similar to: GSM codec is a good choice ???

Hello, I am trying to make a load balance at my box using two conections. I have compile my kernel with this patch routes-2.6.13-12.diff (tha I get from this website: http://www.linuxvirtualserver.org/~julian/#routes). The problem is that when I try to balance using weight sintaxe (i will put the script bellow) some conectios just drop. So I can enter some pages but other I could not...

Dear, I need to create a custom device extension in order to dial a local queue. Suppose my queue number is 8888, how can fill the Dial field from the custom extension ??? Because if I put just 8888 or Local/8888, I don't succeed. Thanks a lot

Dear all, I have to install an E1 card in my Asterisk 1.4.23 server and here is my short question: Is it necessary to install or update any Asterisk/Zaptel/Any extra module or the default installation is good enough to just plug and run the E1 card ???? Thanks a lot Alejandro

Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and Celeron), and last days when I call from one extension to another of the same PBX after I dial the number the rings sound after 20 seconds. In the CLI log, when I debug the AGI, I see always goes good until dialparties.agi, and after that there are 20 seconds without any log, and so the ring sound. I've read

I set the following in smb.conf: encrypt passwords = true unix password sync = true As says smb.conf # This boolean parameter controls whether Samba attempts to sync the Unix # password with the SMB password when the encrypted SMB password in the # passdb is changed. 1) Always I had understood if I change the smb passwd, the unix passwoed in /etc/passwd did too. However this does not

Dear all, I'll install Asterisk 1.4 in an IBM xSeries 226 server with four HD's available, using CentOS as the OS. What's the best RAID type recommendation ??? RAID 1 or RAID 5 ??? Regards Alejandro

Dear all, I have an Asterisk SIP server in a LAN environment and I want your opinion in order to decide the use of an audio codec: What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip calls ??? Thank you !!! Alejandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, > history() gives Error in savehistory(file) : no history available to save although I can scroll throu history with C^uparrow an C^downarrow. How can I make history() work and/or show the current history in a file, so that I can choose from previous commands? The web did not throw up anything useful. TIA --Christian -- Christian W. Hoffmann, CH - 8915 Hausen am Albis,

Hi, I was wondering if anyone could tell me how m and s are calculated for a t distribution? I thought m was the sample mean and s the standard deviation- but obviously I'm wrong as this doesn'y give the same answer. Thank you -- View this message in context: http://www.nabble.com/fitdistr-for-t-distribution-tp23550779p23550779.html Sent from the R help mailing list archive at

Hello, I'm having issues connecting throu PRI with the following error "Requested transfer capability: 0x00 - SPEECH" Below are the logs: == Using SIP RTP CoS mark 5 -- Executing [97052660 at voipphones:1] Set("SIP/4856-00000003", "CALLERID(num)=xxxxxxxxx") in new stack -- Executing [97052660 at voipphones:2] Dial("SIP/4856-00000003",

People, I will install asterisk on my Debian Etch box without a PSTN interface card. I want to use only softphones for the moment. My question are: 1) Is it enough to install with "apt-get" the asterisk 1.2 or do I have to get asterisk 1.4 manually ??? 2) Do I have to configure a dummy PSTN interface in my case ?? And if you have a debian-asterisk howto, I really thank you. Regards,

Dear all, I have an Asterisk SIP PBX using GSM codec at peers and in voicemail sounds files (I have Spanish sounds). But now I have a problem because I have to use G.729 mandatory at peers, and I have GSM in voicemail sound files. I can't let Asterisk do trascoding because I have no a DSP in the CPU, and I don't want to degrade the PBX performance with trascoding tasks. So how can I

Dear people, is it possible to have presence using Asterisk 1.2.13 / SIP with Linux/Debian Etch??? I'd like to see if my intranet contacts are available, busy, disconnected.... Thanks a lot Alejandro

Dear all, I need to build a PBX based on Asterisk for a call center. I have worked with raw Asterisk but it's hard to work for big implementations think. Also I have worked with Trixbox CE for a small bussines and it was prette good, but I have not have many features like ACD. I know there is another version called Trixbox PRO -specially Call Center edition- that's not free but has got

Just a quick update. The cat.c in the com32/modules directory works fine. Just added a printf("\nPress Enter to Return to Menu\n"); at the end since I had some uses display the messages, and then not release all they had to do was press enter to rerun the menu. The display.c32 still does work, but know it doesn't lock the system so only a hard reset works. Ctrl-Alt-Del works. I

Dear all, I have Asterisk 1.4 and I'm using the voicemail feature. I edit /etc/asterisk/voicemail.conf with "envelope=yes" and after that I left a message in a given mailbox near 11:00 AM. When a dial the voicemail number in order to hear the message, the Astreisk server close the cal and I get this error from te CLI: [Apr 10 14:09:08] WARNING[12955]: file.c:563 ast_openstream_full:

Hello everyone - I am trying to use virt-viewer to connect to KVM virtual machines running on a CentOS7 host. It works great when running directly on the host, but I have not been able to figure out the magic connection string to make it work from another computer. On the host, I set selinux to "permissive" and stopped the firewalld service. No change, so it is not related to

Hi, I am new to VPN an OpenVPN with shorewal. I tryed a lot and read a bounch of howto''s but nothing helped so I came here. I want to tunnel all request to my server 141.48.XXX.XXX from my home network throu port 443. I want to do this because this is the only way I can connect to my server using ssh or ony other tool or port. On Port 80 Apache is running, so I only have the https port

Dear all, I'll implement a VoIP system using Asterisk + SIP with softphones; I need to connect LAN and VPN users (about 100-150). What version/installation of asterisk do you recommend tyo me ??? Does Asterisk@Home or Trixbox match to my scenario ???? By the way, I use Debian Etch as OS server. Really thanks. Alejandro --

Hi, I have a rather simple problem I have to solve, well I thought it would be simple, I''ve run into a problem. I think I must be missing something fundamental. I am trying to build a Linux router that simply throttles everything down to certain bandwidths. That is, no priority queuing ...etc, just slow all traffic down to the specified rates, which are 64,128,256,512 kbit. We want