similar to: Memory overcommit

I am having a hell of a time with transfers. First the Snom issues: The transfer button on the Snom 220 does not work. I have read about setting break key off in the advanced page of the web config but the Snom 220 has no such option. At the moment I am having to use the # transfer hack which makes this phone look really stupid to have buttons on it that cannot be used. Anyone know how to

Good day all I'm still looking for a SIP client that will work on fedora core 1? Thanks

Hello, As I understand * don't supports T.38 in Zap channels (please correct me if I'm wrong, BTW is there plans for such support?) I believe it's should support T.38 in "pass-thru" mode. I mean setup like this: Hardware gate with T.38 ------ Asterisk ------ Hardware gate with T.38 But I had troubles with this setup (no faxing) while two gates conneted directly with same

When using buildinstall and CentOS 5.2 to create my own CD minus a bunch of unneeded RPM's and plus a few of my custom RPM's I find that the install CD that gets build does not install an initrd which renders the system unbootable. I also notice that when I run buildinstall with the -debug option it says: unpacking

Has anyone here done this? I got it compiled just fine but when I make a call I do not get any audio going either way. The * box is not behind any sort of firewall or nat. My H323 client (gnomemeeting) is behind NAT but I have it set up properly to work through NAT and it will talk correctly with my other regular x86 box running H323. One odd thing I note is that when looking at the UDP traffic

I cannot get the transfer button to work on a Snom 190, I cannot get the # to work either. Any ideas? Regards Thorben -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041207/5ac06747/attachment.htm

I am trying to get a CentOS 6.3 serial console to work with IPMI SOL. I think I have everything working but maybe the wrong baud rate on the getty: $ ipmitool -I lanplus -U root -H 10.0.0.14 sol activate Password: [SOL Session operational. Use ~? for help] ?????????? 10.0.0.14 is the DRAC IP address of the CentOS 6.3 box. It accepts the password and connects me to the console but then when I

Is there any way to play background music on a sip phone while the phone is not in use like many legacy pbx's offer? Could you take 7960 and use the 6th line in a similar fashion to the all setup maybe? Thoughts ideas? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

I have a client who is looking at replacing their PBX, and I'm interested in putting together an Asterisk solution for them. One feature that would really, really get their attention is if I could do the "Vonage" thing, where if a PSTN caller dials a direct extension (coming in over PRI) both the user's deskset _and_ an external number (their cell phone) would ring, with

Greetings, Xen developers! I have been using AoE with Xen to great effect and I am very pleased with how the whole system has worked out. The two technologies really complement each other. I consider it the ultimate setup. Aside from the network switch (eventually to be remedied) my serving infrastructure has zero single points of failure. However, this virtualization paradise has a flaw:

Hi All, Can someone explain to me the difference between g711's ulaw and alaw codecs? Is it just different header info or is the actual payload in each encoded differently? I have thus far noe been able to find any difinative information onthe matter. All I've managed to find out that they are "similar", they sound the same and that it doesn't matter which you use. Could

Hello all! I am trying to make a stripped down CentOS 5.2 distro which will all fit on a single CD containing just the base and a few other packages I need. I run buildinstall like so: sudo /usr/lib/anaconda-runtime/buildinstall --debug --version 5 --product CentOS --comps /var/www/html/centos/5.2/os/x86_64/repodata/comps.xml --release My-CentOS-5.2 --prodpath CentOS

Can some people post some configurations they've implemented when deploying an * system for let's say 25-50 stations and maybe a larger 200 station system? I would assume some kind of chassis with some DSP boards and some kind of system board with a hard drive for running the system and storing the voice mails - obviously I'm interested in specific chassises and boards used and

I'm considering that board as a mail and voip gateway for home use. In view of all those statements about how little resources asterisk needs, did anybody already try running asterisk on it? Thanks, Bruno.

When we receive or make a call to the outside - they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi -> Asterisk --> network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this

I have dovecot configured to use an LDAP database for mail delivery with dovecot-lda. All of of my user accounts and aliases work fine except for postmaster@ which consistantly bounces. I have compared the entry for postmaster with another working alias and see no difference. When I send mail to postmaster I get the following in the logs: Jun 11 02:11:13 server1 postfix/pipe[12083]: 6D6CF33444:

When I try to flash my 7960 with SIP I get messages like this in the tftp server logfile: Aug 27 02:01:17 home tftpd[32590]: tftpd: trying to get file: P0S3-03-0-00 .bin and the phone says something similar on the display for a brief moment and puts a funny char where the space in the filename above is. Seems like around 1 in 4 of the 7960's I have flashed with SIP have this problem. Anyone

I have several 7960 phones with SIP image (7.3) and Asterisk 1.0.1 on FreeBSD. When I have 2 active SIP calls on the 7960 phone there are no available lines for additional calls. I tried to configure 2 lines to the same SIP server but it's still limited to 2 calls. How to utilize all lines? -- Called user -- SIP/user-acc6 is ringing -- SIP/user-acc6 answered SIP/x.x.x.x-09a9a000 --

I want to DELETE my current installation of CoPilot. When I try wine uninstall REMOVE, I get the message that the program is already installed. CoPilot does not provide an option to uninstall. I tried out of curiosity to RENAME the folder and reinstall an older version that may work better but still it would not let me install, so I renamed the folder back to its original name and CoPilot is

I am a 62 year old beginner in Ubuntu. Thanks for any help, guidance, suggestions on this problem! I have used commands like the following to be certain that I have an active com port: dmesg | grep ttyS rm com1 ln -s /dev/ttyS0 com1 So I suspect the solution lies somewhere in WINE. If all else fails I will get a USB to SERIAL converter cable and try that. My wife has an Abbott Labs Freestyle