Kristan
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Posts posted by Kristan
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Thanks guys, much appreciated
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Cool thanks - have you got any specific links I can have a look at and pass onto the customer?
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Has anyone hooked one up to their PBXnSIP install? I've got a customer asking if it's possible and I've found some devices which say they're SIP compliant door entry phone/door release devices, but they're rather pricey if they don't work. Has anyone got experience in this? Or maybe using other traditional PBX based entry phones and ATA's?
Thanks
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I'll try an upgrade tonight and see how it goes, thanks!
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Strange problem from one of our customers -
1. Call comes direct into extension.
2. Person can hold / resume fine - no probs.
1. Call comes into hunt group, gets answered by receptionist.
2. Transfers (attended) to another person.
3. Caller gets put on hold and resumed.
4. Caller can hear speaker, but person on PBX can't hear external call - it's just totally dead.
I have the full PBX logs... but they're rather lengthy! I'll post them once I've gone through them to see what's happening, but if anyone has any ideas or knows of this already....?
PBX version is 2.1.3.2316, polycom phones all on 2.2.0.
Thanks
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Great, I'll find our MSDN sub and get a copy installed to test with.
IPv6 is a looooooong way off yet for us thou
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I've just had the question asked, will pbxnsip run on windows server 2008.
I hadn't realised this was RTM already!
Anyone tried this yet?
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Do you mean buddy lists in the polycom sense or watched extensions?
I've disabled the buddy stuff on our installs (mainly though ignorance - I don't know what it does or how it works
) and the watched extensions works fine (on polycom SIP 2.2.2 and 3.0) -
The workaround we did was to ignore the state of the account (in a 2.1.6 build). That has the disadvantage that when the device starst up and there is a call active on the extension, the phone will not show that. When the call stops and the extension has a new call, then everything will be okay again.
Is this in 2.1.6.2448? I upgraded our PBX and still had this issue. I had to roll back due to snom problems, though that's a different story.
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I like the sound of that
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And apart from that I would put the files that the phone downloads directly into the tftp directory. According to the above procedure that should send the neccessary information directly to the phone.
I tried that, but according to the above rules because the file the phone requests has a "provisioning" in the title, it tries to match it in the pnp.xml file (which it does) and sends the standard config.
I can't really do it manually as there's a million and one options you can't set on the polycoms via anything other than the tftp/http files (and typically the one I want is one of these).
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We were thinking if we should start another project that plays MP3 files and convers them to linear, then streams then via RTP to a local MoH port. But maybe it makes sense to search if such a project already exists and just properly document how to use it. Unfortunately, MP3 players seem to assume they run for a few hours maximum, so they don't have to take care about memory leaks.
I've tried to find such a thing in the past but had no luck
I guess it's a fairly specific requirement. I imagine you must have 90% of the code to do such a thing already anyway?Agreed about memory leaks, winamp used to be terrible!
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Ok, thanks for that. But I want to keep the provisioning side of things as it stands, I just want this one phone to get a slightly different config file to the rest. What would be the best way to do that?
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Anyone any ideas? Could do with getting this one sorted if possible!
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You could do it with an agent group with no agents - you can have up to 9 individual "spots" on that.. but from memory it always plays them in order from the start.
An alternative would be to use audio in instead of wav for your MOH and have the spots queued up in winamp or similar, you can then just set that to loop and you'll get what you're after.
To be honest though, we've not had much success with MOH from audio in, it tends to die after a few days and callers get screeched at like a fax machine.
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I assume you're after some kind of disclaimer to be played before being connected to someone?
IVR nodes again :-)
Just create one and point it to the file you want and use the E string to match and the extension you want it to go to when they've listened to it.
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Yeah, it didn't seem to pick it up, just used the default again.
The URL being requested is via HTTP and is
/provisioning/polycom_sip_mac.cfg
whch in pnp.xml is mapped to polycom_sip.xml.
Is there a way I can put an exclusion in the regular expression so that it won't match for a particular mac address?
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I need to give a particular phone a specific config, different to all the rest. I've got custom polycom_phone.xml and polycom_sip.xml files in my html folder, and the phone requests the file polycom_sip_MAC.cfg from the PBX. I've tried dropping this file in the html folder, the PBX still seems to generate it's own.
Any ideas? I'm sure it's something simple, I just can't remember.
ta
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You'd basically setup 3 hunt groups, say 101 for sales, 102 for support, 103 for accounting, then have something like this :
!^1$!101! !^2$!102! !^3$!103! !^(2)(1|2|3|4|5|6|7|8|9|0)(1|2|3|4|5|6|7|8|9|0)$!\1\2\3!
The wiki on the IVR nodes is pretty good, but basically you have pairs of strings to match and numbers to dial, separated by ! marks. The ^ and $ mark the start and end of strings to match.
So for the first one : !^1$!101!
It strips the ! marks and separates them out to
string to match : ^1$
number to dial : 101
String to match is therefore "start of string" 1 "end of string" - so the number one by itself.
I'm not sure what the timeout pbxnsip has for determining if you've finished entering the digits, but it seems to work ok for the installations we've done this for. This, combined with the new "start" action for the auto attendant means you can build your own menus, get the customer to record them exactly how they want and direct the calls accordingly.
Hope that helps.
PS - I'm sure the pattern matching for the 3 digit extensions can be condensed down - I've just never managed to get it to work!
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I've come across this problem.. and I ended up doing it all in IVR nodes as it was easier, as you can do a string like this :
!^0$!8240! !^9$!400! !^(1|2|3|4|5|6|7|8|9|0)(1|2|3|4|5|6|7|8|9|0)(1|2|3|4|5|6|7|8|9|0)$!\1\2\3! !E!401!
Which matches zero and goes to 8240 (direct voicemail dial for 240).
9 which goes to 400 (dial by name auto attendant).
Then any 3 digit extension number, which dials said number.
Then if nothing is dialled, it calls the IVR node again to repeat the message.
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Well the problem is that when you call after hours to turn it on again then the PBX would wait until the next morning...
That's the way I would want it to/expect it to behave. Maybe I'm strange
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3cX
in PBX Lounge
Anyone used this or know how it compares to pbxnsip?
One of our potential customers is using it, and I want the dirt on it :-)
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I'd have thought the way to do it would be that you can dial and change the state whenever, but the next time the off/on boundry is passed it goes off or on according to the schedule.
That's the way my central heating works anyway
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The mib imports into PRTG ok, but seems to fail when trying to read them from PBXnSIP. I'm no SNMP expert, but I can do it using the manual oids.
I guess it's a fairly specific requirement. I imagine you must have 90% of the code to do such a thing already anyway?
IVR & SOAP
in IVR Node Setup
Posted
Hi guys,
A customer is wanting to see what option people are choosing on the IVR nodes when calling in. If I understand the documentation correctly, the PBX will make a soap call (which I can log), but does the SOAP then need to return a value to send the call to after that? What does the PBX do if it doesn't get a response?
Thanks,
Kristan