Vodia PBX
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I verified that the time is correct by looking at the current calls at the STATUS screen.
Hmm. Seems that the logic is screwed up again. That clarification for the explicit times was missing one negation - in other words the times you specify there are the times when the PBX would not call the cell phone. We'll turn it around, that will make life a lot easier.
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major hickup as in 50 INVITES all at one time? it shouldn't create 50 RTP stream.
Even if there is no RTP 50 INVITE are a challenge. Think about TLS, DNS, ENUM and all the blows & whistles that need to be taken care of. Plus people are waiting in line for address book lookups for all these 50 agents in their personal address book. Ouch!
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TOC TOC ... there is someone ?
Well, as said - the FAX recognition did work in our test environment. To me it looks more like a problem with the environment. Maybe the PSTN gateway does signal out of band, and then the PBX does not analyze FAX tones any more. Maybe try turning RFC2833 DTMF off on the gateway and tell the PBX to perform inband detection (admin/settings). And also set the codec preference to prefer ulaw, maybe there is a problem with alaw.
If that works we can find out what we can do to use the out of band-detection cabability of the gateway to save CPU power.
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well this issue is not only the auto attendant but internal users who are trying to call each other!
is there no way to make a dial plan that only calls with 1XX should use the trunk that connects the two systems and all other calls beggining with 1 IE 1XXXXXXXXXX should go out through the regular sip trunk?
please advise asap
I FIGURED OUT HOW TO DO IT...NO PROBLEMS AT ALL!
all i did was made a dial plan as follows:
Pref Trunk Pattern Replacement
100-NY 1xx
100-NY 2xx
110-Callcentric *
so if the number beginning with 1 has three digits it goes out to NY trunk and if it has more than three digits it uses callcentric.
is this good??
I would change the preferences a little bit to 100 and 101 to make crystal clear what the preference is. But apart from that, that looks fine.
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UPDATE:
It looks now like the MWI is cleared because the phones are resetting. Ever since the update to 3.0 the phones have been having frequent resets. I have not noticed it on my phoneset - although other users see as many as 25 resets a day.
I tried the latest firmware on the phones but the resets keep happening.
Could there be some change from 2.x to 3.0 that requires a tweak in settings on the SPA-942 phones?
You mean "reset" like in "reboot, crash"? Does this happen out of the blue? Or is there a specific event, like an incoming phone call? Maybe it is just re-retching the configuration and for some reason thinks that it changed (which causes a reboot on the SPA devices).
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1XX needs to stay the same because the cards are already printed...do you have any options whatsoever to resolve this?
Well, you can deal with that. But then you have to work with timeouts (in the AA, direct destinations). It works, but is not very beautiful.
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Am I missing something here???
Hmm. Does the PBX show the right time (e.g. in the log)? Maybe the domain is in a different time zone...
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Hi what would cause the system to continuously change the ports of the phones?
[3] 2008/11/04 09:22:58: Source address for sip:712@xx.xx.xx.xx has changed to udp:xx.xx.xx.xx:50702
[3] 2008/11/04 09:23:06: Source address for sip:715@xx.xx.xx.xx has changed to udp:xx.xx.xx.xx:50703
That is usually caused by routers that are not "ready for SIP" - their NAT implementation changes the ports. Or your refresh interval is too short and the router already closes the NAT binding.
The result is that intermittently you cannot call the extension.
It is a serious problem and you should fix this either by changing the router or by making the interval short enough for these routers.
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We just cut over a few companies to this platform and have CO lines setup for them as they are doctor offices and need things simple and as close to the way they are use to things as possible. We unfortunately don't have the incoming calls showing on the co lines. We originally got support when we couldn't get incoming calls and were told to have all trunks as outbound only. This seems to be our problem but need it resolved and working properly ASAP.
Oh, so you are using multiple domains on the system? I assume that the inbound traffic does not have the domain name in the request?
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i am in the process of setting up two offices in two different states.
i have one pbx in each location, i have successfully trunked the two pbx's but have the following issues with dial plans.
