David McGowan Posted October 4, 2007 Report Share Posted October 4, 2007 The PBX takes the freedom to resort the list of proposed codecs according to its internal preferences. As long as the user agent promotes all available codecs it is fine. The 2114 build also has a global codec preference in the system admin/settings/ports section. Did you see that? I see that, nice... So the above sip.cfg is fine because it will sort everything out for me? I see the default is: 0 8 18 2 3 Which means: "0" (G.711 u-law), "8" (G.711 a-law), "18" (G.729), "2" (G.726) and "3" (GSM) If G711 is working then that is the best quality because it uses no compression right? thanks for all the help! david Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 5, 2007 Author Report Share Posted October 5, 2007 Yes. IMHO in a PBX-environment (business) it does not make sense to compress voice too much at the cost of bad quality. Hey, it is your customer calling and you don't want to leave a bad impression. G.711 has the big advantage that there is no transcoding from digital PSTN. Transcoding means reduction of quality, even if you are using higher quality codecs and even if the transcoding happens in the PSTN gateway. We tried it - Transcoding from G.722 to G.711 is not as good as pure G.711. Those low-rate codecs are okay if you are somewhere in the woods and the primary goal is to get connected (e.g. dial-up modem, satellite, ...). Then your customer will understand that you sound like in a tin can. Quote Link to comment Share on other sites More sharing options...
phsean Posted October 5, 2007 Report Share Posted October 5, 2007 Regarding the Exchange UM Caller ID issue posted earlier: I removed the polycom files from the html directory, restarted PBXnSIP and my phone (just to be sure) -- no dice still. So I ran a call log and stripped out my test call: Log File You can see that starting with Line 234, the From header changes from the calling extension 2503 to the recipient extension 2412 when the call is referred to the unified messaging server... why that translation takes place looks like the problem. Quickie explanation of what's happening in the log: 2503 calls 2412, 2412 rejects call, 2503 receives "2412 is busy" message, hits 2 to leave a voicemail, and does so. (Exchange UM is running on port 5060 and PBXnSIP is running on the same machine, 192.168.11.4, port 5070) Thoughts? Thanks. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 5, 2007 Author Report Share Posted October 5, 2007 Hmm. That must be a problem with the trunk setup. It is revolving about the RFC3325 issue, on how to tell the other side what number to display and what number to use for authneitcation. Can you play with that trunk setting a little bit? Maybe we really need another setting in that area, but I hope not. Quote Link to comment Share on other sites More sharing options...
Krom Posted October 6, 2007 Report Share Posted October 6, 2007 Can is be said that 2.1.0.2114 is GA and ready for a production environment supporting multi-domain? Has it been "certified" to work with Polycom 2.2.x with bootrom ver 4.0? I know that both Polycom and PBXnSIP has certified and supported interop, but is regression testing maintained to be able to know which version of Polycom will interop with which version of Polycom. Or is that up to the end user of both products? Krom Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 7, 2007 Author Report Share Posted October 7, 2007 Well, we'll plan to make the 2.1 version GA very soon. The version that we were using with the Polycoms for testing the pbxnsip 2.1 was 2.2.0.0047. The bootrom is version 4.0.0. I did not notice any specific problems with that version. The buddy lists are still a problem, but that was also a problem in easier versions of the Polycom version. The good think now should be that the Polycoms can really put a lot of calls on hold (12) without running out of CPU steam - this was a problem in the previous releases. The last "official" certification we did was done by Tekvizion on a much older version. IMHO the updates since then both from Polycom and from pbxnsip side were improvements. If you want to play safe you should stick to that version or wait until there is nore feedback from the community. Quote Link to comment Share on other sites More sharing options...
Shawn Lawrence Posted October 8, 2007 Report Share Posted October 8, 2007 Here we go with 2.1.0.2114: http://www.pbxnsip.com/download/pbxctrl-2.1.0.2114.exe. Changes: Deal with the SDP bug in Linksys and this image also contains automatic provisioning for Linksys devices. We have tested 2.1.0.2114 and can finally confirm that this release works properly with G729 on a Linksys business phone. Thanks for fixing this for us. The bad news is we've found a new bug. When we upgrade our systems, all of ithe personal announcements on the mailboxes disappear. I tried changing the mailbox setting to Name Announcement and then back to Personal and it didn't make a difference. Any ideas? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 8, 2007 Author Report Share Posted October 8, 2007 When we upgrade our systems, all of ithe personal announcements on the mailboxes disappear. I tried changing the mailbox setting to Name Announcement and then back to Personal and it didn't make a difference. Any ideas? 2.1 has more than one personal greeting... Unfortunately it is not so easy to migrate the old prompts to the new prompt style. I guess manual explainations are more complicated than just doing it automatically. so check out http://www.pbxnsip.com/download/pbxctrl-2.1.0.2115.exe - it should migrate those files. Quote Link to comment Share on other sites More sharing options...
phsean Posted October 8, 2007 Report Share Posted October 8, 2007 Hmm. That must be a problem with the trunk setup. It is revolving about the RFC3325 issue, on how to tell the other side what number to display and what number to use for authneitcation. Can you play with that trunk setting a little bit? Maybe we really need another setting in that area, but I hope not. Tried all five Remote Party/Privacy Indication settings, but no effect... flopping that shouldn't need a service restart, should it? If so, we'll have to retest it after hours this evening. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 8, 2007 Author Report Share Posted October 8, 2007 flopping that shouldn't need a service restart, should it? You don't have to restart the service for that. Quote Link to comment Share on other sites More sharing options...
David McGowan Posted October 8, 2007 Report Share Posted October 8, 2007 I am now using 2115 and I'm not sure if this is an issue with the newer versions or not... I have an Agent group that calls for example 3 extensions... All of these extensions are set to redirect to a cell phone after a certain amount of time, which works if the extensions are called directly... But when the extension is called from the Agent group the cell phone does not ring... david Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 8, 2007 Author Report Share Posted October 8, 2007 Agent groups and hunt groups do not respect redirection settings. Otherwise your office will end up with a lot of redirected calls (and loops). The only way to "redirect" hunt group calls is to use the hot seating to another extension. Quote Link to comment Share on other sites More sharing options...
phsean Posted October 8, 2007 Report Share Posted October 8, 2007 You don't have to restart the service for that. Okay. We're keeping up with the latest builds (putting 2115 on tonight). You had mentioned not having the same issue with your UM server; please let me know if there is any other Trunk debugging that I could do that would be helpful to get this fixed up. And thanks! Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 8, 2007 Author Report Share Posted October 8, 2007 You had mentioned not having the same issue with your UM server; please let me know if there is any other Trunk debugging that I could do that would be helpful to get this fixed up. Well, regarding the UM trunk there are two settings: The redirect flag and the "Assume that call comes from user". The last one is important if you want to accept trunk-in-trunk-out calls (which Microsoft Exchange does) - then you need to charge an account for that. Someone needs to pay the bill, even in a IP-PBX. Quote Link to comment Share on other sites More sharing options...
Shawn Lawrence Posted October 10, 2007 Report Share Posted October 10, 2007 2.1 has more than one personal greeting... Unfortunately it is not so easy to migrate the old prompts to the new prompt style. I guess manual explainations are more complicated than just doing it automatically. so check out http://www.pbxnsip.com/download/pbxctrl-2.1.0.2115.exe - it should migrate those files. That appeared to do the trick. Thanks for fixing that. Quote Link to comment Share on other sites More sharing options...
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