Tom Waterman

I have the problem on my SNom 820 with firmware 8.2.25 17587 and my snom 320 with firmware SIP 8.2.25 17563. I am using version 4.0.1.3452 (Win32) of PBXnSIP

Hi, I already have a ticket open on this issue but it has not been updated. My ticket number is ETO-219733. In version 4 I can no longer browse my phonebook on the phone. I get the first 32 and then when I go to search I get a bad server response code. Any word on this? I have 2 sites to upgrade but this functionality is key. Thanks for the help! Tom

I don't think it is a gain issue at all. The trunk between pbxnsip and Exchange 2007 works for everyone but this one extension. When people go to leave him a voicemail it cuts out after about 9 seconds. The only thing I can find on Microsoft's site talks about the gain being to low and Exchange diconnecting the call because it thinks the call is done. Oh well. I'll do some more research but this is not a PBXnSIP issue. Tom

Hello, I finally have UM working but I have one user whose voicemails get cut off after 9 seconds, but only his extension. Is there a way to adjust the gain from the pbx to exchange. I can't think of any other reason why this would happen. Thank you.

Ok I have this porttion fixed but now I have run into something else. I can only record 1 custom greeting. Even in version 3 I could only record one greeting. is this a bug or am i missing something. Everthing I read says I can have 5 or 9 greetings and I can't get 2. Thanks for the help. Tom

Well I finnaly fixed the issue. This client is still on the version 3. I found that if you have exchange 2007 and you install SP1 which this customer did, it more often than not will break UM. I just delete the entire config from exchange and started over. Within 45 minutes it was working. Thanks Microsoft. Tom

We are currently using version 4.0.1.3446 and I have noticed a funny (not really) thing happening. None of my users can change their voice mail greetings. When the log into the system from the phone they can record a new greeting and set the new greeting but it still plays the old greeting. I have even gone into the extension on the domain level and I can see the new greeting I recorded but it still plays the old one. I have even deleted the greeting out and it still plays the old one. How is this possible. It will only be a matter of hours before I have some angry folks. Please help. Ok the angry folks are here. I was reading the new version 4 manual and it states on page 147 that I can have up to 9 mailbox greetings. Even if I can get one to save it never lets me record more than one. Tom

Ok I managed to work through the first error but I am stumped on thei one. When I send the call to exchange it states: Event Type: Warning Event Source: MSExchange Unified Messaging Event Category: UMCore Event ID: 1109 Date: 4/15/2010 Time: 2:10:11 PM User: N/A Computer: JASCOMX Description: The Unified Messaging server has received an inbound call that has an invalid extension "3300" for UM dial plan "PBXnSIP". The call ID is "e8e8c8e9@pbx". For more information, see Help and Support Center at http://go.microsoft.com/fwlink/events.asp. Now I know it is getting the right extension number because 3300 is the one I want. In my UM Dial plan I have 3300 listed under the Associated Subscriber access. Also the Exchange server is sending bac a 302 Moved Temporarily error. Can anyone tell me what I am doing wrong? Thank you for your help. Tom

Hello all. I am going to install a certificate on the PBX so it will communicate with Exchange 2007. My question is do I HAVE to install it (the certificate) on all of the sip phones? and what about devices(extensions) that don't support certificates? Thank you for the help. Tom

Hello all, I am trying to get UM running on an exchange 2007 box. We have been working on this for almost a year of and on. I followed the setup guide and my voicemail calls go to the exchange server and then I see this error message in the event viewer: The Unified Messaging server rejected an incoming call with the ID "01e3c6f6@pbx". Reason: "Cannot find a valid UM IPGateway for 10.xxx.xxx.xxx. A UM IPGateway must exist for 10.xxx.xxx.xxx and must be linked to the UM Server via a UM DialPlan/UM HuntGroup." Has anyone ever seen this? In the event viewer I also see: The IP gateway or IP-PBX "PBX@DOMAIN.COM" did not respond to a SIP OPTIONS request from the Unified Messaging server. The error code that was returned is "0" and the error text is ":The certificate chain was issued by an authority that is not trusted". The exchange server has a certificate on it. I backed the certificate up and imported it on the pbx which are on the same domain. But how do I get it in the pbx software? I have it backed up as a .pfx file. I am not sure if the 2 itrems are related but I am dying to get this to work. I would greatly appreciate any help. Tom

