3 way call between 2 extensions

I am new to iPhone world. I often call my colleugeat their extensions and need to conference them. I have stored their extentions with a pause (,) in the contacts.
I can the first contact with out any trouble. But when I call the second contact it asks me to dial in the extension. I never had that problem in Android.
Contact 1 - 999 999 9999,111
Contact 2 - 999 999 9999,222
Call contact 1, iphone dials 111. now I put contact 1 on hold and calls contact 2, the call gets struck and asks me to enter the extn which is 222.
Does anyone have any solution for this? I tried using a hard pause (;) instead of ',' and it did not work either.

Simply put, it works

Similar Messages

3 way calling question - NOT SOLVED

I asked this question:
I know how to perform a 3 way call, but what if the 2nd person I called doesn't answer and I get voice mail? I don't want to merge that call.
How do I hang up on the 2nd call placed without hanging up on the original?
And recieved this response:
Three-Way Calling
Merge two calls to set up a three-way call between you and two other parties.
Setting up a Multi-Party Call
1. Establish a call, either by dialing or answering an incoming call.
2. Touch Add Call. The first call is placed on hold, and the Dialer displays.
3. Enter the second phone number, then touch.
If the second person does not answer, press the Sendkey twice to end the connection and go back to the first person.
4. Touch Merge. The two calls are joined into one, 3-way call.
However, there is no Send key once you place a call...only END CALL.
PLEASE TRY AGAIN!!!

legg24 wrote:
I'm a power user in sales on the road all day and it seems EVERY smart phone i get forgets about this "feature" ... it's a necessity!!! And no tech support person ever knows the answer because they're techy and barely talk on the phone much less conference!
When i tested it just now (because i've been trying to figure this out for 3 days), instead of pressing Merge Calls, I hit the Back Arrow and was offered 2 options:
1) Return to Call in Progress (YES!!!!)
2) Access dial pad (not sure why I'd want to do this)
Hope this helps!
Maybe to try a different number (i.e. calling the cell phone instead of the office number you called originally)

Need a way to equally distribute calls between queues for unknown area codes

I have UCCX 8.0 SR2 installed where my script
is utilizing an XML file with 240 area codes. The script queries the Calling Number against the retrieved data from the returned data of the XPath. Therefore, in my script I have set Enterprise data for a variable called "Unknown" for calls that have not been defined by area code in that file which is then sent to a single queue. My customer does not have a the option to perform percent allocation in the cloud nor are they using Callmanager/gateway to perform this type of distribution since the analyzing of the area code is done at the script level.
I need to find a way to distribute these calls between two queues evenly within the scripting. For example for every odd numbered "Unknown" call the call is sent to the queue B and every even numbered call is sent to queue A. How can I script for this in UCCX 8?

What we did is to count the number of calls handled by these CSQs. Then do a MOD by 2 on this "count1" (1 to include the current call) to find if it is even or odd and then send to the respective CSQ.

How can I evenly split incoming calls between 2 employees without a shared voicemail?

I need to load balance a low volume of calls between 2 DNs.
Incoming calls need to be evenly split between Employee1 and Employee2. If they don’t answer the call, it needs to go to their personal voicemail, not a common voicemail account.
I think the Hunt pilot and line group is what I want, and I can utilize a circular distribution, but my only hurdle is voicemail. I can get voicemail to go to a single account, but I need a method to simply split incoming calls to 2 employees, and (if no answer), drop into their personal voicemail.
I cannot figure this out, any suggestions?
I’m running CUCM 9.02, Unity Connection 9.02, and UCCX v10 (I mention UCCX as a tool I could leverage, Employee1 and Employee2 are not agents or part of a call center)
Thanks for any assistance.

