Vodia PBX
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Posts posted by Vodia PBX
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Yes that is correct. EIther method will do fine.
Well, the list of agents is stored in the agents XML. The DND state is stored in the extension XML. There are a few scripts available at http://wiki.pbxnsip.com/index.php/Shell_Scripts; it is for Linux but you should get the picture and this can also be done from Visual Basic.
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Now that an extension can log in and out of individual ACD Goups
I was wondering where I could look, programmatically, to see what an extensions staus is in a particulat ACD Group.
In the Database I can see if the extension is logged in or out, but that is Global from *64 and *65 Feature Codes.
Programmatically: You mean, for example, by looking to the file system or using SOAP requests?
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Which codecs do you recommend?
I would just take 0 (ulaw).
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Well, Ill try to explain more in detail. Our trunk provider has registered numbers in different countries in europe which are forwarded to our support line. All numbers except those for UK, Norway and Finland work fine. The provider has tried forwarding directly to one of their phones and it works. When those numbers are forwarded to our support line (an IVR) all we here is silence. The call is connected and I can see it in "calls" but we can't hear the IVR.
Denmark works Finland doesn't.
DENMARK (45) NATIONAL 1 +4569918175
FINLAND (358) HELSINKI (9) 1 +358942419025
Try forcing a specific codec on the trunk. Probably the provider has a problem when the PBX answeres with more than one codec (see http://wiki.pbxnsip.com/index.php/One-way_Audio).
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Sure. Here you go.
http://www.counterpath.com/assets/files/19..._User_Guide.pdf
Well, it was more about the XCAP specification and examples. Anyway, I guess we can also get those documents.
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I have now upgraded to the latest version but we still have the same problem with numbers forwarded from Norway and UK.
Any thoughts? All help is very appreciated.
Looking at the old messages I must admit I don't exactly get what the problem is. Is it a problem related to RTP or is it a problem related to the phone number (+47 or 0047 or 01147 and so on)? Maybe you can get a fresh LOG...
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No .... RHEL5
/etc/redhat-release
Red Hat Enterprise Linux Server release 5 (Tikanga)
I use your RH4 build. Should I be using something else?
Try ths: http://pbxnsip.com/protect/pbxctrl-rhes4-3.2.0.3132.
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We have a somewhat unique situation in which some options on our auto attendant should transfer a caller directly to a user's voicemail box rather than ringing their extension. When using the internal voicemail on pbxnsip, all we had to do was have those options transfer to 8xxxx where xxxx represents the user's extension. When using Exchange integration, the wiki states to ensure that the direct to voicemail prefix is removed (I tried putting it back in and using it just to see if it would work. It didn't.). How can I accomplish a direct transfer to a user's mailbox from an auto attendant when the mailbox is hosted on Exchange UM?
Transferring to 4xxxx doesn't work (where the external voicemail system is defined as 4$u, and 4* is setup on the dialplan to go through the exchange UM trunk), as it seems that that triggers the subscriber access greeting rather than the mailbox for the user. The only way I've found to make it work is rather a kludge. It involves creating a second extension for each of the users in question, associating that extension number as an additional EUM address on the user's Exchange account, and setting the extension to transfer to voicemail after one second in pbxnsip. Then, when I want to transfer directly to someone's voicemail from an auto attendant, rather than to their usual extension, I transfer the caller to this new extension. While this works, it requires an additional extension license to be used for each user that I want to implement direct to voicemail dialing for. Am I missing something obvious, and is there another way around this? Thanks in advance.
I guess we should just investigate why the auto attendant does not send the call to the Exchange...
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I was wondering if there are any plans to allow for customisation of the Message button behaviour through the PNP interface. Currently, the message button gets configured for one touch access to voicemail rather than allowing the display of the message summary. Second, the message button is coded in the phone config files to use "registration" to call back for voicemails. While this works fine when using the internal voicemail system, it completely breaks the message button when using an external voicemail system such as Exchange. With Exchange UM, it would be better to be able to set the behaviour to "contact", and the callback number to the Exchange Subscriber Access number. These are the only two things that are keeping me from using the PNP system for provisioning my Polycom phones, and having to fall back to using an FTP server to manually do the provisioning. Thanks in advance.
Kind regards,
Tom Schara
Enercia Ltd.
Good point.
The PBX sends the following content to the phone:
Messages-Waiting: yes Message-Account: sip:840@pbx.company.com Voice-Message: 1/0 (0/0)
The "Message-Account" tells the phone where to call, and that destination includes the "8" prefix that should send the call to Exchange. However, I just verified that the Polycom phone sends the call actually to sip:40@pbx.company.com.
This is a configuration problem with the phones. The msg.mwi.x.callBackMode is set to "register", we have to check if there is a way to make the phone the provided account in the MWI attachment. If we want to provide the "contact" for call back, we'll have to change the PBX code.
