Andrew D Kirch

This works perfectly.

I have a phone system with extensions in the 200 range and an option 2 for sales. I have set Extension Input in my AA to dial after 3 digits (201 202 203 etc) Instead it matches the 2 and dials the Sales person who is quite annoyed at the call volume he is receiving. The system info is below: License Status: allthingsIT License Duration: Permanent Version: 2.0.2.1675 (Unix) Working Directory: /root/pbxndata IP Addresses: redacted default redacted MAC Addresses: 00065BAF6788 0007E9189E9D Calls: 742/518 (CDR: 1552) Uptime: 26 20:42:47 (23177) WAV cache: 0

"Please enter an extension or stay on the line for additional assistance" Could we possibly have an option to make the above go away? I want customers to enter an extension, not be rendered into a vegitative state by the above message repeating over and over and over again with roughly a second of silence between repetitions.

Having seen quite a few users go glassy-eyed when I say "ooh just hit *99 and it'll take care of it", I agree that star codes are confusing. I've spent about 20 minutes this morning playing with shared line appearances on the Snom phones to no avail. I set up CO lines for my two trunks (local x2) (longdistance x1) , and used the co lines names as shared lines in the snom phones. After this didn't work I went over to the wiki and couldn't find any documentation on shared lines, line grouping etc. Is there any possibility of getting some of this added to the wiki?

With a large number of accounts and/or aliases, PBXnSIP becomes hard to navigate. It would be much quicker for doing call routing and autoattendant routing if after the system has 25 extensions I can filter for each of the different types of account (extension conference auto attendant ivr etc), and filter out extensions and just show the "special" accounts.

Sangoma is available for Windows, and their hardware uses kernel/windows drivers that support a common UI called WANPIPE. I've used the hardware (I have an S518 ASDL card handling my entire network on a Linux router) and they are unbelievable. My concern here is that we'll get into an Asterisk scenario with a timing dependant situation. Sangoma cards feature multi-channel T1, T1, and POTS interfaces with Octastic Echo Cancellation (pretty much the best stuff on the market). Having native PSTN interfaces for PBXnSIP holds it back from Asterisk as a competitor and forces the use of quirky third-party equipment such as Audiocodes. Asterisk requires a PRECISE timing system accurate to 1/8000th of a second! This is a absolutely an Asterisk limitation, not a fault of Sangoma which could roughly care less about timing as it generates timing internally. I think the Sangoma stuff needs to exist in a separate thread and use some sort of inter-process communication to move the voice data to PBXnSIP outside of REALTIME. to prevent the invention of a timing issues and clock jitter/skew. Perhaps reuse PBXnSIP's sip stack to take the data from WANPIPE "SIPify" it and have PBXnSIP use a sip trunk to communicate across the loopback. Just a thought.

If you can avoid the calling card mentality (and I understand if you cant! ) Broadvoice (www.broadvoice.com) has international calling for $19.99/month flat rate. I have used them for 2+ years and with the exception of an upstream carrier issue they had at the beginning their service since has been wonderful. Maybe you should take a look. (No I'm not a BroadVoice schill)

I'm a long-time Asterisk user and always considered the parking lots in Asterisk well implimented. I can stack up calls in parking slots and just let the recipient of the call know which slot their call is in so that they can pick it up at their convenience. From what I can see in the WIKI however, PBXnSIP seems to be only able to support one parking slot per pickup group. It really should be trivial to have a larger parkinglot in the format of *86<slot> or *86<group><slot> allowing for multiple call parking and retrieval.

The flipside is, if you aren't using SNMP monitoring with PBXnSIP, why is that? What values does the PBXnSIP SNMP support lack which would make your life easer? I for one would like to see the following OID's added total voicemail voicemail by account (these allow me to ensure my managed PBX users are checking their voicemail because the boss wants to know that calls are being returned) system load (since PBXnSIP overrides the system SNMP we need some health values from the system, perhaps they could locate the standard MIB folders in windows/linux and announce them as well?) trap on dropped call (send an SNMP trap somewhere if the system thinks a call dropped) 5 minute system call load average (running adjusted average of the number of concurrant calls on the system, and no I don't want to have to hack my monitoring software to make this work ) Let me know what you guys think. I personally believe (unlike others) that PBXnSIP is ready for the prime time, and that means monitoring and accountability. Regards

Microsoft Mail Internet Headers Version 2.0 Received: from localhost ([xxx.xxx.xxx.xxx]) by AllThingsIT.com with Microsoft SMTPSVC(x); Mon, 16 Apr 2007 14:38:01 -0400 From: <pbx@allthingsit.com> To: <xxxxxxx@allthingsit.com> Subject: Voicemail from xxxxxxx xxxxxxx x (317xxxxxxx) Date: Tue, 16 Apr 2007 14:35:21 -24000 -2400? Wow! it's like a whole new greenwich mean time. Ooh, and Tuesday is the 17th, not the 16th. Andrew

resolved, thanks.

One of my users has accidently blacklisted a business contact he speaks with very frequently. I have searched the wiki, and the forums for mentions of the blacklisting feature and have found no way to remove a user from the blacklist. I therefore am asking how one might go about doing this. Thanks in advance.

Please add a forum section for SNMP.

I am looking for documentation describing the soap transaction between the PBXnSIP SOAP client, and a SOAP server to move CDR's. I already have the CDR formatting, I just need to know what the transaction looks like, and if there's already a soap plugin for Apache to receive these CDR's.