Vodia PBX
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Last week emails started going out to my employees that make no sense. For instance one user received 22 emails saying she has a missed call from ext 341 we do not have an ext 341 it does not exsist as a hunt group, Call Queue nothing. That happened to a few other users also, btw the phonne never rang. Then users such as myself get emails as missed calls from ext that do exsist but the call never came through and the user never made the call.
Another user keeps getting multiple call trace reports, he was in a meeting not at his desk for 3 hours and received 15 in about 20 minutes. I restarted the server this weekend and this morning all was well, however, they are starting again. Any clue where to look? What to look at? How to stop these?
The email from 341 does not neccessarily have to come from an extension. If there is a number configured on the PSTN gateway, the PBX would take that number.
Check the date stamp, maybe those emails were sitting in the spool directory for a long time and suddenly they are set free.
Restarting the server will not help. Check the spool directory if there is some nonsense. Maybe the PBX has the permission to write into the spool directory, but has no right to delete there? That would definitevely explain a lot of bogus emails...
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We are looking into the possbility to send all recordings out by email. There were some technical concerns, but after receiving a 15 MB email today (unrelated topic) it seems that email attachment size plays no role today any more.
And the nice thing is that once the email has left the PBX, it is not a PBX problem any more plus there are so many ways out there to sort emails into the right folders, that we don't have to worry about it.
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Seems like "Voip Phone 1.0" has a small problem with the registration. The phone coming from the IP address "10.255.109.71" never reregisters, the phone coming from 10.255.109.195 has the same problem.
Maybe this is because the re-registration interval is shorter than 32 seconds. Just an idea; it should be easy to fix that in the Voip phone software.
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Wish the issue was that simple, but we are using IP address instead of hostnames on the trunks?
Also as outbound proxy?
If the IP config of the host changing during operations?
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I can't find black-list. That is exactlty what I want...
It is part of the address book. The contact type describes how the PBX should handle incoming calls.
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They have upgraded to new version of pbxnsip and works fine do you know if there were any issues with pbxnsip and polycom 650 sip phones?
Overall, it should work fine.
There were a few minor hickups that have been addressed in 3.0 (e.g. TCP keep-alive, no buddy-lists was confusing the display).
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Just a quick question does pbxnsip support the presence feature on the polycom soundpoint 650 Ip phone. As we have a customer who has one but the speed dials keep disappearing we have checked the Config files and they are ok, we can not reproduce the fault (on trixbox). The phone registers and he then presses a speed dial button and gets a busy tone then hangs up and they all disappear from the screen of the phone.
You can use the setting "Watch the presence of the following extensions" in the extension to provision the extension that you want to watch with presence.
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Has anyone had this problem, i have installed 2.1.12.2489 2008/7/11 Linux (RedHat ES4) version and the 2.0/2.1 2007/11/16 File Archive (All OS) US English Prompts, i have unzipped the file and put the whole folder in the PBXnSIP directory but when i dial VM or try a feature code i cannot hear any system messages.
Is there another directory i need to put the files in? Any advice as its very urgent.
The folder name must be audio_en (all lowercase) and it must be readable. You might have to restart the service so that the system gets the files.
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I am attempting to use a calling card as a way to hide my cell phone caller ID, and send my office phone's caller ID. I am running 3.0.0.2976 (Win32) and am using your ANI fields to send caller ID to my ITSP. I find that nomatter what is filled in the trunk's account field, or in the ANI field, the calling card displays my cellphone's caller ID to whomever I am calling.
Hmm, yea that ANI field might have no meaning in the calling-card... At least in the way you are using it (SOAP might look different). Did you try to make an anonymous call? There is a flag for that.
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[5] 2008/07/11 18:34:07: Received incoming call without trunk information and user has not been found
That smells like the problem. Could it be that DNS is instable and then the outbound proxy of the trunk is not clear? I guess you get the above log message only when it fails, but not when it works okay?
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I am running 3.0.0.2976 (Win32) and if you select any music on hold other than the domain default, you only get silence
Ehh - you are right. Will be fixed in 3.0.0.2978.
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They're Polycom's. So far, we've tried it on 601s, 650s, and 550s and they all exhibit the same behavior.
The phones are on version 4.1 BootROM and 2.2.2 sip.ld
Hmm. Does that mean you also have a problem with attended transfer when there are two different registrations? Hard to believe. Maybe there is a flag that controls this behavior on the phone?
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Anyone seen this before or have any ideas what could cause this behaviour??
Nice movie!
We have seen a problem with the buddy list subscriptions. In the latest & greatest the PBX answeres with 4xx code when the phone wants to subscribe for it, that seems to solve the problem with strange icons on the 2.2.2 version. If you can, give version pbxctrl-3.0.0.2976 a try (http://www.pbxnsip.com/protect/pbxctrl-3.0.0.2976.exe, only Windows).
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On a phone with multiple registered extensions, placing the call on hold before using *85 to park call will always result in a failed parked call on our phones. Placing the call on hold with a phone that has more than one registered extension just "rolls" over to the second extension and tries to initiate the park on that second extension.
Whow. I believe that is not very intuitive. Also for attended transfers. What phone type is it?
