Tom Waterman

OK I have attached the pcap of the communication between the phone and the server during the phone boot. I have in the PnP firware settings only snom370.bin and it is placed in the html directory in the pbx folder. I have even deleted the generated file for this extention in hopes it would rebuild it, and it does but the firmware is still not updated. Thank you for the help. Tom Firmware_update.zip

In the tftp folder I put a file called snom370.bin In the PNP settings under the firmare settings for this phone i put the same file name of snom370.bin However the phone fail to pick it up. I have no other tftp service running on this machine and the phone do pull down thier configs properly. Any ideas? Tom

I just upgraded a customer this morning from 3.2 to the newest release of 4.2 on the website. The phones have firmware 7.3.23. I can make inbound and outbound calls however I can't make extension to extension calls. PLease help! tom

Hello all. As you can guess I have a network with no INTERNET access. So I have the new firmware for the snom phone on my PBX. I know I could start IIS on this windows box and point the pbx to that to get the updates. But I was hoping that I could just put it in the HTML or tftp directory. Has anyone done this and what would the firmware path look like in the PnP settings. Thanks for the help! Tom

Hello all. I have a user who has 2 different greetings for her mailbox. They are running an old 3.2 version. Is there a way to delete all of her greetings and just let her start over? I know you can in version 4.x but I don't see and easy way to do it in 3.2. She is at a remote site but I do have access to the servers. Thanks Tom

Ye4ah I looked at those with not much success. I have to PBX packup from the Windows box and I moved it to /usr/local/pbx on the CentOS box. I have removed the old windows controler and placed the Linux controller in its place. So now all I need to do is get the service to start which is where I am falling short. Tom

Hello all. We have decided to move our PBX from a Vitualized Windows server to a dedicated Linux box running Centos5. I have the CentOs install completed and I have the pbx working directory from Windows. Can some one give me quick directions how to make this work. I know I need to copy the directory to Linux and download the new controller for Linux, which I have done. I just need a little help with were to put the directory and any permission modifications that will need to be done. I am familar with Linux, I am just not a pro. :-) Thanks in advance. Tom

I was able to get the codec locked at g711u and I am still having packet loss. Any other ideas? I am taking a beating here.

Matt, I thought about that and checked it. Here is my config. I believe this locks it but obviosly it does not. Tom

On thing I have noticed is that on the trunk between PBXnSIP and Callcentric the are using G729A and we want G711u because the audio quality is better and we have the bandwidth. We have a dedicated 3 meg pipe for the pbx. Could this be part of my problem? And how do I get the trunk between Callcentric and the PBX to stay at G711u? Let's hope this is a step in the right direction. Tom

Matt, here are some of the issues I have found. We do have issues with ext to ext calls on the same lan. When the conference calls happen they are not always on the lan. Some dial in from outside sources. This is a virtualized server with plenty of juice to run pbxnsip. The switches are new linksys by cisco switches. I know Cisco switches would have been better but we needed 8 and I could not get the company to swallow that bill. I have looked at the CPU usage and it is very very low. I do have some low MOS scores so I know the PBX knows the quality is bad. So I am now running wireshark captures on both interfaces and sending them to a shared drive. I spend about 6 hours a day in wireshark.I have made one change to the pbx since Monday and that has been to increase the ram from 2 gigs to 4 gigs. Any thoughts? Thank you! Tom

I have used both of these troubleshooting guides with no luck. The audio dropping happens everyday in all configurations. It happens on an extension to extension call, on a Callcentric to extension call. I have to figure out what is going on before people get more upset. I am thinking about moving the PBX from Windows to Linux as I have exhausted other options. I do have a support ticket open but have not heard anything back for 2 days. :-( Tom

I have noticed a few things with this. If you watch the recording folder during the call it will show ther file as ConfXXXX.wav after the call is done it changes the file name to msgxxxxx.wav. Why is this? and is there a max size to the recording? My boss was on a call and had recorded it and now can't find it. I am looking in the recordingts folder and I do not see it. It was approx 2 hours long. We are using them same version.

Hello All. I am looking for some help with an on going audio issue that we are having here at out office. We are currently running PBXnSIP 4.2.0.3958 (Win32. We have a dedicated server that has 2 network connections. One is connected directly to the Internet via a 3meg dedicated fiber pipe. The second connection connects to a Ethernet switch. All of my phones plug into the same switch. The entire PBX is on its own network. Yet still I get audio loss. I am not talking about 1 way audio. Let me give an example. Last week I have 10 people in the conference room at out office who are dialed into a PBX conference extension via a single pod phone. I then have 4 folks from our Denver office dial into the same conference room. They are calling into the Auto Attendant like any normal person would do. During the meeting while the new Director of Operations is speaking the sound cuts out or becomes distorted. It is her voice that no one can here for a few seconds and then the sound comes back. This is happening more and more often. It happens in all types of situations this is just a single example. I had previously gathered some wireshark captures and I could see the audio was breaking up leaving the PBX to the phones but I am sure this is not the only case. Can someone please give me some assistance. I have already read all the stuff in the old wiki and most of it does not apply to our setup. One last thing. Our provider is Callcentric. Thank you for your time. tom

SOrry for the late reply. I was off the 11th and 12th last week. We are running version 4.2.0.3958 (Win32).