Queuing is not enabled. I have no desire to queue the calls.
In the Hunt Pilot settings:
When I chose “Do Not Forward Unanswered Calls”, calls drop after (what appears to be) the Line Group’s RNA Reversion Timeout is met. Never goes to voicemail.
The same when I select “Use Forward Settings of Line Group Member”.
If I select “Forward Unanswered Calls to” and manually select my VM pilot DN, missed calls finally go to Unity Connection, but not to any voicemail account. If I add the Hunt Pilot number as an alternate extension to Employee1’s unity account, then all unanswered calls will drop into Employee1’s personal account, even the calls missed by Employee2. This is not the desired solution. I need Employee2 ‘s voicemail account to answer the calls missed by Employee2 as well as Employee1’s voicemail account to answer the calls missed by Employee1.
This brings me back to my original question. Is there a way to get this to work with Hunt pilots and line groups, or should I be trying something else? I just need calls to DN:111 split 50/50 between 2 DNs (DN for Employee1, and DN for Employee2), and have missed calls drop into the personal voicemail of the DN that missed the call.
I’m hoping someone who has gotten this to work can share their Hunt Pilot and Line Group settings.
Thanks again.

Optimize the performance of the RFC call between ECC and CRM

Hi,
We are planning to extract sales orders, sales activites and service orders to dispaly it on the PDF factsheet of the account.
As of now, the PDF factsheet takes a long time to retrieve the data from ECC to CRM. Can you please suggest us on ways to optimize the performance of the RFC call between ECC and CRM.
Thanks in advance,
Vamsi.

Hello,
[SAP Note 636906 |https://service.sap.com/sap/support/notes/636906]is quite useful here.
Many times, the performance is poor due to function module CRM_CCKPT_EXPORTSUMMARY. This function module gets the customer number, the sales organization and the fact sheet view. If in CRM customizing, you use complete view (001), then all the views in ERP including all the info blocks will be retrieved, which will cause performance issue.
To solve the issue, please use a limited view to retrieve the data from ERP - especially a view, which does not contain info block 013.
Hope it helps
Joaquin

Can Firefox Hello be used for three or more way calling or is it only for person to person?

Can Firefox Hello be used for three or more way calling or is it only for person to person?

hello again, i've asked in the development channel for firefox hello about this issue: multiparty video-conferencing is on the long-term roadmap but it might still take a lot of engineering (firefox hello is based on the webrtc standard which is a peer-to-peer protocol between two machines only - multiparty calls might be a rather tricky thing to figure out on this basis)

Is it possible to pick up an incoming call from an extension that is not physically ringing?

Is there a way to pick up an incoming call from an extension that is not physically ringing?
For example if an incoming call was ringing at EXT 1019 and they were away from their desk momentarily, could the call be picked up from another extension by pressing a number sequence or something?
we are using UC560. if you could tell in detail how it can be setup(config) and how it will work.Thank you in advance.

