Vodia PBX
-
Posts
11,111 -
Joined
-
Last visited
Content Type
Profiles
Forums
Events
Posts posted by Vodia PBX
-
-
FYI:
After installing the latest versie (3.1.0.3043), the translation to the e.164 works!
When calling from a pstn line to my pbxnsip setup the call is translated to the e.164 format.
OCS is now able to do a reverse number lookup and showing the correct contact name.
I'm happy
Well, so we just have another reason to do this painful internal conversion into the global telephone number format!
-
Is there a fix for this???
Check this out: http://wiki.pbxnsip.com/index.php/Global_Configuration_File
-
Hi i have a client who was just setup using a windows server and a PRI card/line...they are complaining of major call quality issues mainly on oubound calls...the people they are calling are saying they sound distant etc....please advise what steps i can take to resolve this!
Well, you can start with http://wiki.pbxnsip.com/index.php/Troubles..._Trunk_Problems, though this is primary for SIP trunks.
I would also check if you have a echo problem, a jitter/packets dropping problem or simply a volume problem. If internal calls are fine then there must be something with the PSTN termination/gateway.
-
On a windows system this is what i would do but because i am using a CS 410 in this case how would i do that??
You can use sftp (in most Linux systems) or psftp ir SecureSSH to load the files on the system. See http://wiki.pbxnsip.com/index.php/Installi...th_secure_shell on how to log in.
-
Is there anyway to upload a .wav file for an agent group instead of having to do *98 prompt? i have a client who has professionally recorded the prompts.
You can do a trick/workaround - just record it with *98 and then repace them on the file system. You usually can easily see by the timestamp what file you have to replace. Also you can quickly listen to them before replacing them.
-
this actually isnt a snom issue its a pbxnsip issue.
Right now my snom VM light is on. I just loaded up eyebeam to the same exension and there is also a VM light. So from eyebeam i called my VM and it says.. " 20 saved messages" I hang up. Now my snom and eyebeam VM light goes off.
So pbxnsip is not telling the phones there are no messages.
I have my VM forwarded to email and marked as READ. i assume this is the problem.
I checked again http://www.ietf.org/rfc/rfc3842.txt. IMHO the MWI bodies of the PBX are correct. If there is no new message, then the PBX will say "Messages-Waiting: no". That is the correct behavior.
-
We've got a customer asking about this, so now I'm asking you guys
The basically want to pick up the phone, enter a client code, then dial the number, and have the PBX record this somehow.
Is that how it works? What does that end up in the CDR as?
Yes it is in the works. The address book already has a CMC field, and the ACD and the hunt group are able to display the CMC on incoming calls.
And with snom phones, you can also enter the CMC code during the call. That CMC will show up in the CDR.
-
The Inband DTMF detection is on. Im still having the same problem.
Can you get an Ethereal trace? Then we can see what the problem is.
-
Outbound, after hitting the check on the Snom phone it takes at least 8 seconds till it connects / rings.
Yes, that is a "feature" - a few seconds until the FXO has dialtone and a few seconds to dial the ten or eleven DTMF tones. Faster has the risk of getting instable.
-
I have a new CS410 install, we are using the built-in gateway, there seems to be an 8 second delay until the call goes through. bypassing the CS410 call goes through right away, any suggestions? I already tried upgrade to 3.1.
Inbound or outbound? Inbound requires to listen to the 2nd ring for caller-ID detection; outbound required (slow) DTMF dialing.
SIP trunks are just "instantaneous" - caller-ID is immediately available.
-
huh? you can do dialog states with UDP? I have a few phones that work just fine, I dont like doing them on polycom phones however.
The problem is the initial message after subscribing for BLF. This message contains a list of the buttons, and depending on the length of the names this packet can easily get longer than 1492 bytes. If you just have 6 buttons, then that message size is usually enough.
-
The AA answers the call from the cell phone. When I try pressing "1" for accounting the AA does not recognize the "1". It just keeps playing the message.
Well, then you have a problem with the DTMF detection. Try turning the PBX inband detection on (System/Settings) and see if that helps.
-
It is interesting because it was through an ITSP, and it happened in the middle of a call. I would have expected it at the beginning. The other interesting thing is other people were on the phone, at the same time, and they did not have any call quality issues.
That's really strange. Maybe that call was technically never really in connected state? Or maybe there was a problem with a Re-INVITE.
-
May be this may seems a quite stupid question, but since it's our first installation, we'll take the risk. :-)
We have a dial plan like this:
110 Our_VoIP_Provider 8*
120 Our_PSTN__Gateway 9*
In the address book we have to put a 9 in front of the number to route the call to the PSTN Gwy.
This works, but callers won't be recognized when calling inbound, for the numbers do not clash.
The question arises since I've seen all dial plans examples less or more like our.
How do you conciliate address book numbers and dial plans?
