Kristan

That's exactly what I was looking for, thanks!

We have a pair of customers who run near call center type operations who want to get some more details statistics from pbxnsip. At the very minimum, they need to be able to identify the maximum number of concurrent calls in a day, abandoned calls in a day (ie not answered) and number of calls lasting under 5 seconds in a day. I understand stuff like this will be available in 2.2, but we need this before they go live at the middle of next month. I'm thinking of writing a small C# console app to parse the CDR files in the directory and just send an email containing the details. Can someone give me the details of the structure of the CDR file, or a better way of doing what I need? Thanks

Yeah it works, we tried it between two a polycom 550 and 650.. You have to change the codec preference in the pbxnsip xml file, I think the details were posted in the RC1 thread. To be honest we couldn't tell the difference between G.711 and G.722

The only way I've had this working reliably is to do format the file system to reset the whole phone, then change the server at boot to TFTP, and have it reload everything. Not ideal in the slightest, but it works. Just hope you don't need to change the list of extensions being watched too often! I'm still confused as to how the phone actually knows what extensions to watch, as when I was testing I couldn't see anything in the generated config files regarding the extensions. You can try what gotvoip suggests here, I'm afraid I've not had a chance to yet : http://forum.pbxnsip.com/index.php?showtopic=381

Thanks Bill, we had something along the same lines at another client with Snom 360's, thought in that instance it was the phones that were having a fit as they were receiving lots of notifies all at once as a hunt group rang. Seems we have all the difficult clients In this case it's definitely the server, as I can see the RTP from the phone to the PBX is still ok. I have a suspicion it may be OS/hardware/something other software related, but I'd like to know what it would take to cause the PBX to stop forwarding RTP. I wonder is it possible to have the PBXnSIP service run at "above normal" or "high" priority in case it's something on the server hogging the CPU or blocking.

Hi all, I have a strange problem with an installation of PBXnSIP. The customer is complaining of audio cutting out on some calls, and sure enough they are right. I started a wireshark capture and luckily enough managed to get a conversation with a break up almost straight away. The setup is: ISDN-> Parlay ISDN Gateway --- SIP/RTP ---> PBXnSIP --- SIP/RTP----> phones From the streams, the breakup appears to be due to the PBX not sending RTP traffic. At the time of the breakup, between the parlay and the PBX I can see a steady stream of RTP going TO the PBX, but nothing FROM it. At the same time, between the phone and the PBX, I can see traffic FROM the phone, but nothing TO it. It's like the PBX just decides to stop sending RTP, then carries on again. The duration of the break is about 8 seconds. There was also another call in progress, same as above, but terminating at our SIP trunk instead of the ISDN gateway, which also sufference idential loss for exactly the same period of time. It doesn't appear to be network related, as I can see both sides of each stream arriving at the PBX, they just don't get passed through, it's like the PBX just stops for 8 seconds, then starts again. Is there anything that would cause this?

I'm interested in this also. We're about to do a large (50+) handset deployment with Polycoms and PBXnSIP, and there is a requirement to get the attendant panels working. Using the "watch calls/presence" from the extension menu will populate BLF on 601/650's and attendant panels, but beyond that I can't get anything to work. As far as I can tell from the Polycom manuals, you should be able to use the <attendant/> tag in the phone file to specify which button does what, but so far all my attempts to do this have failed.

Tried it with a stopped service this morning, upgrading from standard 2.0.3.1715, still the same thing. Also it didn't put any provisioning files in the HTML folder, so I'm not sure if it actually did much. And in this instance, a repair install with the service stopped didn't fix it either. In the end I copied the PBX folder to a new location, uninstalled PBXnSIP, re-installed with 2.1 and copied the config files back over again - this seemed to work, but I'm concerned there may be settings/features that are in the new version that were overwritten by the old files. Do you know if there are any like this or not? I've got a fair few PBX's to update to 2.1 and could do with knowing the best way to do it. UPDATE: It seems the version on the website doesn't contain any of the HTML files for the autoprovisioning of phones.. Was this only in the RC installer and didn't make it to the final one?? Thanks

Hi, Can anyone explain how the BLF functions work on the 601's? I'm having great difficulty getting these to work reliably. I've only managed to get it to work after resetting device fully. A normal reboot doesn't seem to work. As far as I can see, the config files don't specify the extensions to watch when they're sent to the phone, so I don't understand how the phone gets the list? Help?

Can I request that the ability to set or unset the service flag from the web admin be added? Would be handy for some of our customers. EDIT: Also, can we have them log in the call log when it's set or unset (or an option to get them to log to the call log). Thanks, Kristan

Should the "upgrade" option with the installer work? Tried it on our office PBX (v2111) and it seemed to work, but after checking the version it was still 2111. Renamed the exe, did a repair install and it put the 2115 exe in...

Excellent thanks!

Just an update to this, it doesn't seem to have broken on the latest and greatest version (yet), however I have noticed you can replicate the noise. If you're the first call to get put on hold after a service restart, you get a second or so of the nasty screeching, then the music kicks in properly. Same for agent groups etc., anything that uses the MOH. I guess the encoder isn't starting until someone actually needs the moh and it takes a second or so to do whatever it does?

