jlumby

Well, I have been monitoring it as it increases, and the following values have increased. The 3 values I am giving you are at 62 meg right after a restart of the service, at 82 meg (2:45 of uptime) and 163 meg (18 hours of uptime) pending_packets: 102, 2615, 17306 callback_objects: 1531, 9009, 53076 timeout_objects: 1530, 9008, 53075 registrations: idx=5838 size=40, idx=5870 size=72, idx=5798 size=71

I have always set them up manually, and not used the web interface. I know they work fine manually. I have 3 that are currently running that way

I am running 3.1.1.3094 for cisco multi domain support and I seem to be leaking memory. The following is what my pbx is currently displaying: Uptime: 0 03:10:50 (87MB/2046MB 18% 27316352-0) WAV cache: 1 If you notice I am currently using 87MB of Ram. It was up to 1989MB, and I was getting complaints of poor call quality. I stopped/started the service, and it dropped to 62MB, and now that it has been up for just over 3 hours, and it has increased 25 MB I am running it on a dedicated windows 2003 server with all of the latest microsoft patches. The CPU utilization stays very low (less than 5%) I have logged into other customer installs that I have done (all equal to or less than 3.0.1.3023) and they are all sitting in the 60 meg range.

The trick is to set the proxy on each line to USECALLMANAGER and then define the ip or FQND of the pbxnsip server in the callmanager section of the xml config file

OK, in my case I am ok just sending 911 to my ITSP, however, since I am going to have multiple branch offices at different physical locations, they want to identify my location based on my outbound ANI. The problem is I only want to send that specific ANI when dialing 911, I want to send a different ANI for all other calls. Is there a way to do this?

Are you using the web interface, or uploading the config files yourself? I have found that if you edit the config files yourself, and then upload them to the device, they get invalid characters in them (something with the ssh file transfer), so I have had to upload them, and then edit them in VI once they are already on the appliance.

Since then I have found documentation that patton smartnodes have the option to operate in this manner, however I have not shelled out the $2500.00 to test it yet.

I was reading the brief documentation on the Emergency Destination Number field that the domain, as well as each extension has. The documentation has me a little more confused than before I read it. To the best of my understanding, the online documentation (http://wiki.pbxnsip.com/index.php/Release_Notes_3.0.0#E911_Enhancements) said 2 things whick kind of contradict eachother. The first is that "Now when someone dials a number listed as E911 number, the PBX will always send that call to a trunk." and the second is that "so that E911 essentially works like a speed dial" So here is my understanding: The system sees the extension dial a number that is listed as an E911 number (where is the list that the pbx is checking against?) and therefore when it discovers this match it speed dials the value that is currently in the Emergency Destination Number field. The problem is this. The documentation says that the pbx will always send the call to a trunk. On the PSTN, what number would you be speed dialing other than 911? The second issue is if I setup ITSP account to support multiple E911 locations, they want me to identify the location based on the caller ID. If the system's Emergency Destination Number field worked like a speed dial, then I could point it to an extension that had an ANI value corrisponding with my E911 location, and then set that extension to forward all calls to 911. The problem with this is the documentation states "Now when someone dials a number listed as E911 number, the PBX will always send that call to a trunk." since it would be going through an extension to get the proper ANI, this would not qualify as a trunk destination for the Emergency Destination Number field. So with all of this said, what is the best way to do multiple E911 locations on 1 system when the ITSP identifies the location based on CallerID?

I am very fond of the the Cisco ASA5505 firewall. It has PoE ports plus fully supports QoS. Not to mention it's fully SIP aware, so you never need to wory about private IP addresses getting embedded, and passed out across the public network. Since it also inspects the packets, you do not need to open up the RDP ports in the firewall, it will do it dynamically. The DHCP server supports option 66 and 150 for auto provisioning phones. I cannot think of any shortcomings of the unit.

It is interesting because it was through an ITSP, and it happened in the middle of a call. I would have expected it at the beginning. The other interesting thing is other people were on the phone, at the same time, and they did not have any call quality issues.

I had a end user that originally thought that they were dropping calls, however with some further investigation, I came to find out that if they wait long enough, the call would come back to life. During the dead period, I saw that the call state was listed as idle. What might cause this? It is so random, and infrequent, it might be very hard to duplicate, and/or catch in the act. I recently upgraded to 3.1.3044 and am not sure if this is related.

