Vodia PBX
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the problem is i cant afford to reboot the service since there is contacntly ongoing calls on it
You don't have to reboot to change this setting. See http://wiki.pbxnsip.com/index.php/Global_Settings on how to do this on the life system.
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We are running rollup 7 and having the same issues. We have tried 4 different carriers as well as a t.38 fax locally.
I see the t.38 establish but it fails like above... does anyone have Exchange t.38 fax working? anyone?
We are working on a solution that transcodes inband into RFC4733 DTMF. In theory it is working, but in the real life we still have problems. Stay tuned, we want to include this into the 3.3.2 build.
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Since Upgrading to 3.3 we experience repeatedly a full system outage due to missing licence. We can see the licence code and it is correct. But above the licence code we read:
Current license state: No licence
License duration: Permanent
All we need to do, is just restarting the service, than the licence is coming out. This works for app. 3 to 48 h.
What's the best practise to localize the problem?
- The history is full of non-relevant events and messages: Which Level is appropriate?
- It seems that the system misses the licence from one second to another.
- It seems to have no bearing with a 3-minute-limit or similar.
- We have an Dongle-based NFR-Licence with Upgrade Protection Jan 2010
Maybe a defective Dongle could cause this problem? When the sytem "hangs" again, we only see there is "no licence". No idea to check if the dongle is "active". Any tool to do this?
There was a report about instable USB, which was even causing jitter every couple of seconds. A defective dongle, I doubt that. I believe that the regular polling of the dongle causes the driver to go crazy. We slowed the polling down significantly, but it seems the problem is still there.
What OS is that?
If the problem persists, ask for a MAC-based key instead (even if it expires after some time). That should help you to stabilize it until the root has been found.
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What about on this call. It looks like it is only sending out 1 invite to 205 and the extension is only showing 1 registration. Is the message-summary and dialog subscription causing problems?
[...]
Alert-Info: Auto Answer
Well, the PBX tells the phone to "auto answer". So it is not a problem with the number of registrations. This works only if the Polycom was automatically provisioned by the PBX, as the default config does not support the intercom mode.
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Maybe someone might know about what happens when 2 outside callers join the conference. Is this a getway thing to join the callers or PBXNSIP? or is it the snom phone that joins them?
Is there enough bandwidth between the PBX and the phone? If the phone runs the conference, keep in mind that there are two calls between the PBX and the phone. If you are using handsfree mode, you might want to check out the latest conference phone (snom meetingspoint).
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So the question is: When the V4 can be expected approximately? Will it provide hunt-group calls be also routed to cellphones?
We are starting to make builds for version 4, but it is far from being a release candidate yet.
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250-AUTH NTLM
Well, the PBX does not suppotr NTLM authentication. I guess you need to change the Exchange setup so that the regular authentication is accepted. Or maybe you can just whitelist the IP address of the PBX and skip the whole authentication part.
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Many thanks for your reply,
It is working and It was easy!
What should I do it if during business houres we have a humain receptioniste and all incoming calls goes to here? IVR? How?
You can put a one-person hunt group in from of the AA (the only person in that hunt group would be the secretary). However you will obviously loose the FAX functionality, unless the secretary wants to detect the tones and redirect the call to the FAX...
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On your #2 item: i agree it would be great if the DND button would make the phone ext. BLF light come on. This would be great to let phone users know someone has their DND "stuck" on.
Regarding the DND button there will be a fix coming out in 3.3.2. That area was not very clean and should be better now.
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what a pain this is.
got a polycom phone I wiped all the settings. provisioned it for http and I kept getting 401 errors. arrgh
changed the setting to tftp with the same auth (101@domain1) same password and it worked!
I pulled the tftp manually and it just references URL's so I assume the auth is still used as it should however it seems the initial file request is BAD.
Maybe send me a PM with the provisioning info and we'll give it a try from here. If you have Remote Desktop or at least a HTTP login we could even try to get it working. Seems this is a problem of the test environment.
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Maybe just turn TLS off. In the email settings there is a flag called "Enable TLS", try turning it off.
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anyone!
Oh the AA does not use regular expressions. Just put a "F" into a "User Input" and then the FAX extension into the "Destination". The AA will detect the FAX tone similar to a DTMF tone with the name "F" and will behave as if the user entered "F" during the annoucement.
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If that's the case, we'll be trying to get some real certs instead of the one's from our own enterprise-CA.
This should be a certificate problem. IF you get one, make sure that you use a 1024-bit certificate. I remember there was an issue using more than 1024 bits.
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Aha. Dit not try it... But I believe it should be simple. -
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yes, an i want to and its not passing the setting to the phone.
so what am i doing wrong? running 3023and I have this snom_3xx_custom.xml in my html directoy
The XML looks fine. The question remains if the phone loads it (or tries to), you can see that in the log of the phone. If not, you might have to move to a newer version - not sure where the additional custom file was introduced.
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I was hoping I could somehow modify the ANI by using a different dial plan. So if I wanted to use ANI 310xxxxxxx I would dial "44"(or something) in front of the number and have that dial plan modify the ANI, but I guess that cannout be done?
Can the "replacement" values modify the ANI?
