Mitel Forums - The Unofficial Source
Mitel Forums => MiVoice Office 250/Mitel 5000 => Topic started by: pradeepgali on August 28, 2014, 06:21:46 PM
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Hi, I have Mitel 5000 Connected to Mitel 53XX phones. when i am in a call and if i get an another inbound call from PSTN to Mitel, i can see Call Key 2 is blinking and the PSTN caller is hearing Ring back. I would like to restrict the 2nd call, so that the PSTN caller can hear the busy tone. I tried to make changes in program keys and this doesn't work. Can anyone please help me with this?
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What type of trunks do you have? There is an option on ISDN for busy instead of camp on. The flag is enabled per station.
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I am using SIP Trunks. I have tried enabling "Camp-Ons Allowed"= NO and Yes. No change in behavior. I am actually trying to see two behaviors.
1. Busying the 2nd call from PSTN by sending busy tone to PSTN(Disable call waiting)
2. Restrict maximum number of calls on trunk basis(I can see this option in SIP Trunk Groups, but there is no option to change).
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Sorry, maybe I'm missing something here, but if you have 2 lines of your prime dn on your phone, the very existence of the second line means the system won't return a busy signal if you're on the first. I would think you would have to eliminate the second line of your prime in order for the system to return busy.
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I could not figure-out how to eliminate the 2nd line(CAll Key) on the phone(They are non-programmable keys). So that 2nd phone call coming in will not get ringback(Should get 486 Busy here from Mitel).
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I could not figure-out how to eliminate the 2nd line(CAll Key) on the phone(They are non-programmable keys). So that 2nd phone call coming in will not get ringback(Should get 486 Busy here from Mitel).
They should be. You would probably need to modify your Key Map to remove the second call Key, though.
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If i press Program Key 397 and select the call key 1 or call key 2, it say's Non-Programmable Keys .
Or if there is any way to disable call waiting, Let me know. I will give a shot and see how it works.
Please find the attached phone image.
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I don't believe you'll be able to remove them using the FAC 397, you'd have to go into SysDiag and modify the key map that's assigned to your phone.
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How can i do this? I could not figure-out how to get into sysdiag. Could you please navigate me?
Thanks
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Each phone has a Key Template assigned to it, it is a shared template that can be assigned to one or many (sometimes all) phones. The best way is to create a new template and assign it to the phones. For more information look in the DB Studio at System->Phone-Related Information->Key Assignments for more information.
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I changed all the keys to unassigned key. But still the display shows Call key 1 and Call Key 2 and some other keys as shown in figure. Because these are non-programmable keys(Keys cannot be modified). I just repeating what i am looking for: I am trying to disable call waiting(2nd call coming from PSTN to Mitel to the same phone should here busy tone).
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Pradeepgali, lets backup. Do you have access to the Mitel System Administration and Diagnostic software? The changes you have to make, will have to be done in there.
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Yes, I have access to Mitel System and Administrator. Please let me know what i need to change in this.
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Ok, you need to launch DB programming by clicking on the phone icon.
Then as acejaveling said, you'll need to browse to System->Phone-Related Information->Key Assignments
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I went to the navigation you mentioned. I have double clicked each and every key and saw the description of it. No key says" Call key1 or Call Key2". As i said these Call Keys are not programmable keys. I think they are hard coded keys. Looks like we could not delete these keys. We have to look for alternate on how to busy out the 2nd incoming call (EX: Disable call waiting or restrict MAX number of calls to 1)
Please find the attached image.
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They are not user programmable keys, so you're looking in the wrong folder.
What you need to do is first find what Key Map is assigned to your phone. To do this browse to System -< Devices and Feature Codes -> Phones -> 1004 -> Keymaps. See what the keymap is for Standartd IP/Digital Telephone Keymap Group. Go ahead and double click on the number so it will take you to the keymap.
You should now be back to System->Phone-Related Information->Key Assignments -> IP/Digital Telephone -> (Map #). At the very top line it should say Keymap Type in the left column and something in the right column. Double-click on the right column. You should now get a pop-up that shows you a picture of the phones, and then numbers to indicate each button. Do you see that?
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I went to navigation you mentioned and i am using default key map. I followed the procedure you sent and i have attached the popup image where i can see numbers assigned.
Please find the attached pictures.
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Since you're using the default keymap, you're going to want to create new one, and assign yourself to that. Otherwise any changes you make will effect everyone; unless your intent is to modify everyone.
