Kristan
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Posts posted by Kristan
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In a similar scenario we setup the following:
HQ - extensions 700-799
Office1 - extensions 100-199
Office2 - extensions 500-599
Each office has a trunk connecting to each other office, and simply in the dial plan there is a line routing the calls, for example:
HQ:
Pref Trunk Pattern
25 Office1 1xx
25 Office2 5xx
The trunks are setup as sip gateways, not registrations, so I don't see how SIP registration times would have an issue. The PBX being called identifies the call coming in by matching the IP of the call to the outgoing proxy IP on the trunk and then routes it accordingly - same as it does for calls coming in from the ISDN gateways.
I tend to avoid using complex dial plans involving stripping characters etc. unless really necassary, and in this case we don't need to do anything special.
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ta!
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A user has recorded a personal greeting and wants to go back to name only. How do you remove it now it's not selectable in the drop down? I've tried setting it back to anonymous, then to name, but neither seem to have any affect.
Help
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Just as a test, try renaming noise.wav in the the audio_moh folder to something different and replacing it with a 0k file (just create a text document and rename it).
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As long as you leave the bit about the odd bug in
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We have a company with an office locally and two in remote sites. All are connected via permanent VPNs with local PBXnSIP installs, local PSTN access and calls routed directly to the SIP provider. Each office has a trunk setup to each other office, and extension numbering is done so that phones in other offices don't overlap, i.e. office 1 is 100-199, office 2 200-299 etc. Each office can then call between each other and it's still an "internal" call and each office routes it's other calls according to the local preferences.
Personally I'd do it this way, we did another branchwhere they had no local PBX, just phones, which registered back to the main office. We routed a local number to the branch over sip, but this install was less successful as the VPN had a habit of dropping, at which point all calls failed - not great! Obviously it was a network problem and nothing to do with PBX's, but the local PBX installs help mitigate the risk of loosing remote connectivity.
It depends on your setup and requirements really. If it's only for a couple of phones in a branch it might not be cost effective to put a whole new PBX in, but if you want belt and braces redundancy, it's probably worth doing.
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Excellent thanks!
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I can't really recommend PBXnSIP highly enough. The PBX itself is great, the support is first class, and for the price I think it can't be beaten.
It takes me no time at all these days to knock out systems, especially using either snoms or polycoms as the autoprovisioning makes everything a breeze. Users like it because the web interface is easy to use and I like it because it's so flexible.
Yes, there is the odd bug that appears but generally it's stable and works well, and if you do find something, the PBX support guys are onto it straight away.
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Oh you mean, a number so that you get directly to the dial by name? Interesting thought... I think at the moment there is always some DTMF interaction neccessary.
Yeah, that's the idea, it would just mean I can basically build a complete auto attendant with IVR nodes and regular expressions exactly as the customer wants. BTW, the regular expressions seem a bit broken - the example on the wiki page for the IVR node matching 3 numbers doesn't seem to work!
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It depends on the type of phone and what it supports - see my post here :
http://pbxnsip.invisionzone.com/index.php?showtopic=598
Snoms do, I think Linksys do, polycoms don't.
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Is it possible (or is there already) an IVR destination for dial by name? I have a customer for who the auto attendant doesn't quite fit and I can do everything they want via IVR nodes, except for dial by name.
As a bodge, I've had to take out all the aa wavs they don't want to hear, so when the IVR node calls the auto attendant, it simply says "for dial by name, press 118". However as they've already selected dial by name from the IVR menu, it' a bit redundant and they would like a way to get into it without having to dial again.
ta
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Can you post the full invite?
In the logging options for the domain, set it to log "other sip" messages.
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I remember seeing something saying you guys were working on this, do you know if there's any idea of release date? We're about to be split between two offices, and it'd be handy to have for ourselves
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It doesn't look like the phone supports config exports. TBH, I'm kinda disappointed in it - the web interface is very different to that of normal snoms, and whatever it's running underneath doesn't look like it's as flexible as the standard snom 7 firmware. I guess I was expecting basically a 300 with cordless handset, but this is clearly a very different beast!
edit:typo
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Just tried a sparkly new snom M3, but I don't seem to be able to get it to autoconfigure.
It requests /m3/settings/MAC.cry
Can I fiddle the pnp or html files, or do you guys need to add something in?
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Thanks, I'll put a request into polycom for RFC4916 support
Which will no doubt fall on the same ears as my request for RFC3326 support
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I install wireshark as standard on all our PBX's, it just makes troubleshooting so much easier. I haven't noticed any major CPU problems, but then most of ours are well over spec'd anyway.
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My partner login is a waste of time, I never seem to be able to do anything from there!
I was speaking to our supplier and he mentioned it was available. I have the release notes but no firmware as yet (it hasn't appeared on the FTP server). One major change is that polycom have launched what they call their 'Productivity Suite'. It's a set of optional features (including LDAP directory access amongst others) that you have to buy an extra license for (per phone). It's not like their stuff isn't expensive enough already!
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Actually it looks like we're all behind, polycom released SIP 3.0 firmware on the 4th Jan...!
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Would someone be able to point me to an authoritative pbxnsip version of the polycom 2.2.2 config xml files?
<snip>
looks like Kristan obtained them somehow -- always a step or two ahead of me
I don't think I'm ahead of anyone! I have the polycom supplied 2.2.2 reference files, but not the PBXnSIP customised versions. I'd be quite interested to have a look at them myself I have to admit, just to see the differences.
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With the polycoms, if you blind transfer a call, the person you're transfering it to see's the external persons number as if it's coming straight to them - that's good. However for an attended transfer, after the initial conversation with the person who's transfering the call to you the call continues as if you're taking to them; the display doesn't change to reflect you're now talking to the external party.
Is there any way to make it do this? From memory, the snoms do it, so I'm assuming it's a polycom quirk as opposed to pbxnsip. What would I be looking for in the polycom documentation?
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Just tried these with the default polycom config and it doesn't seem to work. Is there anything special I need to do to get them to work, or do the polycoms not support it?
Thanks,
Kristan
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I agree - that's why I don't want to have it failover on a 486 busy here , but I would like to failover on a 404 not found, 401 unauthorised or 408 timeout and possibly a few others. We've had instances where the VoIP provider has changed something and as a result all calls come back with 401's and die. In these sorts of cases I'd rather it failover to a secondary trunk - if we're seeing them it's clearly a provider error. Obviously for normal responses like a 407, 486 or 487 it'd be stupid to failover as they're just part of a normal conversation.
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End users are difficult to satisfy with a Wiki page
You could have left it at "End users are difficult to satisfy"!
The main this is just so we can give them documentation that matches the version they see on the domain admin pages.
PBXnSIP and MITEL
in Trunk Setup
Posted
The PBXnSIP side was very simple.
Have fun with the mitel!