This seems like a great feature?
You can make the ANI depend on the trunk. For example, "Trunk1:9787462777 9781234567" would mean: When using Trunk1, then use the ANI 9787462777; else use 9781234567. If you want to use that, trunk names must not have a space.
Maybe that is the way to get this working.
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yea but im trying to add these setting to the PNP that the phone gets.. isnt that what the custom file is for?
There are different layers here... The PnP parameters are for decisions that the admin has to take when using the standard PnP procedure. The custom file is for those "millions" of settings that usually not neccessary; this is just to provide more flexibility. Most admin would never use the custom files.
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I understand how you can set the ANI field in the extension.
But I am at my phone making calls. One call I want to use ANI 310xxxxxxx and on another call I want to use ANI 310yyyyyyy.
How can I have this ANI value change based upon a specific call from the same extension?
Oh, sorry misunderstanding. For that, I would suggest two extensions on the same phone. Many SIP phones allow you to have multiple registrations. This would solve the ANI problem.
However, it comes at the price of increased complexity for the user when picking up calls & Co. There is always a trade-off.
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But when we try to route icalls to a hunt group number, we get an error indicating the number
[5] 2009/04/19 10:09:57: Dialplan Standard Dialplan: 111 goes to extension
[5] 2009/04/19 10:09:57: Could not start call to extension 111 because there is no registration or the extension is busy
If we route the call to an extension that then forwards the call to the hunt group we get the same problem.
Hmm. That simply was not envisioned... Sending a call from the dial plan to an extension was really only for an extension. We may provide you a build that also allows calling other accounts like extensions.
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I think I've heard somewhere, that I can assign more MAC-Adresses to one extension, so this extension becomes available on more than one device. Is it possible to set the 2nd, 3rd, 4th SIP-Identity on SNOM-Phones over PnP-Mechanism of the PBX or do we really have to manually login to every phone and do the settings by hand?
What we intend to do (maybe anyone has an better solution for that?) is:
- assigning on every phone a 2nd and 3rd (...) SIP-Identity, so that two extensions of the PBX are registered by every phone in the office as additional SIP-Identites.
- When a call is received on one of this multiple registered extensions (each one is the only one agent of an Agent Group), all phones are ringing at the same time (I know: this can be done much easier), BUT: As long as this call lasts, the appropiate agent group is busy and the next incoming call is put in the Queue, REGARDLESS on which phone the call was accepted.
I do not have a better idea, since the following alternative solution don't solve the problem:
If we do it without multiple registrations, and put each phone with it's different extension number into the (personal's VIP) agent group, the other phones would ring on the next incoming call to this agent group (which is used only for one special Person). There seem to be no way to set the Agent Group "busy on busy" (or maximum Calls on this Agent Group = 1).
It was already possible to assign a MAC address to multiple extensions. Then the PBX would provision multiple identities to one phone.
Version 3.3 introduced a new feature. You may now also list multiple MAC addresses for one extension (seperated by a space). This makes sense e.g. if you have a cordless device and a desktop device on the same extension.
Usually it is a better idea to solve multiple ringing with a hunt group or an agent group. Having multiple extensions on the same PBX on one phone usually just screws outbound calls from that phone up. You never know what number is calling, and especially for call pickup and call parking things get very messy.
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We have added an additional header called "Related-Call-ID" that makes it possible for the service provider to verify that we are not spoofing the Caller-ID and the outbound call "relates" to another inbound call. AFAIK callcentric.com has already happily implemented this and their customers can enjoy a proper and safe cell phone redirect.
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I need to have all incoming calls that comes in as 0045xxxxxxxx showed as xxxxxxxx
How do I do that???
If you set the country code and the area code, the PBX will try to present the number in a "human readable" format. Be a little bit careful with this feature (backup before doing it), as changing the country code triggers a table scan for the address book and internally reformat the address book entries as well.
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I would like to see a LED status on cell calls.
E.g. if I forward my direct number (DDI / local ext.) to my cell, and a call is coming in (and is forwarded to my cell) the busy LED will light on the phones which have status LED on my number activated, as long as I am talking(with the forwarded call)
We are meeting this demand more & more from the customers, as several other vendors are offering the function.
Any ideas when it will be possible to have this function ??
At the moment is it a little bit "foggy" if we should treat such cell phone calls as trunk calls or as extension calls. So far those calls were trunk calls (like a stupid redirect), but it seems more and more than trunk calls must be also be extension calls. As we are cleaning this area up in version 4, that will be a lot cleaner.
Aha. Dit not try it... But I believe it should be simple.
so what am i doing wrong? running 3023
Agents group drops call
in Agent Group Setup
Posted
Does the ACD connect the call (meaning: does the caller have to pay for the call)? There is a option that tells the PBX to connect the call only when the agent picks the call up; but for a busy group that is usually way too long. What is the event for connecting the call?
If the handset is ringing for so long you should redirect the call out of the queue, for example into a hunt group. It is just not good to keep the handset ringing for so long. You might even redirect the call back into the same ACD; customer will not be amused to get kicked back into the queue (especially at the beginning of the queue!!!) - but this is still better than a dropped call.