Vodia PBX
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Hmm. Currently that is not possible, at least not with the mailbox. IVR node comes to my mind, but the setup will not be straightforward. I like the idea with the max size of zero, but that is not in any version right now. We'll put that into head, but it will not make it into 2.1 any more.
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That should not be a big problem.
Just put into the domain name of the trunk whatever they would like to see. For example, "level3.com", "today-is-a-good-day" or "10.11.12.13". The PBX just copies that string into the request-URI. The routing is being done by the outbound proxy setting of the trunk. There you can use stuff like transport=udp and IP addresses.
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It should be possible to trouble shoot the one way audio independently from the call redirection. You say that making a call through the VPN is no problem?
Does the VPN go up and down? The PBX has problems when the IP configuration changes. Going to the status web screen of the PBX re-reads the configuration, so if you want to refresh the internal tables for the IP config you can just go to that web page.
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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.
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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.
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You should do that on IP level. E.g. in Windows check out the route command. You must make sure that the OS would send the traffic on that NIC.
http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses might also be of interest to you.
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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?
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Can you comment on the Trunk to Trunk feature in 2.1 and how that would get around the need for adding numbers per Jan's post. Is there any configuration necessary for this feature, if OCS is sending in E.164 format? Thanks, Fred
When the Exchange server wants to initiate a call (e.g. through the address book), it must tell the PBX to initiate an outbound call through another trunk. That is a tunk-initated outbound call. We believed that this would not be neccessary (and it is a significant security risk), but that's the way does it and we enabled that feature. You don't need to change anything in the configuration, because this feature is enabled when you click on the "allow redirect" button in the trunk settings.
The E.164 problems revolve around the pattern matching in the dial plan. In 2.0, you could do that but would break your fingers eith the extended regular expressions. In 2.1, you can do this also with the simplified dial plan. For example, the pattern "+49*" is now a valid simple pattern.
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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.
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What you can do is have actually three domains - two "real" domains and one domain that essentially just has the (global) trunk which is visible in all domains. The inbound routing is done using the tel:alias names (e.g. tel:2121234567), http://wiki.pbxnsip.com/index.php/Inbound_Calls_on_Trunk. Outbound is easy, you can use the global trunk in the domain's dial plans.
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Does anyone have any experience in getting PBXNSIP working with speech server 2007?
Yes there was someone using it with the speech server and it worked. The setup was prettys much the same as with Exchange - I guess they are using the same underlying components.
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Well, if you are in the LAN and there is no packet loss than that is a very doable option.
Otherwise you better use T.38. More and more gateways support it, but unfortunately this standard is very much different from RTP and the different vendors are not talking well to each other.
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Here we go with 2.1.0.2113: http://www.pbxnsip.com/download/pbxctrl-2.1.0.2113.exe.
Changes: Some Email servers had a problem with the base64-encoded format, that was changed. Also the SOAP CC authentication (PIN) was not working. Credentials based on P-Preferred-Identity (inbound) were not working. The ACD IM messages could "stall". Linksys provisioning files updated.
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Are you using CO lines in the PSTN trunk? If you put co1 co2 co3 co4 there then you should limit the number of calls that the PBX allows on that trunk to 4.
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Well, we had it! The PBX was sending MIME/alternative, but it was causing a lot of grief, so we took it out and now send only HTML.
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Well, the help/help.php is really hard coded. In 2.1, we changed it at least to help/help. What you can do is putting a subdirectory in front of it, e.g. make the setting look like this: domain.com/pbxnsip. Then it should be possible to have the help page in that location.
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The missed call emails that I am getting are okay.
Are you using your own files in the html directory?
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At the moment we don't support CSTA. I have flipped through some presentations and it looks doable.
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Ehh... Not sure if it is available in 2.0.3.1715. It would be in the System Admin/Settings page. But it will be for sure in 2.1.
In not, there might be a setting called "camp_enabled" in the pbx.xml file. If it is there, change it to "false". That would require a restart of the service.
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Forking calls was supported since version 1.0. Just register an extension twice and they will ring at the same time.
I don't know why some people have so many problems with this trivial feature.
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Ok, I've got the linksys auto provisioning working...
Very cool!!! If you send me a private message I can grab the files and put them into the Linksys Phone forum and also put them into the standard distribution for the PBX. The current dial plan file is attached for your reference.
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Just go to the admin settings and turn "Offer Camp On" off.
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Okay, seems we have to iron the memory during the startup phase... Will be in the next latest and greatest!
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Permission problem in the forum... Try again.
Okay, 2.1.0.2108 is out
in New Features/Versions
Posted
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.