to establish call between these two UAs, first 10-15 second is nothing to hear and then is the quality terrible :( Can anyone tell how to get it work with normal quality ? best regards

hello, my questns are about few * functionality. 1) how can I make call tranfer. Not call parking. If I'm talking with some one a I want to tramnfer call to the another extension, to the other person. 2) how can I make call confernece. Not Meetme If I'm talking with some one and I want to join another person to our talk . I haven't found this in any manual :( hudecof -- mail:

Hello. I have a codec problem to send calls from a SIP device to a H323 gateway. First I'll explain the scenario: - Asterisk 1.2.1 - The SIP phone can use any codec I want. - The H323 gateway can only use g729 (cause it's not under my administration) - SIP phone has g729 configured, so my asterisk doesn't need to "transcode" (I don't have licences for g729) - sip.conf

hello, my situations is as follows. In our comapny we are planing to have *. I'm testing it now. If we will buy G729 codec for * ... UA(SIP) <-> FW <-> (SIP)*(H323) <-> (H323)GATEKEEPER(H323) <-> (H323)AS5300 <-> world the following equipment speeks G729: *, GK ,AS5300. All call from UA to another endpoing go through *, because of mixed SIP and H323

Hello, I connected a Micronet SP5014 2FXS + 2FXO gateway to the asterisk, the problem is i can make call but can't receive calls. If i make a "sip show peers" it shows the micronet is not connected to the asterisk. Does anybody knows how to configure the micronet and asterisk to solve this problem ? Thank you

I tried setting up a Micronet 4port 2 fxs 2fxo device using sip. I found that I could get multiple fxs ports to register with *. I also found that no matter what dialed on as an fxs user when dialing out it would try to go out the fxo side of the micronet. I also had major problems with these devices behind nat. I was having so much fun with these little boxes that I put them back on the shelf

Hi, Has anyone out there managed to do a hookflash transfer with a Micronet 5050s gateway ? We have just tried out this gateway and it seems to do everything we need except this particular feature. Also if you have succeeded where is the hookflash time specified in the gateway. There does not appear to be any parameter for this. Perhaps it is not supported at all. Any help appreciated.

Hi Charles! Thanks a lot for your response. Now I figure why the isbmex driver was not working. On the other hand I've tried to run the command suggested. But no success. I got this information when running the dmesg command after running the modprobe $ sudo modprobe ftdi_sio vendor=0x0483 product=0x0035 [35816.111760] usbcore: registered new interface driver usbserial [35816.111792]

On Feb 26, 2014, at 2:10 AM, Victor Romero wrote: > I got a UPS Sola Basic but it is not working, I'm providing the details. I hope that someone can help me to get it working. > > Please let me know if there is some other information that I should provide :) Thank you for providing all of this information! > OS name and version > Ubuntu 12.04 > $ uname -a > Linux

I got a UPS Sola Basic but it is not working, I'm providing the details. I hope that someone can help me to get it working. Please let me know if there is some other information that I should provide :) OS name and version Ubuntu 12.04 $ uname -a Linux olimpo 3.8.0-29-generic #42~precise1-Ubuntu SMP Wed Aug 14 16:19:23 UTC 2013 x86_64 x86_64 x86_64 GNU/Linux Exact NUT version, 2.7.1

Hi I have a Firewire connected Micronet 1.5TB RAID with a single large ext3 filesystem on one partition on a dual Xeon system. I am checking out from an extremely large cvs repository (don't ask) to this drive over the course of many days, and intermittently I get bad blocks and the filesystem goes read-only. This is not related to any power failure or anything similar. The RAID is currently

hi, I known, that this is not * related, but a lot of members of this ML uses GS BT phones. I have patched the atftp serber to recognize the TFTP OPTION, whis these phone send during boot. Patch includes - another locations for configs, firmware and ring tones - different FW versions for phones - custom ring tones for the phones You can find patch, source/unpatched/ and DEB for

Hello all, I''ve been trying to test HTB performance for different link bandwidths to find potential limits and this is what I have so far: http://home.comcast.net/~msethuraman/htbtest/ Can members please go over the setup, test procedure and the results and answer a few questions? 1. Is the testing methodology okay and can the results be considered accurate? If so, is this a decent

hi all, I known taht this ML is about *, but lot of you are using BT telepnones. I'm using FW 1.0.6.7 for all phone. This firmware support VLAN tagging for QoS on the Layer2. I use it to separe the PHONE network form the PC network, which are in PC connector in the BT. And I also use the TFTP server for provisioning (software, configuration). The problem is, if you set the VLAN for QOS.

Hi all. I''m going on my 3rd week trying to get a simple traffic shapping to work the right way :( !! My goal it to shape the traffic coming from one machine (pc1) to another machine (pc2) throught the "eth0" interface. My test configuration is as follows: PC1   IP: 192.168.105.237   Mask: 255.255.255.0   OS: Red Hat Linux Kernel 2.4.20-8   Rules:

Dears; I am having same problem, that when I place a call from the H323 end point (even if it is not added in the ooh323.conf), then asterisk handle the call and play the wave file in the default context. Also I added endpoint to the ooh323.conf and same thing, it keep goes for default context whatever the context placed. My Asterisk vesion is 1.4.25 My Asterisk add-on version is: 1.4.8 What I

Is Asterisk capable of handling video conference? I am wondering if there is anybody in the list who tried it with NetMeeting(s). If it is possible, is the * required to register in the GK for this purpose? or making it as h323gw only is enough. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello freebsd-stable, i know this is may be the wrong place to ask these question, i just cvsup my 5.1 box today, and during the make installworld it's stop with these following error messages: PASS: Test notdef detected no regression. Running test modifiers PASS: Test modifiers detected no regression. Running test funny_targets PASS: Test funny_targets detected no regression.

Well? I?m not convinced KEA can?t be simple DHCP, though I understand that one would never know that from reading the KEA documentation. The following is a complete KEA configuration suitable to do a single subnet. In fact, you could probably get away with slightly less. I?ve replaced the unique addresses from my environment with placeholders (e.g. <network/mask>), but otherwise, this is a