sudo

I may have spoken too soon. There is no 'Key Event' for button type. See attached picture for the available options.

This Worked! A Thousand appreciations good Sir

Nice, Thx

I have several Snom300's that all come with the soft buttons pre programmed. The last button is a Mute button. Now back to snomONE, im on the button profile and theres no mute option. How can I modify these snom300's settings to move the mute button around? I thought this would be appropriate for the snomone forum and not the snom forum as this has to do with the configuration of the buttons through the snomone webui. Thanks in advance. Sudo

This is great thanks for the reply. To confirm - Change this: <ua vendor="Polycom">FileTransport PolycomSoundPointIP-.*</ua> To this: <ua vendor="Polycom">FileTransport Polycom-.*</ua> or would it be this: <ua vendor="Polycom">FileTransport Polycom.*</ua> Thanks! Sudo

I have an interesting problem regarding Polycom SoundStation's and plug and play. I should preface this by stating that we have the full license that supports pnp of non snom devices. So - The beginning of the pnp.xml is as follows: <?xml version="1.0"?> <plug-and-play> <http> <ua vendor="Polycom">Polycom-FileManager/.*</ua> <ua vendor="Polycom">FileTransport PolycomSoundPointIP-.*</ua> <ua vendor="snom">snom-m3-SIP/.*</ua> </http> The problem happens when trying to connect a PolycomSoundStation as it does not match the "PolycomSoundPointIP" and does not work. In order for me to get a sound station to connect I have to modify this line from: <ua vendor="Polycom">FileTransport PolycomSoundPointIP-.*</ua> to: <ua vendor="Polycom">FileTransport PolycomSoundStationIP-.*</ua> How can I modify this file to accept both PolycomSoundPointIP and PolycomSoundStationIP? I have asked our in house developers to look into this but they said that they needed to know how the system calls on these files in order to know exactly how to modify it. I have to imagine that this has came up in the past. Can anyone help me out on the formatting of the .xml? Many appreciations. Sudo

Thanks for that great idea! It worked. To reiterate what I did: Created IVR Node Made the new IVR node the destination for the AA Created the dtmf match list to reflect the input I want and the destination Works great, There is no invalid string cause the system is looking to match the input. Only problem was dead air (No recording after initial message plays) Placed it to time out every 10 seconds and trigger itself so the message will 'cycle'. Thanks again. Sudo

So I finally just deleted the domain and started over. I think I may know what the issue was. Initially, I was coping and pasting the numbers from an excel spread sheet and I think this is what caused the issue. This second time around I was inputing the DID's (not c/p them over) and I noticed when I hit save, the system would add the dashes between the DID (XXX-XXX-XXXX). Its now working. Cant say for sure that this was the issue, but this is not the first time c/p has led me astray. Thx

The invite it coming from mu voip switch box to the pbx. The trunk is on the right domain and its not global. If I dial the DID from a device registered on the domain, it rings through and I get the voice mailbox.

Ive got a similar issue. The trunk is up, I can call out. The DID is active. But when I call in the call gets rejected with a 404 not found. This is something on the domain. I have tried the same trunk creds on the same box, but different domain and it works. I can call in and out. If I move the same trunk to the defective domain it breaks. All the trunk settings are the same. I have no idea what this issue is and suspect it may be a bug. Anyone have any ideas?

I should also mention that on the snomONE box, I got a snom360 to record using the record hard button. So initiating the call record from an end device is possible.

Im having a strange issue. Im using a Polycom IP330. Ive added the star code *93 and 94 to start/stop the recording. In my tests im using the polycom, a xlite soft phone, and my sprint cell. The soft phone is registered as an extension on the same domain as the polycom. When I call my cell from the polycom, or call the polycom from my cell, it does not record. Theres nothing in the .../recordings dir nor anything updated in the "Voicemail" page of the WebUI. When I call the xlite softphone and initiate the star code (I always initiate the record with a star code from the polycom) the DTMF tone plays through the soft phone as if the button were held down. But the weird thing is the recording actually starts and gets sent out to the Voicemail tab of the webui. When I call into the softphone directly from the sprint cell, the star code does not work. This is all on a snomONE box. When I try this from an old pbxnsip box, it works. Ill compare configs and get back if I find the issue. Does anyone have any suggestions in the mean time?

How? Ive looked over the article you posted and others, but im not sure what I need to do.

Quick question regarding the star code to start a conference call from an extension. *53 by default. Does it require that a conference room be setup and then the star code pushes the call out the conference room account, or does it bridge the calls together on the sip device/extension itself? There is no info here http://wiki.snomone.com/index.php?title=Miscellaneous_Codes Thanks in advance Sudo

But will it be a global change across all domains on the pbx?

I have a customer that is requesting to upload their own, custom error messages. By error message I mean this - If you dial an invalid extension number, the system will play an 'error' message that states it was an invalid extension. In this case they have a 24 hour support queue and a 9-5 support queue, each with non consecutive 5 digit 'account' numbers assigned. The idea is to give out a 'unique' passcode to the 24/7 support customers. That unique password is the non consecutive 5 digit 'account' number. this way they dial the non consecutive 5 digit 'account' number, which rings a queue that has no service flags and will forward out to the on call cell. If the customer mis-dials, they dont want the system default error messages, but a custom "Please dial the passcode again" message. Is this possible on a domain basis? It cant be global for obvious reasons. Thanks in advance! Sudo

I see that the system requirements listed here are pretty vague http://wiki.snomone.com/index.php?title=Requirements I have a ever growing pbx with about 15 domains on it now and im worried that I might be reaching my system resource limit. What is the per user breakdown of the required cpu, hdd space, ram usage, and bandwidth? I just want to make sure I scale this system up appropriately.

Quick easy question that I cant seem to find the answer to: How many calls can the conference room handle at one time? We got it up to 7 today, but was wondering if there was a limit in SnomONE. I know the PRI will cap it off at its limit.

I clicked on the link in snomONE for more info on an agent groups 'agents' tab but theres no info. http://wiki.snomone.com/index.php?title=Monitoring_Agents_in_Agent_Group Can we get someone to update this (and all) snom wiki pages. Are we as customers expected to just figure this out or can we get some help?

Any tips for Linux machines? Im having some issues...

"Should be there"? Any suggestion on modifying the xml to reflect this info?

The customer wants this info in real time to motivate the teams to answer the most calls live.

On the phone would be preferred, but im open to suggestions.

I need a way to pull a report and display in real time, the percent of total calls, per agent group, that do not go to voice mail in a 24 hour period. In other words the percent of live calls that are answered. Looking at the agent reporting, it does not provide this metric. Anyone got any ideas on how I can go about accomplishing this? A thousand appreciations Sudo

Regarding "ringing the longest idle first" setting in the Agent Group settings - I need to find a way to display in real time, either on the phone on webui, the position the agent is in , in the queue (who is the longest idle). For example, there are 10 agents in a queue. They want to know who is going to be the next person rang for the next incoming call. Im not seeing anything on the webui that will display this info. Thanks in advance.