This is a case of having seperate domains that want to call each other. Although they are located in different states, the case is not so much different than having the domains in the same data center, but on different CPU.
The extensions in Office #1 are 1XX and 2XX and the extensions in office #2 are 3XX.My questions are as follows:
1) What is the best way to setup the dial plan that when office #1 wants to call office #2 they can enter their extension number only without having to add a fourth digit to the extension. (ie a dial plan with 3* and a replacement of 3*??)
in addition to this since office #1 has an issue where some of their extensions begin with 1 (which is the possible prefix to all US calls) what is the best way to maintain 3 digit extensions without running into issues?
First of all, you probably need two trunks in gateway mode. Both of them should trust each other, e.g. turn the accept redirect on (that should help with the question 2 below).
I would stay away from prefix 1, it has a lot of problems (1xxxxxxxxxx being one of them). Better choose something in the 3xx-6xx area, if the business cards are not printed yet... You can use pattern 3xx and no replacement to route calls that have three digits. If you want to make a routing entry for x11 then you can just give that one a lower priority.
2) I would also like to setup that if for whatever reason the SIP trunk from office #2 goes down they can backup off of Office #1's PRI line for outgoing calls...how do i set this up?You need to set the setting "Assume that call comes from user" for that. Then the PBX can use the dial plan of that extension, and it can also charge that extension.
3) When someone calls into the Auto Attendant at office #1 i need for them to be able to call office #2 by dialing the three digit extension (currently it gives a message saying that "you do not have permission to call this extension")Hmm. Ideas are: A. Use the direct destinations (if there is still space available). B. Set up "ghost" extensions with static registrations that point to the other system, have no mailbox and an impossible-to-guess password. Not very beautiful, but that might solve the problem if there are noo many extensions in the other location.
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After the upgrade to version 3.0.1.3023 we have been noticing our Linksys SPA942 phones have started having their MWI light randomly cleared somehow on their own without the user first checking the voicemail. With out the indicator on users are not notcing the voice mails they have because they cant see a reminder on their phone.
We tried resetting the phones and the PBX server but it has not resolved the issue.
Hmm. That seems to be a problem only for the SPA942; other phones don't have that problem, right? Any chance to get more insight? Maybe you can the "Log Watch List (IP)" in the Logging to the IP of the phone and monitor the traffic. Does the counter change or the "Messages-Waiting" (yes/no)?
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I know you can make this work but for the newbie it would be nice if there were a simple menu option in the ACD stuff.
It is generally not a good idea to ring all agents "immediately" because this creates a major hickup in the system, sometimes even in the network. IMHO gradually calling all agents is not so bad.
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Is it hard to add a trunk limit?
and then allow certain trunks to not limit? Such as an exchange 2007 trunk group.
I ask for alot of stuff eh.
EDIT I suppose i could limit the trunks in the trunk form eh. just put in co1 co2 etc.
But then it eats my accounts licenses away fast.
In the hosted environment, CO-licenses should not be the problem. But the good news is that we could just use the trunk allocation mechanism to limit the number of calls in that domain.
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On standard key telephone systems, which we emulate with the pbxnsip, the DND key usually causes any blf assigned to an extension to be luminated when the dnd key is activated on the associated phone. This should probably be optional on a per domain basis to make it backwards compatible. We are getting several complaints on this from Telephone interconnects that are deploying this platform
What version are you running? Could be a bug that was fixed in the latest & greatest.
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I got a Polycom IP sound Station 4000 already register and it can make call but when I call this phone I cannot pick up the incoming call I'm running version3.0.2.
I have tried several things to make this IP 4000 work with the PBX N SIP and it still doesn't work, I don't know what is the problem. I have down graded the firmware to 2.2.2 and it does not register to the PBX if I use the 3.0 it register fine. But I can (pick up) answer any calls.
That sounds like the PBX provides a contact in the SIP invite that cannot be routed in the phone. Does the Contact header contain something that the phone can reach? Maybe you can post the INVITE (from the PBX log) here and check if the IP address in the contact header is what you expect.
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We are having the same problem running Windows OS. 3.0.0.2998
Is there an upgrade for this?
