search for: jrob

Has anyone set up Group/Broadcast Voicemail for 50 or more mailboxes?

> -----Original Message----- > From: Robert Goodyear [mailto:me@jrob.net] > Sent: Tuesday, March 22, 2005 1:21 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Is there a way to get inserted into an LEC's > CLIDB? > > > Does anyone know if there's a service out there to -- for a fee -- >...

Hi. I'm having a terrible time with call quality coming into my * box. I'm using VoicePulse over a 1.5/1.5 mbit line. Outbound calls are crystal clear on both the RX/TX sides of the conversation. Inbound calls, though, are HORRIBLY garbled on the RX side. I can barely hear the caller, but they report my quality is fine. Getting loads of garbled sounds and weird echoes. (Could just be

Does anyone know if there's a service out there to -- for a fee -- inject our DID into the LEC's CLI database so a called party gets our associated name? /rg

Does anyone know how to suppress the "Missed Calls" indication -- perhaps on a per-line basis -- on the 7960 running SIP? Reason: I've configured a group of extensions to ring for inbound calls and it seems pointless to accrue missed calls on those line presentations. /rg

I know a SIP client behind a NAT trying to peer with Asterisk behind another NAT is troublesome. Has anyone had any luck doing this by interfacing Asterisk to the WAN using 1:1 NAT translation to give it a public IP while still firewalled? In my instance I'm using m0n0wall, but this is a hardware-neutral question. Thanks. -- Robert Goodyear Brand Up LLC http://www.brand-up.com

Anyone know how to bypass the CONFIRMATION of the user announcement recording in MeetMe? While I like the "please say your name" to announce a user into a conference, I find it confusing and time consuming to make the user to press 1 to accept a recording they haven't even previewed. I'm not a coder, but I'd be happy to comment out the confirmation loop if someone

Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk

I googled on this for about an hour and the most relevant hit I got was, of course, the first hit: http://www.sowerbutts.com/linux-mac-mini/#support In it, he indicates that the stock Bluetooth module "should work, but untested" - he doesn't qualify the statement with anything. Has anyone tried chan_bluetooth or even the Bluz stack on a Mini or a G5? If so, under Linux or OSX?

Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same

I know this is a bit off topic but we are using Asterisk :) Since this list is full of tech gurus w/ all different sorts of backgrounds, I thought I would get the best opinions here. We have several different switches and other telecom equipment at our facilities which all have their own proprietary cdr platforms, which are rather limited. The company I work for is looking to develop their

Hi: I'm a bit confused about CODEC declarations in IAX.conf. In the [GENERAL] section, should I declare NO codecs and no [allow|disallow all] statements in favor of declaring all CODEC rules within each context section? If this is the case, I'm wondering then what exactly should remain in the general context... should it be only PORT bindings and TOS flags? Thanks in advance, /rg

I'm trying to use the SPEEX codec with Voicepulse. Here's what I see in the CLI when I RELOAD: -- Reloading module 'codec_speex.so' (Speex/PCM16 (signed linear) Codec Translator) == Parsing '/etc/asterisk/codecs.conf': Found -- CODEC SPEEX: Setting Quality to 5 -- CODEC SPEEX: Setting Complexity to 5 -- CODEC SPEEX: Perceptual Enhancement Mode.

Has anyone ever seen Asterisk fail to parse files referenced by an #include by a *.conf command? e.g.: #include /etc/asterisk/sip-phones.d/*.conf Where the dir sip-phones.d contains sip extension conf files. This worked fine for nearly a month and then mysteriously stopped working for me last night! Regards, /rg

What's the recommended way to show my exact build of Asterisk -- down to the minor-minor version number? I ask because I am setting up a small testbed and need to keep myself straight and would prefer something more authoritative than a post-it note and my addled memory. If I do Asterisk -V, I seem to only get the point release and one decimal beyond, e.g.: 1.0.5 shows as

I've watched the dialogue about how asterisk has to manipulate the packets from an IAX2 connection to a SIP client. That said, I'm wondering if a previous problem I've been trying to diagnose could be related to that process. In short, here's how I describe it: Outbound: SIP 7960 > Asterisk > IAX2 Audio is perfect both directions. Inbound: IAX2 > Asterisk > SIP

Has anyone experienced a dropped call when bridging? I get an "OK, ready to transfer" from both channels, but when asterisk releases the call, it is dropped immediately by the upstream provider. I've tested against another provider and it works fine, and it also works fine across two different providers, including TO and FROM the one that's acting buggy. Here's a

Has anyone out there scripted the rollthrough migration of the Cisco firmware? It would be fantastic if there was an app that would generate a set of templated .CNF and XML files based on the MAC addys entered, then control and present your .BIN images through TFTP. It could then also send the reboot signal too, walking through the oh-so-ridiculous path from 3.2 (which every 7960 I've

Is is possible to specify the MOH Class when defining a MeetMe extension? I tried exten => 300,1,MeetMe(300|M(class)) But that did not work. Thx, -Rob.

Has anyone out there tried to mix the envelope metadata for voicemails into the audio payload that's stored by Asterisk? I would like to have the CID and Timestamp baked into the beginning of the WAV file, not just as text in the email itself. Thanks! -Rob.