similar to: Got SIP response 405 "Method not acceptable" back from xxx.xxx.xxx.xxx

Hi, I have a raid 1 setup where /boot and the rest lies inside the md0 container. /dev/sda1 md0 has /boot /dev/sda2 md0 has swap /dev/sda3 md0 has / SCSI card is an ADAPTEC AIC79xx Grub as the bootloader Centos 4.1 as the OS version Two 70gb SCSI disks The server was working fine. Then a power outage. checked the board,disk,ram,etc. All fine. Now when I try to boot the server does not even reach

Hi, im using 0.99 stable in RPM form, Im using this for postfix+dovecot+real unix users, now i tried to enable virtual support with this: protocols = imap pop3 ssl_disable = yes log_path = /var/log/dovecot.log info_log_path = /var/log/dovecot.info login_user = dovecot mail_extra_groups = mail auth = default auth_mechanisms = plain auth_passdb = pam auth_userdb = passwd auth_userdb =

the user you are connecting as should have full rights to /var/run/asterisk: http://www.voip-info.org/wiki-Asterisk+non-root hth -----Original Message----- From: Erick Perez [mailto:eaperezh@gmail.com] Sent: Monday, April 03, 2006 9:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Unable to connect to remote asterisk (does /var/run/asterisk.ctl

Hi, haven't found anything in google's, i wonder if there is a comparative page of what to expect from running * on motherboards like the EPIA and similar ones. Since i have not used *ever* such kind of mini atx form factor boards, I have no clue about their performance. SIP-SIP communications, voicemail SIP-TDM communications, voicemail how may users (SIP hardphones and analog phones

Just a question, has anyone knows how to blank or factory reset an AudioCodes MP124-C/FSX/AC/SIP unit (it's a 24 FSX to SIP unit). I purchased them second-handed with no manuals (thank god for the internet!!) but i guess the pdf manual I have does not have the section of factory-reset. Also, any sucess stories with: AudioCodes MP124-C/FSX/AC/SIP

Hi, today a customer asked how to use asterisk with 12 to 16 FXO ports. I can use a channel bank with 16 FXO ports and connect the channel bank with a T1 cable to a T1 card in the Asterisk Server. Asterisk will then send the calls to the Voip provider over the internet. However a 16 fxo port channel bank is about USD 1500 + a t1 card USD 500 + a USD 1000 computer = 3 thousand us dollars + my

Hi, emails to astricon.net seems to bounce (at least for me) I need information about proper & authorized Asterisk training in the Miami, FL area and the possibility of later DCAP testing. Thanks, -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

I've been playing for hours with the dovecot.spec file to build --with-mysql. The resulting RPM builds fine but when dovecot runs it does not recognize the passdb mysql statement. I also noted that the spec file has a version of 1.0-stable and rpmbuild complains about an invalid '-' in the version. Being a newbie in rpmbuild, can someone assist me in building for CentOS 4 on i386?

Hi folks, an Asterisk (www.asterisk.org) box will be built with the following specs and I wanted to know if CENTOS has a built process/config/parameter/special distro to target the AMD Opteron Dual Core plataform: 2 amd opteron 265 dual core 1.8 Ghz 2mb l2 cache 2x1024 ocz c2-6400 dual channel gold GX XTC 2 Seagate 16mb cache 300 GB barracuda 7200 drives another 2 Seagate 500 GB 16mb cache

Hi, I use putty in my windows xp machine. ssh server in a centos 4. The centos 4 machine runs a web server that listens on port 1812, the centos machine is behind a firewall that allows tcp 22 connections only. I am on public internet. Can I forward/redirect/allow my web browser in windows to "see" the web page in port 1812 of the centos machine via the SSH connection? Thanks, --

Hi people, i've seen the wiki looking for a * billing solution but the links point to websites that have not updated their content (or news) section for over a year. Can anyone recommend a commercial-grade (i mean no mom&pop cdr system) billing solution that can start small and then scalate as traffic grows and tested/used with Asterisk before? commercial or open source links are ok.

I just found a Dialogic VFX/41JCT-LS (4 analog ports) in a drawer. I can use it in my house with asterisk at home project. Can I use that with asterisk? Where can I download proper drivers? -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama -------------- next part -------------- An HTML attachment was

Where can I find RPMs for RHEL 3? -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama -------------- next part -------------- An HTML attachment was scrubbed... URL: http://dovecot.org/pipermail/dovecot/attachments/20051024/b30a1923/attachment.html

Hi, I have my httpd process that starts in runlevel3 with the following info # chkconfig: - 85 15 # description: Apache is a World Wide Web server. It is used to serve \ # HTML files and CGI. # processname: httpd # pidfile: /var/run/httpd.pid # config: /etc/httpd/conf/httpd.conf the filename is /etc/rc.3.d/K15httpd However I installed an antivirus that uses a tomcat process that

I had the same problem and I believe it was the payload size of the codec. What code are you using? ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Federico Alves Sent: Friday,

should I use mpg123 with asterisk 1.2.7 or should i use the native player asterisk has? the target machine will receive heavy load. also, has anyone succedded in compiling mpg123 in a dual core pentium with centos 4.3 ? -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

As stated here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe A Meetme room uses Ulaw as the audio codec, so if the other channels use different codecs, then * will transcode. Does the app_conference application works the same way? Or if i have SIP/g729 users and i create a conference with other users also at g729 asterisk will not transcode (when using app_conference)?

Hi, I want to start managing my asterisk boxes with a centralized graphical based interface so I can (due to customers request) give control to customers to add/change extensions to their current PBX intallations such as (not complete list) Add/del/mod extensions sound recordings (ivr or voice attendants) email to fax/ fax to email voicemail to email SIP and ZAP, no IAX needed configure calls

Hi, does anyone know the prices for g729 codecs from Digium? I sent an email a while ago to them but haven't got any response so far. Prices are per unit or volume? Thanks, -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

What host based ids is available in centos repositories? the i can install with yum install package? -- ------------------------------------------------------------ Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ------------------------------------------------------------