Within the Default Keymap, there is a drop-down towards the bottom of the pop-up. From there you'd be able to select the default keymap for 5360 phones like yours. You could then remove the second call key.
If you need to create a new keymap, go up the folders till you're back at System -> Phone-Related Information -> Key Assignments -> IP/Digital Telephone. Right click in the gray space and Add to IP/Digital List. You'll have to choose an id number and then the keymap type. You can then modify this new keymap to fit your needs. After your done, you'll need to assign it to your phone as the primary keymap.
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It doesn't really matter, removing the Call key won't do what you want. Calls will still queue up, you just won't be able to see or answer them.
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I am able to change the keymap and delete call key.I learned something which i never did. But still when the mitel phone is busy in call, and if a call comes from PSTN, the PSTN caller is hearing waiting tone(Mitel is not sending 486 busy here message to PSTN on 2nd call).
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If this is not possible, is there any way we can restrict maximum number of calls on trunk basis?
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First off, I'm pretty sure eliminating the call key won't stop a second call...it just hangs out there in the ether, ringing your phone, until you do something about it. Regardless of how many call keys you have you can process unlimited calls. Instead, try the station flag (DID/E&M RECEIVE BUSY INSTEAD OF CAMP ON (navigate to SYSTEM/PHONE RELATED INFO/FLAGS and add your station to list).
Next, regarding changing keys you're in the wrong place entirely. The USER PROGRAMMABLE KEYS section is something fantastically difficult to understand...don't go ther, ignore it. Instead go to SYSTEM/PHONE RELATED INFO/KEY ASSIGNMENTS/IP DIGITAL TELEPHONE. Pick the map you belong to. If you're not sure, guess, and when you open it you can go to STANDARD LIST to see who's using it. Double-click the little icon left of the words KEYMAP TYPE. If you've done well you'll be presented with a picture of the phone. each key has a number, and a list to the right tells you what the key is.
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I have added the extensions i am using (1003 and 1004) to "DID/E&M Recieve Busy instead of Camp-On. Still the behavior is same.(I can see call key 2 is flashing with 2nd call and the PSTN Caller is hearing Call waiting Tone)
Please find the attached image.
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Did you change the system forwarding path to include when busy on each station?
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The forwarding path was added and the configuration is as shown in figure below. I tried station busy to YES and NO, there is no change in call behavior. Looks like i need to configure this in a proper way. I have also seen System Forwarding Paths (Navigate to System > Phone-Related Information > System Forwarding Paths). There is nothing configured. How should i configure this for my requirement?
Please find the images below.
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Pradeepgali,
First of all there isn't a way to provide the caller with a Busy Tone, on SIP Trunks if the DID/E&M Recieve Busy instead of Camp-On flag had no effect, unless all of your lines are busy. This is a PBX and its main purpose is to deliver phone calls to users. What you are asking a bunch of technicians and programmers whose sole purpose in life is to make sure those calls go somewhere to get answered is not a normal task and therefore not a simple one.
Secondly, the Camp-On flag that you changed under SIP Trunk Group make sure that is set to YES. This flag is for the devices on the system to be able to camp on to the Trunk Group and wait for a trunk to get out on if they are all busy, not the other way around.
Third, you are using the default keymap which tell most of the technicians/programmers here that your original installing company most likely left a lot of things defaulted, which leaves a lot to the imagination on how your system is really setup.
Now, What do you want to happen to these calls when you are "busy" with the call you are on? The options I will provide you will be nominal, but we may need to know how the calls are getting to you in the first place to find out what is really in charge of that call; my guess is a Hunt Group but I have been known to be wrong.
If you don't know how to answer that then go to System > SIP Peers > SIP Trunk Groups > 92002 > Trunk Group COnfiguration and look for the following two values.
Day Ring-In Type
Night Ring-In Type
Let us know what those are so we can direct you to the next stage and hopefully get this resolved for you.
Thanks,
TE
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Thanks for your support.
1. When call came from PSTN to Mitel user, PSTN user should know that the Callie is busy(with busy Tone). We consider this as part of user experience. I understand this may not be normal configuration.
2. Camp-On flag under SIP Trunk Group is set to "YES".
3. You are correct. Most of my configuration is default setup.
When Mitel phone is busy with incoming PSTN call and if the same phone gets another PSTN call, the 2nd caller should hear the busy tone(Mitel should send 486 Busy here to PSTN). This is not the hunt groups. This is an inbound call made from PSTN to IP Phone configured in 5000 with a DID number.