Do we have to add another line in the Dial Plan for calls originating from speed dial codes?
e.g.
110 Our_VoIP_Provider 8*
120 Our_PSTN__Gateway 9*
130 Our_PSTN__Gateway *
Well, this is not a stupid question. The point about version 3.1 is that address book entries and also other "telephone" numbers are internally stored in the "+" format (e.g. +3912345678). Havnig 003912345678 as well as 012345678 is just creating such a big mess that you can never be sure what number you want to dial.
The "9" prefix that indicates an outside line becomes a little bit pointless in SIP. Like on the cell phone you have to push the "dial" button any way, especially in countries like Italy where you can't know how long a number is.
If you do want a "9" prefix, the phone has to deal with it and strip it. It is just to make you comfortable to hear a dial tone after entering a nine, but it has no more technical background. Psychology. IMHO we should just get over it and just dial the number that we want to talk to.
If you explicity want to route +-numbers in the dial plan then you have to use a pattern like "+*" or "+39*".
-
Once thing I am not happy about on the 820, they swithed the arond the check and the X box from right to left.
I think that's because every cell phone has it this way...
-
I had a end user that originally thought that they were dropping calls, however with some further investigation, I came to find out that if they wait long enough, the call would come back to life. During the dead period, I saw that the call state was listed as idle. What might cause this? It is so random, and infrequent, it might be very hard to duplicate, and/or catch in the act. I recently upgraded to 3.1.3044 and am not sure if this is related.
Idle is the state after creating the call object, but having no ringback yet. It is possible to stay in this state if the PSTN gateway does not signal a "ringing" yet. If you are using a ITSP and packets drop on the line, then it can be that the call stays "idle" for a relatively long time (BTW for that reason the PRACK SIP method was invented).
-
More issues ..
On the trunk settings, the NAPNA 10 digit, it totally ignores that as it is still sending +[CC][AC][NUMBER]
Also, when it saves the aliases, it stores them as +[CC][AC][NUMBER] even when entered as [AC][NUMBER] so on inbound the rule from 2.x, 3.0 does not work?
Hmm. That should work (we are using this rule in our Boston office). Can you give a concrete example?
-
Couple issues on test to this version.
All tel (10 digit) aliases had 011 appeneded to them, that is a lot of work to fix!
But then there must have been something wrong with these numbers? If you set the country code to "1" the PBX must think that this is a international number. Can you give an example?
Inbound calls on trunk, unchanged from 3.0.1 to 3.1 will not hit extension based on tel alias (account) as before .. for example we get a 10 digit DNIS and have the account set with that 10 digit as an account alias but it does not match as expected and goes to default (AA)?Well, the point of all these canonicalizations is to solve this problem. There is a conversion rule in the trunk now that you can set. If you choose the 10/11-digit rule then the PBX will assume the trunk is in NAPNA and then automatically convert numbers into the +1(xxx)xxx-xxxx format.
-
i have a client with a polycom 650 with two sidecars attached. they are set up using PNP and i entered in the extension for that extension to monitor the calls of multiple extension numbers but for some reason they will not show up on her screen or sidecars? i have tried rebooting the phone numerous times and clearing registrations but that does not seem to do it...please advise?
If you are using UDP for sending SIP packets the phones cannot receive the packets because of UDP fragmentation. You have to switch to TCP transport layer (or TLS, if you have a certificate that the phones accept).
-
PBXNSIP already has that feature available.
Dial into your Mailbox and enter your password.
Press 9 and it will play back to you all the greetings that were saved/used before.
You can choose which one you want to use.
I hope this is the answer you were looking for.
Addition: You can use *98*1 to record the first message, *98*2 for the second and so on until *98*5. That was introduced in 3.1.
-
i am using a linksys RV082 this should work no problem if setup correctly right?
can you advise which settings need to be changed?
I have no experience with RV082. Check out http://wiki.pbxnsip.com/index.php/Registration_Duration if that helps.
-
well make adhoc a separate key so people who can sell it can sell a snom phone without making excuses why it costs so much for a simple feature most phone systems have built in.
Done.
I notice the snom 820 doesnt have a record key.Also snom is learning...
-
Can you tell me please if the issue with Agent Queue Statistics has been corrected in this release? This is the issue where there are no statistics are collected if the domain is in a different time zone that the server. This was broken when 3.x was introduced.
You mean http://forum.pbxnsip.com/index.php?showtopic=1312? 3.1 is the "next build"...
-
Waht is the max numbers of entries in one dial plan? I need define up 300, will be this possible?
There is no limit.
Though 300 sounds like a lot to me. Usually it is possible to use expressions to make the list shorter.
Polycom IP7000-wont register
in Polycom Phones
Posted
What is the "Generate passwords" (in admin/settings/ports)? Maybe the phone cannot pull down the configuration? Do you see and TFTP requests in the log file of the PBX?