I found in the Event viewer under the application section the following stop error: Log Name: Application Source: Application Error Date: 11/8/2010 11:24:09 AM Event ID: 1000 Task Category: (100) Level: Error Keywords: Classic User: N/A Computer: SRPBX01.mdgn.microdatagis.com Description: Faulting application pbxctrl.exe, version 0.0.0.0, time stamp 0x4c992b3b, faulting module pbxctrl.exe, version 0.0.0.0, time stamp 0x4c992b3b, exception code 0xc0000005, fault offset 0x00179ef2, process id 0x204, application start time 0x01cb6b94d88facdc. Event Xml: <Event xmlns="http://schemas.microsoft.com/win/2004/08/events/event"> <System> <Provider Name="Application Error" /> <EventID Qualifiers="0">1000</EventID> <Level>2</Level> <Task>100</Task> <Keywords>0x80000000000000</Keywords> <TimeCreated SystemTime="2010-11-08T16:24:09.000Z" /> <EventRecordID>3502</EventRecordID> <Channel>Application</Channel> <Computer>SRPBX01.mdgn.microdatagis.com</Computer> <Security /> </System> <EventData> <Data>pbxctrl.exe</Data> <Data>0.0.0.0</Data> <Data>4c992b3b</Data> <Data>pbxctrl.exe</Data> <Data>0.0.0.0</Data> <Data>4c992b3b</Data> <Data>c0000005</Data> <Data>00179ef2</Data> <Data>204</Data> <Data>01cb6b94d88facdc</Data> </EventData> </Event> Luckily this has only happened once. :-)Any thoughts?

I have tried all of those recommendations already. :-) One thing that I am currently looking into in the speaker phone on the Snome phone. It appears that all reports of clipping audio appear to be on calls in which the speaker phone is being used. It is not an issue of one way audio per se because it just drops for a second or 2 in the middle of a 60 minute conversation or words will break up for 4-5 seconds and then it is fine. Any idea of changes that could be made on the speaker phone? Tom

Hello all this is a relatively new problem we have experienced. I am getting small portions of audio clipping or a toal loss of audio (2 to 3 secopnds) in one way. I have done days of research on this issue. We are running G.711U I have a dedicated 3 meg pipe on the outside for the pbx. The inside interface on the pbx is on the dmz and all of the phones connecte to the dmz switch. So from the PBX to the phones it is 100meg switched network on its own vlan. I have wireshark running on both interfaces and I can see in the audio stream that the issue is occuring from the pbx to my phone! I have a hard time understanding why but it is. The streams in and out of the pbx to the provider(callcentric) appear to be fine. Does anyone have any idea where I can start to chase this? I have noticed that it does not seem to matter if the pbx is under heavy load or not it seems to be a random issue. Any advice would be helpful. Tom

This is on a Windows box so there is no syslog server. I do have email notification on but it just told me the service was restrarted but not who issued the request. Maybe in a future version? Tom

I run version 4 on a Hyper V virtual machine. It has been there for a year or so, starting with version 3.2 No problems at all.

Hello all, has anyone ever had the PBX service restart on its own? Mine just did it this morning at 11:30. How can I tell who requested the restart? Thank you. tom

For the cisco ASA you need to make it sip aware. Just having a por tope is not going to help fro the out side. Here is a link to the 3 lines you need to add to the config. First set a global policy map then tell it to inspec sip. Yopu can trouble shoot more after. http://www.cisco.com/en/US/products/ps6120....shtml#configs1 Happy firewalling. Tom Waterman, CCNA

Hello all. I have a client that has exchange 2007 and PBXnSIP 3.4.0.3201 (Win32). Before we went to exchange for voicemail you could dial the direct dial voicemail number of 8 + Extension # and it went to voicemail I removed that for exchange and put 99$u in the external voicemail setting. However if I have someone try to send me directly to exchange voicemail I get a wierd error message telling me that I can not use that extension to make the call. ANy ideas how to achieve direct dial voicemail in Exchange? Thank you! Tom

Whewre do you endit the length of time? I looked in the above mentioned section and I don't see where I can edit that. Should it not just be one of the xml pages in the PBX directory that I edit? Tom

Ok are SP is callcentric. Do you happen to know those? Thanks guys! Tom