Day ring-in Type and Night Ring-In Type is configured to "Call Routing Table". I also attached the screenshot of this.
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I had to do this for an Advice service once.
You will need to create a Hunt Group for Every Extn.
Put your Extn in your new hunt group
Turn off Camp On for that hunt group
In the call routing table send your direct dial to the new Hunt Group
Second call should get busy, this certainly works on ISDN, never tested it on SIP.
Edit:
Just tried it on our SIP trunks and it works, second call gets busy tone.
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NTEDave,
It worked for me too. I am trying to do the same thing without using hunt groups.
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How are the calls getting to the desk? DID, live transfer or auto attendant? I have SIP trunks and just tested this scenario, set the DID/E&M RECEIVE BUSY INSTEAD OF CAMP-ON flag to YES and I think I got exactly what you're looking for: caller dials a DID, rings phone if available, returns busy if on a call. Won't work with transferred calls, but does work with direct dial. Historically we used this for faxes and modems, devices which can only handle one call at a time, but works for phones too.
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i have setup DID/E&M RECEIVE BUSY INSTEAD OF CAMP-ON flag to Yes for my phone, 2nd call doesn't send busy tone(I am hearing waiting tone on PSTN 2nd call). In Trunk Related information > Call Routing Table > 1, i have configured inbound routing. Please see the attached image for reference.
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By changing DID/E&M RECEIVE BUSY INSTEAD OF CAMP-ON flag to Yes 2nd call went to busy state by using 5224 IP phone. Before i used 5360 IP phone and the 2nd call doesn't busy out. Anyway this works for me.
Thanks everyone.
One more question. Is there any way we can restrict maximum number of calls on SIP Trunk? there is a field called "Maximum number of calls", but we could not edit it.
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Did you get rid of the system forwarding? I suspect it won't look busy if that alternate route is in place, because it's not really "busy" since it has a backup destination.
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I deleted System forwarding and now it works fine with 5360 phone too. Thanks
Could you please let me know if there is any way to restrict maximum number of calls on SIP Trunk?(Trunk based restriction)
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Sort of! You could remove some of your SIP trunks from the group BUT YOUR CARRIER WILL STILL KEEP SENDING CALLS. You need to ask the carrier to reduce your trunks. Then go into SYSTEM/DEVICES/SIP PEERS/SIP TRUNK GROUPS...pick your TG, got to TRUNK GROUP CONFIGURATION/TRUNKS and start removing.
I think the figure you've been looking at is in the CONFIGURATION part of the SIP TG, and that is just a count of how many trunks are in the list I described above.
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This sounds like the worst company in the world to try and contact! :) ;)
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Deleting Trunks work.
NTEDave,
We may or may or may not implement this. We just want to know if this feature can be supported by Mitel.
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Pradeepgali,
I got the feeling earlier that you are just doing integration testing to say that your equipment works with the Mitel 5000 or it does but only xyz features work with it; which is why your system is mostly defaulted. Now, with your comment on whether the Mitel can support the feature or not I am almost positive that is what is going on; who makes those kinds of changes to their equipment without knowing what they are doing unless they are testing integration; answer no one. So, most likely you are a SIP provider of some sort since you seem to know more about SIP than you do about the telephony equipment you are working with.
Of course there could be an alternate situation going on here, like you are just trying to learn the Mitel 5000 and how it works on a deeper level, but there are better ways to do that than playing around on a system testing possible scenarios with features of other vendors.
So, what other integration testing would you like to learn about with the Mitel 5000? I think it is a good idea to get this information out to as many people as possible, even if the scenarios are odd to say the least.
Thanks,
TE
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Tech Electronics,
What is you said is correct. I work for Service provider. As part of integration with Mitel 5000 with my provider, i am trying to see how these features work. Below are some of the things i am looking for.
1. Auto attendant: What call made from PSTN to Hunt group and if all the users are busy, the PSTN caller should hear the announcement.
Right now the behavior is PSTN caller is hearing call waiting tone.
2. Call Park: I think this feature is not supported by 5000. (Just want to make sure).
After i am done with this, i am planning to configure 5000 with T1/E1 trunks.
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Pradeepgali,
Not a problem and we can help make sure that you understand it well and make it easier for you to test the features you are looking for if it is possible.
So, here is how a Hunt Group works in the Mitel 5000 as I am not sure what you actual question is you may need to show a call flow of what you want.