Ok, I've got the linksys auto provisioning working, method is as below: 1. Edit spa_1st.txt to remove the extra http:// 2. Edit spa_phone.txt and replace the all the {loop-count} tags with {loop-count 1} tags. In theory, changing the {loop-start} tags to {loop-start 1} should do it, but that doesn't seem to work. 3. Factory reset your phone - it will get the tftp server from your dhcp scope (assuming you've set it) 4. After 10-20 seconds it will get the spa_1st file by tftp and reboot. 5. After rebooting itself, the phone seems to need an extra reboot to get it to pull the config file, so reboot it (or cycle the power). 6. The phone will come back up, and after 10-20 seconds will pull the config via http, reboot and it will register with the PBX, assuming you set assigned the MAC to an extension. Basically the default file tells the linksys to provision line_0, and it doesn't have a line_0. I'm still trying to work out why it needs the extra reboot after the initial TFTP, and why after provisioning it seems like I ca't dial out! Progress at least though. Update: Now have the provisioning working pretty much as it should. Add the line <Resync_Periodic ua="na"> 5</Resync_Periodic> to the SPA_1st.txt file and the phone will pull the http file correctly after the 1st reboot. I've had to edit a couple of bits in the spa_phone.txt as there's two tags which don't seem to work out of the box : {Dialplan SPA} and {TZ DST-SPA}. I'm guessing I need to set these somewhere in the pbx? If anyone is interested, I'll post my files to the linksys part of the forum, as it includes strings for UK DST etc.

You'll need to change the config on the polycom to use TFTP though as it won't do it by default - it's under admin->network->server (or something like that...)

Some more provisioning testing : the spa_1st.txt file has an extra http:// in - the {HTTP-URL} seems to include http://. Also, once that's changed and the phone (an SPA942) is on the latest firmware, the phone requests the file and reboots, and I can see the pbx sends the config (with my extension etc. set), but after the reboot, the phone doesn't seem to have taken any of the settings. Any ideas?

Also, the soundpoint IP 550's support G722. It looks like it's being sent as the prefered codec in the invite, but the RTP gets sent as G711.. I've not had a propper chance to look at it yet but is there anything special that needs to be done to get it working?

Audio drivers are up to date on both PBX's. I'll try it on our RC2 demo box here, see if that does it too.

Just trying this RC now - auto provisioning of our Polycom IP 550 worked fine (once it was set to TFTP), but our snom360's aren't having it. Packet capture shows it pulling a file via TFTP, but the contents of it tell it to go to https://serverIP:443/provisioning/snom_3xx_...address>.xml. The phone then goes off to http://10.6.0.25/snom360/snom360-firmware.htm, gets a 302 and dies. Any ideas?

Hi All, We have two installations now where after an indeterminate amount of time (several days / a week) goes from being whatever is being played through Line In to horrible screeching noises. The hold music in both cases is provided by MP3's played under winamp, using a 3.5mm to 3.5mm cable from the output on the soundcard to the input. PBXnSIP is set to take music on hold from the line in. The music is still playing fine (if you plug headphones in instead of the loopback cable), and a restart of the PBXnSIP service seems to reset the music back to normal. Has anyone else seen this before and have a workaround, or can anyone else duplicate it? Thanks

Was there any progress with this? We've got a client requesting we change the voice prompts to a more "English" sounding voice, rather than have to pay talent and editing to do it, is there already a set in production? Thanks

Is that 7.2 of the snom firmware? We're running beta 7 versions on some of our phones at the moment to try to fix some other issues. I notice they are moing into line with others by going for XML settings and phonebooks. Can you shed any light on the auto-provisioning side of things as to how I'd be able to achieve sending individual config files to each phone using multicast instead of tftp? Thanks

Further to this too, I thought I'd have a play with the address book auto provisioning too. This doesn't seem to work at all. I've got two address book entries in my PBX, and the following lines in my config {Adrbook-Start 0 99} tn_{Adrbook-Speed}!: {Adrbook-First-Name} {Adrbook-Last-Name} tu_{Adrbook-Speed}!: {Adrbook-Number} {Adrbook-End} Which should work, (assuming I labeled the speeddials 0-99 in the pbx) however in the generated file, there is just an empty space. The PBX is clearly parsing the file to trim them out, yet doesn't actually seem to be doing the replacements. Does this feature actually work? While we're at it, would it be possible to request a replacement variable to give the number of the phonebook entry (ie 0, 1, 2, 3) etc. for phones that do them in numeric orders (like the snoms?) Thanks!

Hi, I'm trying to get my snoms to pull the configs via multicast and HTTPS on startup. Basically I have the following: <?xml version="1.0"?> <plug-and-play> <file name="snom360.htm" encoding="ascii"> <pattern>!snom3[26]0-([0-9A-F]{12})\.htm!\1!</pattern> <vendor>snom</vendor> <model>snom360</model> <pnp-vendor>snom</pnp-vendor> <pnp-content-type>application/url</pnp-content-type> <pnp-url>https://{IP-Adr}:{HTTPS-Port}/{Model}-{MAC}.htm</pnp-url> </file> </plug-and-play> If I read the wiki right, and understand correctly, this should mean that if I have a file called snom360-xxxxxxxxxxxx.htm, the HTTPS server will give the snom this file when it requests it's startup config. The idea is that I can have a generic snom360.htm with my domain defaults in, then if I want to provision specific items on individual phones, I can do it by creating one with the mac address and the phone will pull that file. This seems to work ok doing it via tftp, but not via multicast. I have my snom360.htm file in the same folder (html) as the one with the MAC address but it always seems to take the generic file instead of the more specific one. Watching with filemon, the pbxnsip process doesn't even attempt to serve the specific file, instead it creates a "generated" folder and places it in a subfolder named after the mac address. Can anyone shed some light on what I'm doing wrong? Thanks, Kristan

Thanks for that Bill, I think we're going to have to go down that route as the address book on PBXnSIP just isn't working, just a pain as it used to work ok. Thanks, Kristan