It will be included in 3.1.
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Can you force a number? What if your ITSP does not support CID? Before the number is said a text is spoken? Like "Hi this is pbxnsip forking service, number blah blah?" And this "Hi this is pbxnsip forking service" is a wav file which can be changed/customised?
It gets straight to the point and says "978-746-2777 is trying to reach you. Press 1". That has the benefit that you hear the original caller-ID (solving a common problem when redirecting the call) and you hear it right at the beginning.
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have just checked with http
it works!!
so it looks like it do not work with https
:-S Yea we had the same issue with the calendar document. Seems it is something with the certificate and the https paranoia. Maybe you need to get a thousand dollar certificate before your browser eats this........
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Have tried before - and just now again - do not work (??)
Any other ideas?
Are you getting anything? What does the web browser say? Are you using https?
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we can not make this function work "click here to download the list of users."
Try "save as..." and then open it with a text editor.
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under Edit Domain there is this setting, Maximum Number of Calls:
Is this maximum number of trunks or total calls? so if I make an extension to extension call does that count as 1 or 2 calls?
It would be nice if there were 2 options. trunk limits and call limits.
At the moment the meaning is "call", not call legs or "trunk calls". We had this discussion recently, and the result is not clear yet...
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On this, I have been testing build 3033 and have a few questions so on these scenarios:
1) I dial out on my extension that has an ANI set, it properly passes that ANI 40612345678 in the FROM header as set on the extension and using the NAPNA 10 trunk setting
From: "Carl Johnson" <sip:40612345678@192.168.40.218;user=phone>;tag=574149672
2) I dial into the PBX with a header of:
From: <sip:4067890000@rcp.local:5060>;tag=3184c2f731d55ce
then that extension is set with a proper ANI and is setup to redirect calls outside via the same trunk as used above the system isists on sending the users inbound caller-id, which is not what is hoped for can we resolve this somehow our our circuits lock us into to providing out caller-id that is valid and known as a good DID on that circuit?
From: <sip:4067890000@rcp.local:5060;user=phone>;tag=1529366861
instead of the .. I think expected from header of
From: "Carl Johnson" <sip:40612345678@192.168.40.218;user=phone>;tag=574149672
We have introduced a new setting for the trunk that tells the PBX hoe to represent a international telephone number (+ sign in the beginning, 011, 00, and so on). It is still not 100 % stable. Not sure if you want to try that one out.
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i have it set, other things on the inside can relay with outh authenticating. when i look at the log on pbxnsip, i'm getting
[8] 2008/10/30 19:29:41: SMTP: Send EHLO localhost
[8] 2008/10/30 19:29:41: SMTP: Received 250-ex01.tek-hut.com Hello [172.16.1.4]
250-SIZE 10485760
250-PIPELINING
250-DSN
250-ENHANCEDSTATUSCODES
250-AUTH
250-8BITMIME
250-BINARYMIME
250-CHUNKING
250 XEXCH50
[8] 2008/10/30 19:29:41: SMTP: Send AUTH LOGIN
[8] 2008/10/30 19:29:41: SMTP: Received 504 5.7.4 Unrecognized authentication type
[5] 2008/10/30 19:29:41: SMTP Server returned 504
any ideas?? it looks like a setting on the pbxnsip side.
Do you have control over the Exchange server? Maybe you can try turning TLS on, or check what authentication types you can turn on there. Last resort would be to just trust the IP address of the PBX and don't use a username/password at all.
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What estimation time do you think this will be ready? I don't need transcoding, but i do need high volume of media relaying.
And is it possible to I stick multiple NIC's in 1 box and have each domain relay out a given NIC/IP ?
Well, it will definively take time. There is no concrete plan available yet. In the meantime the workaround is to partition the userbase and run several processes on a multi-core. A server farm on a single chip.
MWI message waiting indicator clears on its own after upgrade to 3.0.1.3023
in Extension Setup
Posted
Is it really related to the MWI? Maybe it is something stupid like instable power? Kind of hard to believe SPA reboots on MWI events...