Folks, I have been working on this as well. I have the logo in the HTML folder and if I check the phone the logo link is correct. However I believe the problem lies with the username/password. If I try to http to the file I am using such as htt://pbx/logo.bmp which is located in my html folder it will fail. However if I open Internet Explorer, log into the PBX in one session and then try to browse to my logo it works. any thoughts? Tom

I was unsure where to post the audio quality issuse so I chose here. Please move it if needed. We are currently having a great deal of audio quality issues since out change over to call Centric. One of the issues right now is that the first part of the word that someone speaks is cut off. I have checked and silence surpression is not on, on any of the phones. Could this be a codec issue? I have noticed that the PBX is set to use 1 codec and the phones (snome 320 & 820) are using a different one. I have QoS setup on the network as well. Any thoughts would be greatly appreciated. Tom

I am testing my snom 820 with the new version 4. I am using firmware 8.2.25. My first group of numbers loads fine. However when I push the directory button and try typing a seach I get a bad server response error. I did a wireshark capture on the server and saw no errors. Any ideas? Tom

It sends it out via email. I do not know how to use SOAP(other than to wash with). I actually went to see if I could set up SOAP real quick and I do not have any access to those settings they don't even show up in the PBX Web GUI. Am I missing something? I greatly appreciate the help.

I just had the pbx send me a cdr and I only have the follow items Time Dir From To Remote Local There is nothing about the trunk it used. I am trying to make sure that certain calls go out the Audiocodes gateway and the rest go to call centric. Thank you so much for the help. Tom

Hello all, I just upgraded to version 4 and I love the look of it. I am doing some troubleshooting concering our dial plan. When someone dials a spewcific number I want it to go out the POTS lines we still have. Other I want to co out our Callcentric lines. In the old version 3 under the current calls window I could see the connected call and the trunk it was going out. In this version the trunk is not there in the active calls window. Any way to bring that back so I know my dial plan is wqorking correctly? Thank you!

Yeah I was looking for all log messages that the PBX will write to the log file in windows to be sent to a syslog server I have on another machine. Thanks for the help!

We are reviewing out current PBX set and looking to improve overall paerformance of the system. Our current setup is a single nic that is connected into our dmz. We have a SIP trunk to call centric for our main traffic and we have a couple of Audiocodes gateways that connect to POTS lines for backup. I have a second nic that we can allocate tot this VM I was just wondering if anyone had any design thoughts. I could also connect the outside interface directly to the internet as we have a public IP but then I loose the protection of our ASA 5510. I look forward to any comments. Tom

Hello all, I am trying to do some troubleshooting and it would help it I had an easier to read log file. Will PBXnSIP output to a syslog server? Thanks, Tom

Hello all, we have transfered from a free conference calling solution and are using PBXnSIP and things are going very well. I have a question from one of our sales folks. Can they setup a reoccuring meeting and if so how is this done? Thank you so much for your time. Tom

Could you gove me an example of what an edited file would look like? Thank you so much!

Hello all. I want to edit the vcard info sent out with the conference setup. I want to add our corp telephone number and provide a little more instructions. I can see that this can be done in the system admin tab. Has anyone ever done this? I could use a little guidance. Thank you! Tom

Can anone tell me what to do here? I have written the custom xml and to test it I called it snom320.xml and dropped it in the html folder. All phones pick it up fine so I know it works. But I want only a single phone to use it so what do I do? Please anyone? Tom

I would love to see some explaination. I have almost the same thing. I have snom phones and a few users want custom settings. I can write my xml file and put it in the HTML directory, for example snom_320.xml but thiese settings go to all Snom 320. How can I make it go to only one phone? Can I use the mac address in the file name and if so what does the sample file name look like? Please help Tom

I removed localhost all together, added in my PBX. Which worked. Then the pbx shot a big error about having localhost as a domain and it created a new domain called localhost. I deleted it and added it as a second alias to my domain and it works.