search for: stevefink

...root=/dev/ram Everything is okay on the DHCP end, as I mentioned prior, that I can pxeboot without pxelinux, I just can't chainload the pxeboot or load a kernel. Any help at all would be GREATLY appreciated as this isn't a setup with a lot of documentation out there. -- Steve Finkelstein stevefink at gmail.com

Hi all, I'm looking for some rudimentary insight on GotoIf() which seems to be failing on me in my dial plan. All I basically wish to do is block a particular caller. Sounds easy enough, but my ternary operator/plan currently is not properly being implemented. Can anyone spot where I'm being a momo? All extensions get forwarded to the following macro: [macro-forward] ; arg1 = phone

Hi all, There's a myriad of options these days and I haven't been keeping up to date with what's respectable any longer. I essentially need a provider that will provide me with one DID to start and let me trunk over SIP or IAX2. I'd like to obviously trunk with Asterisk on my end and have full control over the dial plan. This way I can branch out my DID into extensions and have

Hi all, I have a budget to work with and was wondering if there are any folks providing SIP/IAX2 trunking for unlimited inbound/outbound for a flat rate? We're in the budget range of roughly $5,000 a month and we need multiple channels per DID. I'm not sure if something like this is feasible in the world of VoIP -- and I only need to be able to make domestic/USA calls. Thanks for any

Afternoon all, Is there any particular reason why yum fetches rpms for two architectures on almost any update/install I'd like to perform? This includes both i386 and x86_64. Here's a small excerpt after performing a 'yum update' cups-libs i386 1:1.1.22-0.rc1.9.18 update 107 k cups-libs x86_64 1:1.1.22-0.rc1.9.18 update 112 k

Hi folks, Is Xen 3.2.0 available on any CentOS repository or perhaps third party repository? Will using this package require compiling from source? Thanks for your help. /sf

All, I'm trying to hear the asterisk's auto attendant in its default configuration. According to VoIP Hacks in Chapter 4, I found the following excerpt after successfully configuring my SIP IP Phone (Cisco 7960G): In its default configuration, Asterisk has an auto-attendant that can route calls. To try it out, take the IP phone off the hook and dial 2. Then dial the BudgeTone's Send

Hi all, This is a simple concept, however I'm not entirely comfortable with available applications and functions available to me to make this happen. I have a simple dialout macro such as the following: [macro-dialout]; arg1 = callerid number; arg2 = phone numberl exten => s,1,Set(CALLERID(number)=${ARG1}) exten => s,2,GotoIf($[${LEN(${ARG2})} = 10]?3:4) exten =>

All, Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. I'd like to have the SIP phone ring for some arbitrary number of seconds before it is sent off to the mobile phone. Using something like a Wait() within a Dial() would be ideal. Any suggestions? - sf

Using a quad core 1.86GHz Xeon CPU here, running Asterisk 1.4.2. Noticed the following: Cpu(s): 4.3% us, 95.4% sy, 0.0% ni, 0.2% id, 0.0% wa, 0.0% hi, 0.0% si 30908 asterisk 18 0 188m 10m 5152 S 400 0.3 51051:13 asterisk Asterisk is eating up all the cores running the CPU at 400%. Is there something broken in 1.4.2 that needs to be addressed? Any suggestions? There's

Hi folks, I was recently made aware that the only way to currently set custom fields in a relational database for CDR is via the experimental cdr_adaptive_odbc drivers found here: http://svncommunity.digium.com/view/tilghman/branches/1.4/cdr_adaptive_odbc.c?view=log I had no problem compiling the driver, and copied the module to /usr/lib/asterisk/modules on my box and setup

Hi all, Is it possible to do the above yet? I'd much prefer this method over doing a clean install (although I know this is the preferred method.) Thanks, - sf

Hi all, I believe the last yum update I performed installed a new openldap-client for me, including SASL support. I used to be able to perform ldapsearch without explicitly passing '-x' to the command line to pull the directory down. Now, I receive the following: ldap_sasl_interactive_bind_s: No such attribute (16) Would my best bet be to find a source RPM and rebuild the ldap client

Hi all, I've compiled zaptel drivers and reconfigure asterisk afterwards from source --with-zaptel. Modules are loaded accordingly: asterisk-1.4.2 # lsmod |grep z Module Size Used by ztdummy 5472 0 zaptel 194504 5 ztdummy crc_ccitt 3521 1 zaptel my musiconhold.conf: asterisk-1.4.2 # grep -v '^;'

Hi all, I have the following in my extensions.conf: exten => s,4,GotoIf($["${CALLERID(number)}" = "8585979857" | "8585970327"]?15:5) The numbers listed above are known spammer numbers. However, when I call from any other CALLERID, it still directs me to s,15 which is the Hangup() application. Here are logs from the asterisk CLI: -- Executing

Hi all, I've recently upgraded to Asterisk 1.4.2, coming from 1.2.14. Using my existing Cisco 7960G handset(s). I've tried multiple installs of asterisk 1.4.2 with multiple handsets and SIP will not authorize my phone. I'll include some verbose log messages below to show a VALID registration and one where I'm having difficulty registering the phone. Thanks to anyone who can lend