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--- RE: Calling standard extensions from digital receptionist (http://www1.pbxes.com/forum/threadid.php?threadid=1224161290)
Calling standard extensions from digital receptionist
Hi
I have set up a few standard extensions to go to people connected through a switchboard. The extensions seem to work fine if you put them in as follows -
<switchboard number><p><extension number>
eg. 1234567p345
However, if I put one of these standard extensions as an option on the digital receptionist, I get the message "you have dialed an incorrect extension". I have tried using w.s to put in gaps but it doesn't seem to matter how many I put in, it still won't work.
Any ideas?
Thanks
RE: Calling standard extensions from digital receptionist
Could you please describe a call scenario which would require this functionality?
RE: Calling standard extensions from digital receptionist
We have people who are in the same department - eg finance department - but are geographically separate - some use standard extensions on a standard switchboard and some use internet phones connected to pbxes. We have an incoming did number for accounts. At the moment the digital receptionist answers and announces each choice to the caller. If the caller chooses to speak to somebody who has an internet phone, then they are put straight through. If the caller chooses somebody who has an ordinary phone extension, then the caller is put through to the ordinary switchboard and has to choose again to finally speak to the person they want to speak to. We would like the whole thing to be seamless - not depending on whether or not the callee is using an internet phone or an ordinary extension on a standard telephone switchboard.
Basically, across our organisation, we have a mixture of internet phones and standard extensions through standard switchboards. We would like all users to be seamlessly integrated - whether they have an internet phone through pbxes - or a standard extension on a switchboard. That way we can gradually expand our use of pbxes - rather than having to change every user at the same time. Indeed, for various reasons, it may never be appropriate to introduce internet phones everywhere - eg in some countries in which we operate, the internet is not entirely reliable. I agree it is possible to route through to a switchboard and then ask the caller to choose again. However, most people find the idea of having to make multiple choices in order to get through to the person they want to speak to, annoying.
I guess what is needed... and I don't know if this is possible... is for pbxes to recognise when a switchboard has been answered, and only then send the rest of the number dialed (ie the extension number). I thought that this is in effect what using the "w" character would do - by putting in a pause. Unfortunately this does not seem to work at all through digital receptionist and only sometimes in other cases..
RE: Calling standard extensions from digital receptionist
So what you are actually trying to achieve, is to bridge calls from DIDs to an IVR of a traditional PBX (a.k.a. PABX or switchboard) via the IVR (Digital Receptionist) & the extensions of PBXes.
How is an inbound call sent to the ordinary switchboard (PBX)? Is an ATA used connecting its' FXS port to an FXO port of the switchboard? Is an outbound call placed via trunk, to the PSTN number of the switchboard's IVR?
Please keep in mind, that in general hybrid systems are hardly seamless, unless there is a lot time and effort spent working through all possible scenarios.
RE: Calling standard extensions from digital receptionist
An outbound call placed via trunk, to the PSTN number of the switchboard's IVR. This bit is "seamless" - the caller doesn't know they are being transferred to a standard switchboard.
If somehow, pbxes could recognise when the call is answered (by the switchboard's IVR) and only then send the extension number, this would be great
I should emphasise however that we are not just trying to use pbxes to route calls to a did to a traditional switchboard. As I said earlier, we have different setups in different places - including in some places where the internet is not really an option. We are trying to integrate all of our communications - so that the standard switchboard extensions become "part" of our pbxes virtual switchboard.
RE: Calling standard extensions from digital receptionist
The code looks OK. I have not tested your setup. What you can do is:
RE: Calling standard extensions from digital receptionist
I took your advice. So set up extension 303 on dig receptionist as follows -
When the digital receptionist answers you type in 303. This routes to an inbound route <metokote-9303> which in turn is routed to a standard extension number <9303>. On extension <9303> the telephone number and extension are written in the format xxxxxxxxpxxx.
Amazingly this seems to work
Next question then is how to enable ordinary internet phone users(connected direct to pbxes) to dial extension 303 without having to go through the same digital receptionist. As far as I can see see you can't go straight to a sip uri from an extension.
My first attempt was to set up the extension number (303) as a ring group, put in a bogus number, and tell it to divert to the sip uri if not answered. This didn't work.
