Vodia PBX
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If you do automatic provisioning of the Polycoms from the PBX then they should do it already.
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We also wrote something up a thttp://wiki.pbxnsip.com/index.php/Click_To_Dial.
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Well, that is a long discussion.
The main topic is QoS. Most phones that have a switch built-in have no problems with the switching load, as it is done by a piece of hardware. In order to ensure QoS, you practically need VLANs. Many companies solve the problem by just putting all VoIP gear into one VLAN and leave the rest in the VLAN 0 - which is the regular LAN. Then if you tag the VoIP VLAN with a higher priority, you can be sure that all packets make it.
You can also use phones to perform the tagging work. As many PC do not natively support VLANs, it becomes the job of the phone to put the PC into a specific VLAN with a specific priority.
And once that you have different LANs (may they be VLAN or LAN), you need to start routing between them. That means, you need to set up a router (or a PC that can act as a router) to send the packets back and forth between the LANs. This might become a little bit tricky as well.
Also, usually you need to operate a DHCP server in each VLAN. If you run a PC with different VLANs attached, you possible can use the same DHCP server for the different LAN.
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Well then that is the problem. Just keep the name localhost if you have one domain and you have no problem.
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Are they using the domain name "localhost"?
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What firmware are you using? Any chance to move to the snom 6.5.12 image?
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The PBX has a built-in NTP client that calculates the current time. Running NTP while the PBX is running screws up all internal callbacks, so if you must run ntpdate do it on startup, but then not again. Complain to the authors of the pthread library, they don't wait for the CPU time, they wait for the "real" time.
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The 2.0.3.1715 is the latest release. Later builds are builds, but no releases. 1715 seems to be pretty stable and unless you are not missing features there is no need to move to another version.
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Does it support SIP? Is there any pointer?
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Not at the moment.
But if you hold the call, many phones have a hold reminder.
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You mean you want to have service crossing midnight? That really might be a problem. Workaround is to do it to 23:59 then start the other service at 0:01.
Maybe it is also worth a try to have service ending at 24:00 and the next one starting at 0:00. But I am not 100 % sure if that will work.
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Currently not... But we heared this requirement also from other sides. It is somewhere on the to-do list.
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No, if the call should be connected automatically, the other side must support auto answer. The PBX is flexible in signalling that, but there must be some way of indicating this.
I am actually not sure if that is possible at all in FXS, because the phone needs to pick up the line. Maybe you should look for an analog phone that supports automatic answer.
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You can use the logging feature in the extension's registration to see when a phone looses registration. This is very useful to trace problems with registrations.
In Wireshark, you can filter by IP address (e.g. host 192.168.1.2) during the recording, so that the file size stays low and you have a perfect snapshot of what is going on.
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Maybe try snom version 6.5.12 (see http://wiki.snom.com/wiki/index.php/Snom36...es#6.5.12_beta), seems there was a problem with lost subscriptions.
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Check:
- In the domain settings, the setting "External Voicemail System" is empty.
- The mailbox of the extension is enabled in the settings of the extension (check this in the domain mode).
- In the domain settings, the feature code for calling the mailbox "Go To Voice Mail" is set to *97 and it does not overlap with another feature code.
- The license is still valid.
- In the domain settings, the "Mailbox Direct Dial Prefix" is set to 8 (this is important it you want to call the mailbox with the 8 prefix).
- In the domain settings, the setting "External Voicemail System" is empty.
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Is the 10 extension limit a 'hard' limit on the system or can existing extension licenses be purchased. I understand there's obviously resource constraints, but say we sell a 410 to a small office, who then grows to 11 or maybe 12 extensions - is the only option for them to upgrade to the CS425?
Of course it is a soft limit. But we don't want to end up selling every license one by one.
I know when I've tried early versions of the Asterisk analog cards, they didn't work well in the UK - due to differences with impedence and callerID. We generally use external ISDN gateways for PSTN Connectivity, but it would be good to use the FXO ports if need be. Do you know if the hardware\libraries etc support the UK telephone network? (If they don't, is this planned?)Yea, the US phone system works slightly different than in the UK. The first focus was on USA, but later versions also consider DTMF caller-ID and polarity change disconnect indication. Not sure what applies to UK. The good news is that the hardware should be able to do it, it is a standard FXO hardware you find in every gateway.
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I heared that practically only ACME and Nextone are left in the SBC arena. They are hardware based. As far as I can tell they are both pretty good. Don't expect many blow's and whistles, but these devices are rock solid and that is what you expect from a SBC.
Interestingly, the do also RTP relay. On other words, once that the RTP makes it to the SBC, it is only a ms away from the PBX which makes the whole RTP relay discussion quite pointless. Jasomi tried to do "media path optimization", but IMHO in the real life the environments were so difficult that this did not work in all cases.
Maybe we will find out that the good old class4/class5 architecture was not so bad. With the difference that this time, the Centrex is an economical interesting alternative to the CPE-based IP-PBX.
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Strange.
What raises my eyebrow is that the domain index is "2" - do you have more than one domain? If the PBX sends the call to the wrong domain it is no surprise that the PBX does not find the address book there.
Also, make sure that you don't use the NANPA dialplan in the domain. Your numbers start with "0" and that does not look like a U.S. number.
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Well, the trouble ticket procedure is described under http://wiki.pbxnsip.com/index.php/Trouble_Ticket_Processing. But there are a couple of people who can help you directly without opening a trouble ticket. Maybe we should list them somewhere.
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Maybe you have a gain problem. If the gain is too high you would hear static cross-talking. Try reducing the gain and see if the effect goes away.
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Check what the log uses for time stamps. Then you can see what the PBX thinks the time is.