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Sanity Check and commentary. We should soon be installing a new client with a dedicated 10Mb Fiber SIP trunk to deliver inbound 800 calls to 6 or more agent groups. 1 ACD may receive calls from as many as 100 800 numbers. Using Aliases in the ACD seems to be our first option. We might also consider adding more SIP trunks and having group of numbers spread across more trunks. Then using ERE's in the trunks to deliver calls to extensions, we would eliminate the aliases..
Before make the final decision, we want to better understand limitations, benefits in functionality or reporting.
Using more SIP trunks would allow the system to scale as other servers can be installed on a departmental basis.
Managing all of the alias in ACD's is going to be a pain in the #$%#$ if you wish to move or add a 800 number, but I guess that be problemation in the ERE expressions too..
The total 800 number count is near 500 and likely spread across 6 acd's. Having import function on ALIAS using CSV files might help in the future. otherwise COPY PASTE...
Any Thoughts
Alias scale very well as the table in the PBX has a index (performance). Using the "trunk sends call to ..." option in the trunk is not an option if you want to enumerate all possible destinations there (a few ERE would be fine), the PBX has to step through that list for every incoming call.
I would get a good text editor to generate the list of alias names. Simple as it sounds, it can save you a lot of time editing the numbers and getting them into one line. "Emacs" is a science, but maybe there are also other editors that have some macro functionality that can automate editing.
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Importing accounts will need some additional functionality in the real world. With a soon to be real installation of 100+ accounts, we'll need to have these accounts members of predefined ACD and HUNT groups and many will have DID numbers.
We'd like to import from an Excel file extension accounts with first, last, ext, aliases, etc
ACD with aliases (as many as 100) 1st, 2nd, 3rd call stages, what extensions can join
HUNT GROUP members and call stages
What might make sense is the creation of a MASTER excel spreadsheet (Template) for use by all that can be imported by a dedicated function within PBXnSIP. We use a similar tool in our service desk software package. The developers created a series of EXCEL spreadsheets for project, client info, etc...
Designing how a PBX should work is far easier on a peice of paper and this is something the client can sign off on, and once completed any changes can be made as a change request. ($$) changes cost money.
Well, the bulk import feature does exist - check out http://wiki.pbxnsip.com/index.php/Creating_New_Accounts. You can use Excel to create the list. Before you can put that into the input form, you need to export it as CSV. Then you can easily copy and paste that into the form on the web. There is still some "manual" work required; however significantly less than punching the information in one by one. Plus you have the flexibility to import also settings like cell phone numbers, MAC addresses.
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OK - but where do I change this on a SNOM320 ??
Furthermore, it still do not make any sense that I need to do following to get the company name "SUPERFOSS"
I press "7" "8" (both once) and then I have the Superfoss listings???
That make no sense to me???
Changing from 9 to 32 entries must be done on the PBX. We did that already for you, the next software version will show the first 32 matches.
Yes, to get SUPERFOSS you would
- press 7 (wait for the XML),
- then 8 (wait for the XML) and
- then if you still don't see SUPERFOSS press 7 (wait for the XML),
- then if you still don't see SUPERFOSS press 3 (wait for the XML),
- then if you still don't see SUPERFOSS press 7 (wait for the XML),
- then if you still don't see SUPERFOSS press 3 (wait for the XML),
- then if you still don't see SUPERFOSS press 6 (wait for the XML),
- and so on. At any time you can use the up/down key if you see SUPERFOSS and select the entry.
Maybe your cell phone has something called "T9" entry, that is similar. When using the XML on the PBX, you must wait for the answer before you can enter another digit; that is because of the link-nature of the XML documents.
- press 7 (wait for the XML),
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When i go to the Currently Active Calls i see a blank call which says it started at: 2106/02/06 01:28:16 and it doesnt have info for "from" "to" "state" but gives me the option to X the call
is this something someone has ever seen?
What version? Should be solved in 3.1.2.
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You need to turn the logging on ("Log SIP events" and "Log trunk events"). The SIP messages are good, but we need to understand what the PBX makes out of it. Set the log level to 8, then you will see additional messages that tell you why the PBX cannot find the destination of the call.
In the log you see what user name the PBX is looking for and what trunk was identified. Maybe you did not specify the outbound proxy of the trunk or the extension does not exist. With the right logging, such an issue can be resolved quickly.
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I will look into the call issue, however in terms of the RFC it states the following note the " " around the name, however when we get calls from your switch the " " are missing and thus when a charter other then a-z is used it causes the system to break.
I don't understand... The From header that I see is From: "W: Office 1" <sip:9784256666@companya.mydomain.com;user=phone>;tag=648644319, with nice quotes around the "W: Office 1". Maybe they can just take a look at this forum post.
If the quotes are missing when it reaches their system, maybe there is a SIP-aware firewall in the middle that takes the quotes out?
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I am confused....!!
BUT when I press one time at "7" I should be able to see use arrow down to search through PQRS - is that correct??