Hunt Groups take calls and control them in order to deliver them to the members[UCD] or agents[ACD] of the Hunt Group. If you don't want it to allow people to camp on when all members or agents are busy then set the Camp On Allowed to No. If this flag is enabled, callers will be allowed to camp on to the hunt group when all members are busy. If this flag is disabled, the callers will hear busy tones when no members are available. Now, if you want them to hear what some call a RAD we have two options to work with; the Announcement and the Overflow. The Announcement will send the caller to whatever device you put in there, one time, but normally it is set to a Call Routing Announcement that times out to Hangup so it returns the caller back to the Hunt Group without losing their place in the queue. The Overflow works the same way, except it will repeat the process every time its timer expires. You could have both the Announcment and the Overflow set to go somewhere then the Overflow timer does not start until the Announcement timer expires and then it will repeat based upon its timer; so in otherwords there is a delay based on the Announcement timer prior to the first time the Overflow would intiate. The Recall timer is used to send the caller out of the Hunt Group to a device to be taken care of and this process releases the caller from the Hunt Groups control. Now, you also have three audio choices based upon how the caller is currently being handled by the Hunt Group if the Trunk Group is set for Use Next Audio Devices's Audio Source. At anytime during programming you can hit F1 to get a decent explanation of the features in that section of programming.
Call Park is not a built in feature like with some other systems, but it can be emulated by the system in a couple of different ways. Probably the most common way is to create a hunt group for each call park orbit you want and then put a key on the phones that will use it. The setup for this feature is rather simple, but it is not straight forward and I can go into detail on this if you need me to. The reason we don't really need the Call Park feature is that we have what is called Reverse Transfer [default FAC 4] which allows me to pull calls from any device that has an active call ringing, camped on, or on hold. So, basically we can do a transfer to hold to your phone and then page you to let you know and then you can dial 4 plus your extension number and retrieve that call from any other phone in the system.
Hopefully that answers your questions and if not then just let me know what you are looking for in a more detailed question and I or someone else will try to answer it.
Thanks,
TE
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Great,
I am much clear now on hunt Groups now.
Since i did not have an extension configured to Announcement (I also want to know how to configure this). I have changed the value of "Announcement" and "Overflow" fields to Voicemail extension(2500). For the purpose of testing i have added one extension(1005) in "Members".
I made a call to DID associated to huntgroup and the Extension added as member is ringing and the call was picked up.
I tried to make an another call which i am expecting to go to voicemail, but i am hearing call waiting tone.
I have attached the screenshot of huntgroup configuration. Please let me know if need to change anything.
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Sorry, forgot to attach the the huntgroup image.
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ANNOUNCEMENT plays while call is ringing at time set under timers, plays only once, and agent can pick up call during announcement. Same with OVERFLOW, except repeats every time timer expires. Both are designed to assure caller something is really going on, and can even tell caller how many callers are ahead of him and how long he might expect to wait. How calls are handled in hunt groups are highly variable and programmable...ring everybody at once? longest idle? circular hunt? balanced call count? Remember on this system things are designed to GET THE CALL ANSWERED, so it takes a bunch of engineering to not answer. Investigate the CAMP-ONS ALLOWED flag in the HG, which will return busy if no group member is available. Look at the RECALL destination which will send the call somewhere else when the timer expires (voice mail?).
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Pradeepgali,
Well making your announcement and Overflow the same thing is usually not done, but you can do it. Most of the time you want a different message playing from the first one than the one that repeats itself.
For example: Announcement would say, "were sorry but all of our operators are currently busy helping other callers, please stay on the line and our operators will be with your shortly."
The Overflow would say, "Unfortunately all of our operators are still busy helping other callers, your call is important to us so if you know your parties extension you may dial it now or wait on the line for the next available operator." This Call Routing announcement would have an option to allow the caller to dial an extension out of the Hunt Group.
Lastly you would, after a period of time, send the call to a mailbox using the Recall function, where you are sending it to a phone 1004.
So the way you have this setup is that the customer would hear, after the defaul timer of 18 seconds, You have reached the Mitel Voice Processing System, enter your extension now" or something along those lines. Then after another 72 and repeat every 72 seconds thereafter so you would hear the same message again due to the Overflow Timer and destination. You currently have the system to Recall out, but the default timer is 180 seconds, which is a long time to wait for it to ring extension 1004. Since you have the Camp-Ons Allowed set to Yes then the caller will camp on to wait for the next available member, as it wants to make sure the call is answered. Try setting the Camp-On Allowed to No and it should go to Ext 1004 and ring there until its forward initiate timer, default 15 seconds, is up and sends it to whereever you have its system forwarding set for, by default nothing so it may remain there indefinitely.