Please rate helpful posts.
Thanks,
Alex
Here is the inof for the Call Manager Express.
Enabling Call Pickup
To enable Call Pickup features on SCCP or SIP phones, perform the following steps.
Prerequisites
•SIP phones require Cisco Unified CME 7.1 or a later version.
•The PickUp and GPickUp soft keys display by default on supported SCCP and SIP phones. If previously disabled, you must enable these soft keys with the softkeys idle command.
Restrictions
•SIP phones that do not support the PickUp and GpickUp soft keys must use feature access codes (FACs) to access these features.
•Different directory numbers with the same extension number must have the same Pickup configuration.
•A directory number can be assigned to only one pickup group.
•Pickup group numbers can vary in length, but must have unique leading digits. For example, if you configure group number 17, you cannot also configure group number 177. Otherwise a pickup in group 17 is always triggered before the user can enter the final 7 for 177.
•Calls from H.323 trunks are not supported on SIP phones.
SUMMARY STEPS
1. enable
2. configure terminal
3. telephony-service
4. service directed-pickup [gpickup]
5. fac {standard | custom pickup {direct | group | local} custom-fac}
6. exit
7. ephone-dn dn-tag [dual-line | octo-line]
or
voice register dn dn-tag
8. pickup-group group-number
9. pickup-call any-group
10. end
DETAILED STEPS
Command or Action
Purpose
Step 1
enable
Example:
Router> enable
Enables privileged EXEC mode.
•Enter your password if prompted.
Step 2
configure terminal
Example:
Router# configure terminal
Enters global configuration mode.
Step 3
telephony-service
Example:
Router(config)# telephony-service
Enters telephony-service configuration mode.
Step 4
service directed-pickup [gpickup]
Example:
Router(config-telephony)# service directed-pickup gpickup
Enables Directed Call Pickup and modifies the function of the GPickUp and PickUp soft keys.
•gpickup—(Optional) Enables using the GPickUp soft key to perform Directed Call Pickup on SCCP phones. This keyword is supported in Cisco Unified CME 7.1 and later versions.
•This command determines the specific soft keys used to access different Call Pickup features on SCCP and SIP phones. For a description, see the service directed-pickup command in the Cisco Unified CME Command Reference.
Step 5
fac {standard | custom pickup {direct | group | local} custom-fac}
Example:
Router(config-telephony)# fac custom pickup group #35
Enables standard FACs or creates a custom FAC or alias for Pickup features on SCCP and SIP phones.
•standard—Enables standard FACs for all phones. Standard FAC for Park Retrieval is **10.
•custom—Creates a custom FAC for a feature.
•custom-fac—User-defined code to dial using the keypad on an IP or analog phone. Custom FAC can be up to 256 characters and contain numbers 0 to 9 and * and #.
Step 6
exit
Example:
Router(config-telephony)# exit
Returns to privileged EXEC mode.
Step 7
ephone-dn dn-tag [dual-line | octo-line]
or
voice register dn dn-tag
Example:
Router(config)# ephone-dn 20 dual-line
or
Router(config)# voice register dn 20
Enters directory number configuration mode.
Step 8
pickup-group group-number
Example:
Router(config-ephone-dn)# pickup-group 30
or
Router(config-register-dn)# pickup-group 30
Creates a pickup group and assigns the directory number to the group.
•group-number—String of up to 32 characters. Group numbers can vary in length but must have unique leading digits. For example, if there is a group number 17, there cannot also be a group number 177.
•This command can also be configured in ephone-dn-template configuration mode and applied to one or more ephone-dns. The ephone-dn configuration has priority over the template configuration.
Step 9
pickup-call any-group
Example:
Router(config-ephone-dn)# pickup-call any-group
or
Router(config-register-dn)# pickup-call any-group
Enables a phone user to pickup ringing calls on any extension belonging to a pickup group by pressing the GPickUp soft key and asterisk (*).
•The ringing extension must be configured with a pickup group using the pickup-group command.
•If this command is not configured, the user can pickup calls in other groups by pressing the GPickUp soft key and dialing the pickup group number.
Step 10
end
Example:
Router(config-ephone-dn)# end
or
Router(config-register-dn)# end
Exits configuration mode.
Examples
The following example shows the Group Pickup and Local Group Pickup features enabled with the service directed-pickup gpickup command. Extension 1005 on phone 5 and extension 1006 on phone 6 are assigned to pickup group 1.
telephony-service
load 7960-7940 P00308000500
load E61 SCCP61.8-2-2SR2S
max-ephones 100
max-dn 240
ip source-address 15.7.0.1 port 2000
service directed-pickup gpickup
cnf-file location flash:
cnf-file perphone
voicemail 8900
max-conferences 8 gain -6
call-park system application
transfer-system full-consult
fac standard
create cnf-files version-stamp 7960 Sep 25 2007 21:25:47
ephone-dn 5
number 1005
pickup-group 1
ephone-dn 6
number 1006
pickup-group 1
ephone 5
mac-address 0001.2345.6789
type 7962
button 1:5
ephone 6
mac-address 000F.F758.E70E
type 7962
button 1:6

3 way call feature

I am not having a problem getting the 3 way call feature to work, I just can't drop the 3rd call like I should be able to according to tech support. You are supposed to be able to "tap conference" and then "tap the phone icon next to a call. The only place it says conference on my screen is across the top, and when you tap this and then tap end call you lose both calls. The directions make it sound like you should be able to see a list with both callers and tap the "end call" next to whichever call you want to drop. According to verizon this should work, but if does then I'm missing something.