When I press directory and then "P" I use arrow down to see the list - but only 9 listings are showed......
That can not be correct??
There are only 9 listings to limit the amound of data being transferred. It was programmed this way, there must be a limit somewhere. There are customers who have 45000 address book entries, and you don't want to load them all into the phone when pressing a button. Maybe should increase that a little. I believe Cisco can display up to 32 entries, that is probably a better limit.
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I would also like to see two changes to the Forum Display.
1. A new Column Showing a Checkmark or something to indicate the issue has been solved
2. Changing the Name of the subject to better represent what the problem was. To often the subject line does not reflect what the final problem was found to be, and the content of the thread has a lot of good tips for all, but you wouldn't know from reading the subject nor do you know it was resolved.
We are trying to move topics away from the "General Setup" into the forum where it fits best. I am nore sure if the forum software allows such a checkmark, but it might be a good idea to mark the topic. "Close" the topic is probably not a good idea as then it becomes invisible [need to verify that]. But we can use a post icon, or maybe we just change the title and append something like [solved]. Apart from that, we believe that the search engines do a good job making the forum searchable.
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The active calls screen shows correctly but the SNMP value is completely off. The offset is about 20 calls now since last restart about a month ago. If this is in memory then I should be prepared for bad performance and a crash sooner or later.
Any fix for this?
Try upgrading to 3.1.2.3120. That might contain a fix.
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I have 1 trunk from vitelity.net with 2 DIDs 7036985000 & 7036986000. I set up 2 Auto Attendants 800 & 810.
What would be the correct syntax for the "Send call to extension" in the trunk setting to have DID 7036985000 go to AA 800 and DID 7036986000 go to AA 810.
Here's the INVITE packets
INVITE sip:7036985000@207.197.254.13 SIP/2.0
Via: SIP/2.0/UDP 64.2.142.13:5060;branch=z9hG4bK11ed6264;rport
From: "7036239450" <sip:7036239450@64.2.142.13>;tag=as73b20880
To: <sip:7036985000@207.197.254.13>
I am running pbxnsip 3.1.2.3120 (Win32)
Try "!7036986000!810! !7036985000!800!". "!7036986000!810! 800" is a little bit more elegant, as everything else but 7036986000 would land on the 800 account. If you have more DID you can extend the pattern like this:
"!7036986000!810! !7036986001!811! !7036986002!812! 800"
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If I need to see a adressbook member starting with "S" normally I would need to press 4 times on "7", - Correct??
Nonono. You cannot search for "S" only. You can only search for [PQRS] (on of these characters). Then when you have the XML result on the screen then you can narrow down the search for the next character.
It is hard to explain. Maybe we need a video...
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16 34 ms 33 ms 34 ms ix-8-0.core1.NTO-NewYork.as6453.net [216.6.82.42]
17 227 ms 228 ms 228 ms segment-124-7.sify.net [124.7.187.29]
Whow that is a very long delay. When you make a phone call echo will be very obvious and it will be difficult to have a natural conversation. Just a side note...
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3120 Release. Same issue as above DTMF that is transported in Out of band via RTP or Singaling does not work!!! Mediatrix PRI Gateway. Please fix and re-release!!
On the DTMF topic, we fixed a bug in the codec negotiation. Essentially the bug was that the PBX was offering the codec number of the foreign device, not it's own codec number. For example, when the PSTN gateway offers 96 for DTMF, the PBX would answer also with 96. However internally it uses 101. In the fix, it now answers with 101 (which is correct). The problem because obvious for outbound calls when the PBX has to generate a SDP offer. When the PBX offers 101 and the other side answers with 96, then the PBX has to send codec 96, and receive codec 101.
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Do you already have an solution ?
We tried, and I learned a lot Dutch! The phone shows all menus and also the web interface in Dutch. The XML content contains no visible langauge-dependent content (at least I did not see anything), IMHO the behavior is okay.
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With an inbound call agent group set to record all incoming calls answers is their any reason that we cannot have a process scheduled to copy all of the recordings from this folder immediately?
I see that Ad-HOC recordings are listed in the extension WEB access portal. Is this generated on the fly by reading the available ad-hoc recordings or does a database maintain a list of files? A database would likely be harmed by copying these AD-Hoc recordings.
Does anything like this exist on the assigned recording options for EXTENSIONS, HUNT GROUPS or AGENT GROUPS?
The adhoc recordings are essentially mailbox messages. We assume that the number is "low", as much as the mailbox can store.
A simple way of solving that problem is to modify the path for the recording (record_location, see http://wiki.pbxnsip.com/index.php/Recording). If you include the name ($u) in the directory path, all recordings for that agent group will land in that directory.
ACD Log In / Log Out
in Agent Group Setup
Posted
Good point to put then at the front, Bill! We'll do that.
Pre-logout sounds too difficult to me. I guess you'd have to define a preference in a new setting for that.