I have found in 3.1.0.3040 if you disable a mailbox, it still answers, and you can still leave, and retreive a message.

I am having the exact same problem. The folks at PBXnSIP are aware, and are working on it.

I would suggest that in the individual extension configuration web pages that a next account, and previous account button be added. This feature would dramatically speed up mass changes. For example right now if you want to change a feature for a bunch of accounts, there is a lot of lost time clicking and navigating. If you wanted to change the dialog permissions on a bunch of accounts, right now you need to save, and hit list, find the next account, click on the edit icon, and click on the registration tab, and then enter your value. If there was a previous/next button, you could hit the save button, then hit the next button, and you would be to the exact same location on the next sequential account number, and ready to enter your value.

I am seeing the same thing with people going over 3 stages when using the mailbox escape account

I am trying to setup an Aastra 57i in a typical squared line button configuration. For those of you who are not familiar with this old PBX term, it would be so that the line buttons on the phone corrispond with CO lines on the system. This would enable someone to take a call, put it on hold, and by remembering what line it is on, go to any other Aastra phone, and press the line button to pick it back up. I have done this successfully with Snom, is the Aastra 57i capable? If so how?

My understanding of the * in the Explicitly specify park orbit preference field is that the system would prompt you to enter the park orbit that you would like to put the call in. It does that successfully, however there is some sort of permissions issue where if you type in an orbit number that is anything other than your own extension number, it fails to park. If you type in your own, then it works perfectly. My understanding was by adding the * in the Explicitly specify park orbit preference field was to alleviate needing a button for each park orbit when you have a large install where you need to control where each call gets parked so that the posibility of someone retreiving a call that is not for them is very minimal. The operator can simply hit the park button. The system prompts for the zone, and then they would type in any zone number (preferably one that matches the extension of whom the call is parked for) Then when they page, the person could hit the single retreive button (on any phone), and type in the park zone number, and have their call, eliminating the chance that someone would pickup the wrong call, from not having full control over the orbit number when parking/retrieving. This also removes the need for multiple park/retreive buttons, or memorizing star codes.

I am still having problems with version 3.0.1.3023. it does not allow me to park on any orbit except the one that matches my extension number when using the * in the Explicitly specify park orbit preference field. Are there permissions I am missing, or a newer version available?

I left all other fields at their default value, the only thing I filled in was the "Bind to specific IP address (IPv4):192.168.3.64" I noticed after the reboot, it still responded to it's other ip addresses both for SIP, as well as HTTP I checked the pbx.xml, and I saw <port_bind4>192.168.3.64</port_bind4>

I am attempting to bind the PBXnSIP system to a specific IP so the other one is available to run IIS (as well as I do not want it to be able to respond to sip requests on the other IPs). I filled in the IPv4 address in the settings/ports/Bind to specific IP address (IPv4): field, however after a full reboot, the web, and sip ports respond to other IP addresses assigned to the system. Are there additional steps I need to take? I am running windows version 3.0.1.3023

My browser says it is complaining about java script://void(0)

I have been seeing the same thing.

I have seen many issues with Snom 360s and using domain names. They tend to lock up, or do weird things about every other day if you use a name, however if you use an IP, they will run for an average of a month or 2 without the same weird issues.

I have just ventured into setting up multiple domains in the hopes of hosting if all of the testing goes well. I am having some issues with the second domain. For example sake, lets say that I setup the DNS records as follows. 75.136.163.71 = voip.company1.com 75.136.163.72 = voip.company2.com Under the PBXnSIP domain setup, I created a domain with the FQDN as the domain name, and I set the IP address as an alias. I am having the following issues when testing. When I put a call on hold, the caller does not hear hold music, and they cannot take them off of hold. Upon inspection of the packets, I see the IP of company1 within the payload of the packets sent from the PBxSIP machine (not the from address) When TFTPing down phone config files, The request from the phone goes to the IP of company 2, however the IP of company 1 responds, and some firewalls block this since the data is coming from a different IP than the phone made the request to. I know it is possible to only bind the PBX to specific IPs, however is it possible to do it at a domain level? I realize it is possible to point both DNS names at the same public IPs, however there are other (non voice) aspects of my hosting that sharing an IP will cause issues with

In a future version would it be possible to add 2 more values to the daily call email. In addition to the total time, It would be handy to see total outbound, and total inbound time. This would make it simple if you have an ITSP that charges by the outbound minute, or vice versa.