Thank you kaebfly for your assistance. Adding to what was being said, the caller who starts the three way call is a sole connector of the other two other parties. So it bridges the communication for all. If you end the call, it ends for the other two parties. To maintain the communication with the second caller, wanting the third party to be disconnected, ask the third party to end the call on their end. Afterward, you will be on a normal call between two parties.
Thanks,
edw@vzwsupport
Follow us on Twitter www.twitter.com/vzwsupport

Unlimited Nights and Weekends and three-way calling questions

Sorry if this has been asked before but I couldn't find the answer searching the boards.
With Unlimited Nights and Weekends, if you initiate a call at 8:50pm on a weekday that lasts an hour, will you be charged minutes for the entire call (60 minutes) or only the first 11 minutes? On a related note, what if you make a three-way call at 8:50pm on a weekday? Are you charged the same number of minutes as a call to a single number? What about after 9:01pm? In other words, is there no difference between a call to a single number vs. a call to two numbers? BTW, all of these scenarios are for calls inside the United States.
Thank you!

wcaterino wrote:
Sorry if this has been asked before but I couldn't find the answer searching the boards.
With Unlimited Nights and Weekends, if you initiate a call at 8:50pm on a weekday that lasts an hour, will you be charged minutes for the entire call (60 minutes) or only the first 11 minutes? On a related note, what if you make a three-way call at 8:50pm on a weekday? Are you charged the same number of minutes as a call to a single number? What about after 9:01pm? In other words, is there no difference between a call to a single number vs. a call to two numbers? BTW, all of these scenarios are for calls inside the United States.
Thank you!
To answer the first part, you would be charged for the first 11 minutes because the Verizon network recognizes when the call rolls into a free minute period. As far as 3 way calling is concerned, you are actively calling 2 numbers so the minutes to both numbers count againist your minute plan and then would transition to free minutes after 9pm. Hope this answers your questions.

Evenly distribute calls between numbers?

(UCCX 8.5.1)
I'm trying to find a way to evenly distribute calls between multiple offices.
Example:
Call comes into the main number, checks Holiday/ToD then sends the call to the first number, sends the second call to the second number and repeats until the last number has been given a call. After the last number, the script sends call the next call to first number and so on.
Is there a way to count the calls and have the count direct the call and then reset after the last number has been reached?

The easiest method is not the most accurate method: random numbers.
Say for example you flipped a coin, it will land on heads 50% of the time and tails 50% of the time. Kind of.
So if you wanted an even distribution between two numbers, you could randomly flip a virtual coin, and then send the call to the "heads" number or the "tails" number. And this will get you in the neighborhood of a 50/50 split.
Take that idea to a 3 sided coin...err...let's drop the coin analogy for now, and just say three possible numbers: 1, 2, 3. Then randomly pick one. Well, statistically, you would pick each number about 33% of the time. Almost.
If that doesn't work for you and you need an actual circular routing algorithm, then you will need to cache the last number used, so that you can increment by 1, and move to the next number.
Without using files and or databases to do this, you could use a Global Session Object to store this cached information. Each and every call in the system can read/write to this Global Session Object, you just have to worry about race conditions, but honestly, since UCCX is not multithreaded, the chances of that happening are very slim, and the worst that can happen is you skip a number once out of every million calls. Ok, I made those numbers up, but I think my point is still valid.
Search for some of my posts on working with Sessions, then Global Sessions, and see if you can get something put together to get you started.
If you want to go the file or database route I can help there too.
Let me know how you want to proceed.
Anthony Holloway
Please use the star ratings to help drive great content to the top of searches.