I then set the ring group to divert to a digital receptionist. This dig rep has only 2 options - t & i - both point to the sip uri. This works
The only problem is that there is quite a long delay before you are finally connected. Can this delay be shortened somehow?
This seems a very complicated way to get things to work from an internet phone. Is there an easier way?
Thanks
RE: Calling standard extensions from digital receptionist
Oh, I thought it would be working fine calling them directly.
Zitat:
The extensions seem to work fine if you put them in as follows -
<switchboard number><p><extension number>
Only calling thru Digital Receptionist would have been a problem which was solved by SIP URI routing.
So for calling directly leave everything as it was, just call 9303 in your example. You can also name this 303 instead of 9303 if you don't want to dial the 9.
RE: Calling standard extensions from digital receptionist
Calling just 9303 direct from a pbxes extension doesn't work consistently. On some extensions it works sometimes and on others it doesn't. The sip uri thing seems to work 100%.
When I said it worked in my first post, I later discovered it was only working consistently from that extension - not others. I guess it must be something to do with the speed with which it connects.
How does the sip uri connect differently?
RE: Possible DTMF issue
That indicates there is a DTMF transmission issue, with certain SIP UAs on some of your extensions. What kind of SIP User Agents do you use?
RE: Calling standard extensions from digital receptionist
we have some users with Linksys 941.s, some with Nokia E90 (using sip), some using softphone x-lite, and some using ordinary telephones and calling in through a did routed to the dig recept. I will go and do a test on each one and see what happens. Will let you know shortly.
ok - have done some tests now -
E90 (using sip) - doesn't work
Xlite - works
Linksys 941 - haven't got one with me to test at present but it did work with my initial tests a few days ago but then seemed not to the other day
Through ordinary DID using Dig Recep - we know that works thanks to your earlier solution
Using passthru - this works too
So it could be a problem just with E90 although I am not sure whether Linksys 941 works entirely consistently - can't test fully until monday.
My earlier solution works - only problem is complexity (having to set up a dummy ivr for each std switchboard extension) & delay before call goes through. Is there anyway to minimise delay? I could live with complexity
IAs far as the E90 is concerned, I checked the sip settings.dtmf inband and outbound are both set to "on". Interestingly if I dial <switchboard number><p><extension number> on the phone and make a sip call, it works. The only difference is that you hear a few secs of the beginning of the std switchboard announcement before it puts you through.
RE: DTMF issues
Regarding the E90 DTMF setting, get rid of the Inband DTMF. Leave only the OutOfBand setting and ensure it is set to RFC2833. No SIP Info either if you can remove that too.
In essence with both settings on, the SIP stack creates double or triple digits for every key pressed. So the PBXes DTMF detection routine has a hard time recognizing what key was pressed.
Personally I would also edit the "dtmfmode" in the extension setup to RFC2833 only, to avoid the double digit detection.
Regarding the SPA-9[46]x IP phones, there is known issue with double DTMF digit generation, which does not exist on the SPA ATAs, since they have an extra setting controlling DTMF generation in their web interface, which is curiously missing from the web interface of the SPA-9[46]x IP phones.
I have notified the program manager of the whole SPA IP phone line last April, but the bug has not been addressed yet, even on the latest 6.1.3(a) firmware version for the SPA-962 phone. I have followed up with him regarding this issue last week, so it can hopefully be included in the 6.1.3 beta version.
We will see if it is fixed permanently this time. I haven't found a reliable work-around so far.
RE: Calling standard extensions from digital receptionist
Thanks for the info.
I turned off the dtmf inbound on the Nokia E90. Don't know how to check whether outbound is set to RFC2833 and I don't know what you mean by no sip info - there doesn't appear to be a setting for that. I also changed the dtmfmode on the extension to RFC2833 only. Unfortunately this seems to have no effect - still doesn't work.
Given what you said about the bug on the Linksys SPA941, it looks like we are left with my work-around (using a "dummy" dig recep) - and will have to put up with the delay. I put a litle recording on it saying "please hold" - so hopefully people will hang around long enough for the extension to start ringing - it seems to take about 9 secs
- Is there any way in which the dig rec can choose the default options faster? (without the caller having to select anything).
Thanks for all your help so far.
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