To create a standard Announcement and Overflow Message Center you could go to Voice Processor > Devices > Applications and right-click in the open area to the right of all of the other applications and then select, Create a Call Routing Announcement. By default all CRAs start with recording 001 in their Day and Night Greetings which says, "You have reached the Mitel Voice Processing System". Also by default the Digit Translations have nothing in them and they timeout to hangup. So to make an announcement you would change the Day and Night Greetings to another Audiotex Recording that is not being used. I normally set this to be the last two digits of the CRA. So if my CRA was 2505 then my recording number would be 005 for both day and night greetings. To record those you have to have an Administrator Mailbox set up in the system. So just create a standard Unassociated Mailbox and then go to Voice Processor and put that extension in where it says System Administrator Mailbox. If there is one already there then you are good to go and just need to know the password to get into it. It depends if your database was made local or on a live system as to what the password may be, but try the same number as the mailbox or just hit the # key and see if either of those works. If it doesn't then go into the mailbox and change the password. After you get into the mailbox by dialing the extension for Voice Mail and then pressing * it tells you to put in your password. If this is a new mailbox it will walk you through the setup process. Once you are in the mailbox it will tell you how full the hard drive is and at that point you can dial 9,3,1 and it will prompt you to put in your recording number, use all three digits. In my example of 005 you would put in 005 and it will then want you to record something and press # to stop recording and # again to save it.
Once you have your announcement and overflow messages setup you can then go back to your Hunt Group and put the extension numbers for those CRAs in there. I suggest setting your timers down lower for testing purposes. Keep in mind that every 6 seconds is considered one ring. If you create an Associated Mailbox for the Hunt Group and then set the Recall to be the Voice Mail application then it will go to that mailbox to take a message.
Hopefully that is all clear as mud to you.
Thanks,
TE
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i am able to make the changes you suggested below. Now i can hear the announcement as soon as the call made from PSTN to Mitel if i change the "Announcement" Timer from 18 to 1. If this timer is set to 18, i can hear waiting tone till 18 seconds and then i can hear the announcement. I am fine with changing this timer, but my concerns are below.
1. As soon as the PSTN caller hear the announcement, the call was answered by Mitel(Mitel sends 200 OK message to PSTN). Is there any way we send announcement as early media to PSTN?(Just what to know the possibilities).
I have attached the screenshots of hunt groups. Correct me if i have configured something wrong.
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Pradeepgali,
Alright, so maybe we need to explain as to what is happening here with this scenario and how Early Media works with the system. So, if you remember I explained Announcements and Overflows, when populated with a CRA, as being like a RAD [Recorded Audio Device] in that it answers the call and plays the message and then hangs up. The Hunt Group does a little magic in that it doesn't really release the call to the Voice Processor when it is an Announcement or Overflow so when the CRA hangs up the audio from the Hunt Group is returned based on whether the call is ringing, camped on, or transferring. The other part you may be missing is that even though you see that you can program the system and voice processor from one interface you need to consider them as being two separate devices that have limited communication between them on the state of calls; ie the Voice Processor doesn't care to know the state of the call it just does what it is programmed to do. So, hopefully that makes more sense as to why you got a 200 OK message versus a 183 Early Media message from the system.
Now, the system does support Early Media, if you have it turned on, and by default it is turned on and you can see that by going to System > SIP Peers > SIP Trunk Groups > {Trunk Group} > Configuration > Early Media Supported [YES|NO]. To see this flag you have to have Online Monitor turned on under the View.
So, to me Early Media would more like Music on Hold being sent to the caller prior to the call actually being considered answered by the system. I have a document that I wrote somewhere that explains when the system determines the call to be answered, but I haven't seen it in a long time so it may take a day or two for me to find it. Anyway, if you turn on Music on Hold through your Trunk Group Configuration you should see the 183 message as it plays the Music on Hold to the listener instead of sending 180 Ringing. If you change the Trunk Group Configuration option for Music on Hold to something providing audio you should hear it. You could also change some of the audio options to Use Next Devices Audio, which means it would then look at the Hunt Group and Extension's settings for audio in certain conditions.
Hopefully that answers you question about Early Media on the Mitel 5000 as that is the most I have had to deal with it up to this point. Maybe someone else in the forums would have a better insight into this than I do and can expand on the answer.
Thanks,
TE