H.264 doesn't seem to work... until I initiate a 3-way call

Hello,
I seem to have a weird problem with iChat video conferencing. As the post subject indicates--I believe iChat doesn't automatically use H.264 (at least on my configuration) in two-way video calls but does it in multi-way calls, or more specifically after a multi-way call. I discovered it by:
1. initiating a normal 2-way call - video's rather blocky, bandwidth use ~200 both ways
2. adding a 3rd person to the call - video even blockier, still same bandwidth usage
3. removing the 3rd person from the call - video suddenly gets much better (compared to the initial 2-way call, not only to the 3-way call!) with the same bandwidth usage - which makes me think it's H.264 that suddenly gets activated.
I get this with all of my iChat contacts (obviously I had them all check, I was just stunned by the sudden quality increase I got). I'm on a 2.16GHz Core 2 Duo MacBook Pro so it's not a CPU speed issue. The only thing I can think of is that it might be due to the fact that I had migrated my system using Migration Assistant from a 12" PowerBook which might not have been H.264 capable (?) - at any rate I tried trashing iChat's .plists but to no avail.
I'd be very grateful for any ideas...

OK here's what the logs say--not surprisingly, it seems like the MacBook Pro is H.264 capable. What puzzles me is why iChat is looking into a "G4" file but I guess it doesn't matter much if it still finds it OK. I also tried starting a conference etc and having a look at the logs but the sheer amount of data spat out by iChat is rather daunting--is there anything specific I should be looking for?
200048.274556 Loading hardware rules from hardwarerulesH264-G4.plist
200048.274665 Checking H264 hardware rules
200048.274715 Satisfied rule #1 (20:640x480)
200048.280877 Default video size is (640x480)
200048.288558 ### NEW STATE: to: VC_INIT, from: VC_INVALID
200048.289254 ### NEW STATE: to: VC_IDLE, from: VC_INIT
200048.289303 ++++++++ confStatusMapRemoveAll
200048.289362 ( ) -- mapUserIDToSecurityInfo cleared --
200048.289939 UPnP init passed
200048.290123 NATTraversal created
200048.290233 QT automatic setting = 16777216
200048.290291 Bandwidth based on QT prefs = 0
200048.321291 BWD: GotBandwidth: 244585/6555894 (up/down) @[192.168.1.168:255.255.255.0]
200048.321325 BWD bandwidth: 244 kbits up, 6555 kbits down.
200048.321341 videoNetworkCapable: 1 -- 2 max participants.
200048.321349 audioNetworkCapable: 1 -- 9 max participants.
200048.321434 vcCapabilities returns: 0x0000003f
200048.321476 VCCapAudio
200048.321503 VCCapVideo
200048.321526 VCCapMultiAudio
200048.321552 VCCapMultiVideo
200048.321575 VCCapMultiAudioHost
200048.321614 VCCapMultiVideoHost
Thanks very much guys,
-e.

Why call between WebClient agents fails?