However, we have found that if instead of placing the call on hold we can use transfer and use *85 or *85+extension and the parking feature works.Okay, that sounds like a short-term workaround.
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I have searched the forum for this topic and couldn't find anything. Is it possible to block-reject an incoming call from a specific caller ID. The scenario is this:
An extension has a DID # and tel:alias associated with it. Is it possible to block an incoming call if the call dials DID number directly? I couldn't find it...
I think the black-list would be appropriate? That would work for calls into an extension. For calls to other accounts, you could think about using a IVR node and do from-based routing.
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Maybe the issue is this:
Our office has a hunt group, in which all of our phones ring when a call comes in (sorry for leaving that out). Someone takes the call and tries to park the call on the extension of a coworker. If the phone that takes the call has 1 registered extensions, the park feature works fine. If the phone that takes the call has more than 1 registered extension, the park feature fails.
The phones are Polycom phones. Do you think it is an issue that is phone related?
Maybe you can capture two SIP packets: The INVITE that hits the phone, and the INVITE from the phone that contains the star code. I think if we see those packets we can say if that is the problem or not.
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Are there any know issue with fw 7.3.4 ?
Is it safe or useful to upgrade ?
From pbxnsip perspective I see no reason to move to this version. 7.1.33 is fine.
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I have the same version of PBXnSIP running on 2 different independant servers. If I setup the tapi client on my laptop to connect to one of them, everything works fine, however if I set it up to connect to the other one, then when my extension rings, I pick it up, and the call immediately terminates. The PBX logs the following
[5] 2008/07/09 17:02:20: Not setting dialog state of non-existing call port (call-id=b9578b47@pbx#4049)
[5] 2008/07/09 17:02:22: BYE Response: Terminate 86837862@pbx
Can you tell me what I might have configured wrong/different on the second system.
At first glance that does not sound like a TAPI problem to me. Can you check if click to dial behaves correctly? Maybe something "stupid" like a missing dial plan entry for that route or some stange characters that cannot be dialled.
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There is a reoccurring error message in one of our servers stating:
SMTP: Cannot resolve smtp.domain-messaging
I have checked all of the .xml files that I can find trying to pinpoint where the typo may be. Can you give me a few hints on where to look for this entry? Obviously the .com has been left off. I've restarted the service thinking that it might have been cached but this did not work.
This is on version 2.1.6.2450 (Win32).
Check the spool directory. But an upgrade to 2.1.11 will fix that problem as well.
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I may have not been clear in my original post. I apologize.
A call comes into my phone with 2 extensions 1 and 2. I answer the call on extension 1. The call is for another person on extension 400. I try to park the call from extension 1 to extension 400 by placing the call on hold then dialing *85400 at which time I hear the "failure" message. The call remains on hold on extension 1 and I can pick it up and continue the call. I just can't park it.
If I attempt the same scenario on a phone with 1 extension only and park a call using hold then *85400, the process works.
Maybe no misunderstanding... The problem is that when the phone starts a call, it needs to use an identity. Chances are, that when the call comes in to extension 400, but the outbound identity on that phone happens to be the other one, - then the PBX says "well that extension does not have an active call" (*85400 means park my calls on orbit 400, not park calls of [the foreign] extension 400).
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Yes, I think the delay was about 1 sec on this call, but it does seem to vary a little bit. It does seem to only happen when calling their hunt group. I tried an extension directly and the auto attendant and both seemed to have no delay connecting the audio stream. There is 4 extensions in stage 1 of the hunt group(none in the other 2 stages), is that to many?
Yes, I seems we have some trouble with the hunt groups and TLS. What you can try to do is use TCP or UDP instead (the latest versions have a setting for that in admin/PnP Params, so that you can provision this automatically). We need to find out why it is so slow, I cannot believe it is the encryption.
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Can you come up with a solution? Logging out the other identity just doesn't seem right to us...
IMHO it makes sense in most cases, but we can make it possible to turn this off. But we can add a global setting that suppresses this...
The new setting will have the name "multiple_hot_destking" (default is false) and it will be included in 3.0.0.2976.
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Here is the tcpdump from the pbx.
Hmm, why are all packets trunkated to 54 bytes? Anyway. Must be a tcpdump "feature".
In this trace the delay should be around 1 second, right? Does that happen only if you call a hunt group (= a lot of SIP traffic) or also if you call one extension directly?
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Setting the redirection using the web site works like a charm, but this is unfortunately not the feature we are looking for. The simplicity of *70 is great, we find it extremely usable, _if_ this issue could be resolved.
I agree on the conflict regarding outbound calls and don't see an apparent solution, as far as we are concerned, it doesn't really matter which identity is being used.
Can you come up with a solution? Logging out the other identity just doesn't seem right to us...
Okay, so we need to take a look into *70 again... But this will be in 3.0.
Possible Voicemail Transfer Bug
in Mailbox
Posted
Yea, the PBX does not wait until the user enters more digits. It is generally a good idea to make sure that all extensions have the same length (at least if they are numbers). Otherwise there are a lot of challenging problems when users try to punch in an extension number.
The idea is to make better use of the # sign to terminate input. But that is not such a small change, it is in so many places...