MarkRussell

I am not sure what you're looking for. Obviously the hide number button on the Skin sets the system to send anonymous and is a toggle button, does your user require the same sort of thing but changing the action to 141 in which case this is not possible.

You can however use a button to send 141, which they have to press each time they dial a number. In which case program a speed dial button to be 9141 as below. (you will still need the number replacement set up)

So when the dial out instead of dialing 9 they press this button then the number. This of course wont help with click to dial.

If they want a toggle option for users, the only option I can thing of is to replace the trunk access code fro 9 to say 7. Then create a call routing user as extension 9. When a user dials out using 9 the script captures the number they dialled and prefixes it with 141 if a certain variable is set for that user. If not the call goes straight out. There are a couple of issues with this. 1. Call reporting won't be a lot of use. 2. All callers when they dial out will here 1 burst of ringing as the script answeres the call. Then the normal call progress tones will be heard. So if you call a number thats engaged for example, the callers will hear, 1 burst of ringing then an enganged tone. So this option is far from ideal, but I have used it a couple of times where the limitations were not as important as the flexibility.

Another option may be to set up a second sip trunk, and use number signaling in the second trunk to always present a certain number (ie a 141xxxxxxxxxxx number) 

Honestly there a so many ways of manipulating the presented number, the real question is what is the customer expectation of how to use the feature, then work back from there. Npt everything is eactly possible but you should be able to give the customer something that meets their needs with a little give an take.

OK, to do this you need to go into special number replacement in the trunk group profile, and use this replacement.

You say the problem is only inbound calls ? What type of trunk is it, Sip Authenticated or IP Authenticated ? Is the incoming call a ddi direct to a user or via a script ? what version are you running ? The fact that this is only incoming calls and not outgoing suggests a problem with the sip signalling of IP address, well that would account for no inbound audio, but no outbound adios is very strange. I would suggest looking at a wiresharke trace to get the info as to where the traffic is going.

Firstly I would ask why so many log files, do you have 100's of phones ? Have you checked the logs. A quick search for *warn will find basic errors that include a description of the problem, such as wrong pins, licenses etc. *fat or *err will also highlight more serious issues that need resolution and probably support from Swyx.

The documentation for tracing is included in the Swyx Troubleshooting course, and I am not sure I can publish it here. However it is possible to adjust the tracing output for each of the swyx modules using regedit.

Navigate to HKLM/SOFTWARE/WOW6432Node/Swyx/(module)/CurrentVersion/tracing

In there as you have seen are all the components of that module. The tracing level can be changed using these values

Each trace module has a level threshold associated which defines the amount of trace output generated. Below is a list of the decimal value of the registry entry and its description.

    •    Off            0    no output
    •    Fatal         1    fatal error messages
    •    Error         2    error messages
    •    Warning   3    warnings
    •    Info           4    informational output
    •    Info2        5    more informational output
    •    Info3        6    even more informational output
    •    Debug      7    debug messages
    •    Debug2    8    more debug output
    •    Debug3    9    even more debug output

A higher level includes all output of the lower levels. E.g. if a trace module is configured to Info2 every trace statement from Fatal to Info2 is included in trace output. Levels “Debug” and higher are only available within debug builds.

I would strongly recomend backing up the hive before making changes as if you have a problem you need support on they will need the original traces. Most of the keys are self explanitory as to their reference but I would also suggest booking the 1 day troubleshooting course if availabnle from your supplier.

You need to copy the preprossing .vbs file (the rse file is only used to create the vbs. it makes sense to keep both together though in the database so you cna then edit) To test this I would put it in a user scope first. If you put it in global then every call onthe system, internal and external would be effected.

With 11.50 you need to be using the push notification service, did you install this after the upgrade (its on the dvd) ? and of course the new mobile app.

Sorry to reply in english my german is VERY bad 🙂

OK Now I understand, or rather I understand your fault, I dont undertand why its happening. So a couple of things to look at I think.

1 in active calls when you dial the number can you see it going out on the VODA trunk group.

2 While the link manager trace files are usefull faults like this are easier to see in my opinion in wireshark. So if you can grab a trace and have a look at the sip flow it would be useful. Any problems you can always PM me if you dont want to put the pcap here.

I have thought of another possibly less technically complicated solution. Right a batch file to launch Swyx and replace the desktop icon for Swyxit with this batch file.

The batch file Runs SwyxIt and waits for it to log on.

Uses the CallTo or Tel ref to dial a script number with a post dial digit.  This script would know the users extension number and the digit.

The digit dialed would be the result of an ipaddress subnet query eg

(

for /F "tokens=14" %%i in ('"ipconfig | findstr IPv4"') do SET LOCAL_IP=%%i

)

With a simple rule the Status Text is updated based on the Post Dial Digit.

While I believe the powershell option is the best way, the only downside is the understanding of loading powershell modules. Too many times I have seen code written that loads the ippbxmodule many times and cosumes all the memory on the system.

Using a prespossing rule means easy deployment and just a simple file to be loaded onto the desktop. The code would be easily transportable to different systems and new locations could be easily added.

I am not sure I fully understand. But if you are call ing a number of a group directly then you will never get voicemail. This can only be done if you call a call routing user that has a rule to ring the group and if there is no answer or noone is logged on in that group then the call flow should go to voicemail.

Hope this helps.

An der Übersetzung ist Goolge Translate schuld... 🙂

Ich höre auf vielen Systemen viele Anfragen zur Zusammenarbeit mit Teams, aber wenig darüber, was erwartet wird.

Tom hat recht, ich denke, dies sind Anfragen für PM bei Swyx, die über Sales / Account Manager an sie gesendet werden sollten.

Ich halte es für sinnvoll, in der Anfrage mehr Details anzugeben als nur "Ich brauche die Integration in Teams". Einige davon sind meiner Meinung nach bereits mit einem SIP-Trunk zwischen den beiden Systemen möglich. Aber was sonst. Der Anwesenheitsstatus ist offensichtlich (in beide Richtungen oder in eine Richtung). Wie steht es mit der Zusammenarbeit? Kontakte. Was müsste ein "minimal lebensfähiges Produkt" beinhalten, um von Wert zu sein? Was verlangen Endbenutzer von einem einzelnen Kunden? Ist Teams Client bis zu einem Telefon-Client für die Telefonzentrale oder würde der Benutzer erwarten, zwei Clients zu verwenden, in welchem Fall was benötigt wird.

Es gibt viel zu überlegen, und für einige davon ist möglicherweise keine Swyx-Entwicklung erforderlich. Kluge Personen wie die in den Foren können möglicherweise MVP produzieren, wenn bekannt ist, worum es sich handelt.

Dies ist nur ein persönlicher Gedanke und spiegelt nicht die Meinung einer anderen Person oder eines anderen Unternehmens wider.

My origninal script had post dial digits but I removed to simplify. But this (using PDD) is really a great way of doing this.

I am at a trade show in th UK for the next couple of days but the video should be online late thursday night..... I hope :-)