Hi,
I have just recently installed the mySAP CRM 4.0 IC WebClient (CRM40 SP00).
I have been testing the installation with CCS (Contact Center Simulator) and for a basic CCS supported tests (Inbound Phone call, Outbound Phone call, Chat) everything seems to be ok, but a call between WebClient agents fails due to an java error. The Phone Call between WebClients which should work according the CCS webpage.
What should I do to get WebClient agents to call each others? Any configuration or customization tips? Note that I do not have CC integration in place, just the IC WebClients.
Here is the test done in details and the error reported by WebClient:
0) I restarted SAP CRM to be sure no pending sessions exists
1) Open the CCS to see that no agents are logged in with WebClient
2) Login as java_ic with WebClient
3) CCS shows java_ic Workmode (=status) as 'Not Ready'
4) Switch java_ic to ready by from the WebClient
5) CCS shows java_ic Workmode (=status) as 'Ready'
Then I do the same for agent 'sap*' with another MSIE browser window. At this point, the CCS shows Workmode for both agents as 'Ready' and I have two WebClient browsers opened. I also verify that both has Phone selected from the channel selector in the WebClient.
6) With java_ic agent, I press the Dial Pad, enter the sap* agent phone
number (Phone: +15550002; shown in the CCS for both agents logged in) and then I press the Dial button on Dial Pad to call to +15550002.
6.1) After pressing the Dial button, the WebClient displays a pop-up in
the middle of the screen with info: "Your request is being processed." After this there is an Info message under the buttons in the WebClient for java_ic saying: "No account information found".
7) Java_ic WebClient shows that it's doing an outbound call and is 'Dialing' state.
8) sap* WebClient window has blinking button with labels 'Accept' and 'Reject'. So, it clearly got an notification of the Inbound Phone call coming from java_ic. But there is also an error message under the buttons in the same WebClient webbrowser screen (not in the JavaConsole, not in separate alert message this time) for sap*:
"Java error: type= APPLICATION ERROR, ID= GETIAD_FAILED, description=".
9) With sap* WebClient I cannot answer or reject the call. Nothing happens. The browser Status Bar says: javascript:void(0) when I press the Accept button.
10) At this situation, the CCS shows that the agents's sap* Workmode is
switched from the previous 'Ready' state to 'Not Ready' automatically (while the sap* WebClient status is still in Ready). So, the sap* automatically, due to the error, changed the status to 'Not Ready'.
11) java_ic remain on 'Dialing' status for all the time trying to get connected with sap*.
I got informed that the call between WebClient COULD BE seen as a "call transfer" between clients by the CRM system. This might be something to do with the problem in calling between WebClients.
Because the java_ic reports: "No account information found" when placing the Outbound Phone call, one could assume that account details should be passed along with the call transfer. Possibly sap* cannot receive the
inbound phone call due to a missing account information thus reporting java error: "Java error: type= APPLICATION ERROR, ID= GETIAD_FAILED, description=".
Best Regards,
Jani

[loaded] com.genieo.completer.download.plist Adware! [Remove]
[loaded] com.genieo.completer.ltvbit.plist [Support]
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[loaded] com.macpaw.CleanMyMac2Helper.diskSpaceWatcher.plist [Support]
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Your machine is slow because you loaded a bunch of third party junk on it.

Three way calling..... Problems....!!!

The problem is when you merge 2 parties into one; it is either you hear nothing at all from both parties, or you only can talk to 1 person putting the other person on hold. Simply, "merge" just does not work.
I called at&t 10 times to find a solution, still no go.
Talking to a service representatives did not help much. What they do is that they just remove 3 way call feature and put it back on to your account from their at&t system, or they suggest power cycle the phone, or call apple.(duh!) At times, it worked momentarily then voila, the same problem continues.
Then I talked to a tech support, I do not know what it is called but the tech disconnected my iPhone 3g from the network and reconnected it. She said this process is different from partial remove/add features that customer support rep did, it was rather total reset from the network.
Right after the talk with tech support, I was able to connect & merge 3 people (my friends) together that I could not before. Next day, I tried three way calling for the same 3 people but it did not work ever since.
I test called three way call today, calling 611 and Apple iPhone support number 800-694-7400, and it works! However, problem still exists when I try to connect my friends, the same 3 people mentioned above. What gives?
Does the problem have something to do with my friends' carriers? One has T-mo, the other one has Helio, and the third has vonage VOIP home service.
Please let me know.

bump

Three way calls issue

Hello,
I am having difficulty with my phone. First problem. I can make three way calls, but I can't end a call with one person without having to end the call with both individuals. I was on a very important phone call today, long distance, and this really messed me up! Second, when someone calls me, and I put that person on hold to answer another incoming call, when I end the conversation with that incoming call I have to hang up on the other caller on hold!
I have Verizon. Does any one else have this problem??
Please help. It is so frustrating.

Verizon is a CDMA carrier, so your options are limited. This describes (from the iPhone User Manual):

Three way call

Is it possible to drop a caller during a three way call without having to as them to hang up on their own? Is this a limitation of VERIZON's service?

If YOU made the call you should be able to hang up. If someone else made the 3-way connection I wouldnt think so.

3 way call between 2 extensions

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