route phone extension call to In group

[rrb555]

Hi,

Vicibox 4.0.2 installer
VERSION: 2.6-395a
BUILD: 130221-1736
Single server

How do I route a call to phone extension to IN_Group?
Sample: call to 101 phone extension since i didn't register it on xlite it will go straight to VM. what I want here is i'll call to 101 and it will drop to any in_group NOT 101 VM

[steve745]

a little more detail on what you are tying to accomplish will help I use simular configurations for routing incoming call. I use DID's and Ingroups maybe edit the dial plan for queue and use members. Are your agents Loged into the dialer or just phones with ext like a regular PBX? How does the Call end up at ext 101?

[williamconley]

To get this functionality, you'll want to just create a second ingroup in a new campaign with the phone in question set up as a remote on-hook agent in that campaign with that ingroup selected. so when a call comes to this new ingroup/campaign it will ring only that one phone. then you set the drop call action of this new ingroup to the failover ingroup you wanted earlier. ingroup-to-ingroup works. and it has the added bonus of making it easy to add a 2nd phone to the new ingroup in case you want to ring two phones before droppping to the 2nd ingroup.

don't forget the "on-hook" portion of the remote agent to allow "no answer".

[rrb555]

1. I've created new phone extension under Admin > Phones > 101
2. setup ext 101 using xlite > registration ok > call using other phone ext sample 102 calling to 101 > extension 101 ringing
3. unregister 101 by exiting Xlite Application > call again using ext 102 to 101 > directly went to VM of 101
This is what I want to do, I will not use ext 101 in any softphone application. But i still wanted agents able to reach it and the call will be routed to DID/In group so I can make an inbound campaign.

additional info, i can always buy DID from my VOIP, setup it to DID then forward it to In group then to a campaign. VOILA a simple inbound settings
But here I just want to make use of the local phone ext then setup a simple inbound campaign so agents can conference with other agents.
I am quite bit aware of the local closer but I believe local closer doesn't have any capabilities of engaging to a 3 way conference it use to transfer the customer.

or i am open for any suggestion

[williamconley]

to 3-way with a local closer you use the "dial with customer" button with the "consultative" checkbox checked after choosing the agent from the "agents" link in the transfer dialog.

if you do not want to send the data to the closer, you can use the closer's phone (dialplan number) along with the "dial with customer" button and it will consider this a purely 3-way call without sending any data to the closer's screen, but you may get a busy signal if the closer is logged in. thus, the above method which does send the data is preferred for logged in closers.

[rrb555]

Hi William,

to 3-way with a local closer you use the "dial with customer" button with the "consultative" checkbox checked after choosing the agent from the "agents" link in the transfer dialog.

I clicked on "agents" and no one is showing

so far here are my settings
Campaign: TESTCAMP
Allow Closers: Y
Default Transfer Group: AGENTDIRECT
Allowed Transfer Groups: AGENTDIRECT

[williamconley]

perhaps you could have some agents log in (and log in to the agentdirect ingroup during login) and they would appear in the list. otherwise, they cannot accept calls through the agentdirect ingroup anyway. it is also possible your user does not have permission to view the other agents status. this is controlled by the user group settings.

[wwwanaya]

I want to make the same thing.

Create an DID from a local extension. Without DID provider.

[williamconley]

I want to make the same thing.

Create an DID from a local extension. Without DID provider.

Change the "context" of a phone (under Admin->phones) to "trunkinbound". That's all you have to do.

[wwwanaya]

It say me the error: the person at the extention {My DID ext} Is unavailable...

[williamconley]

It say me the error: the person at the extention {My DID ext} Is unavailable...

1) Did you create the DID before dialing it? (After all, if you want to test DID 5554442222, you may want to create it before dialing it ...)
2) Output from the asterisk CLI?
3) It is entirely possible that the response was that of the default DID (which may cause what you described ... since there is no way for the default DID to know in advance where you want it to go before you configure it ...)

In all cases, asterisk CLI output of your problem will be extremely helpful in deciding where to look next. And before I forget:

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

[wwwanaya]

Nice party we got here with the contexts...
Ok so I'll create a DID with XXXXXXXXXX 10 digits number, I use the ext. 101 maybe that's the error, LOL. I'll let you know.

[wwwanaya]

Here we are:

I created a DID

http://snag.gy/S9t9L.jpg

Then, I create the phone with the context that you said:

http://snag.gy/RSfn2.jpg
http://snag.gy/Cx7PM.jpg

when I call from another ext. the asterisk show this on the cli:

[May 30 20:53:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 30 20:53:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 30 20:53:09] == Using SIP RTP CoS mark 5
[May 30 20:53:09] -- Executing [5554442222@default:1] Dial("SIP/8001-0000000f", "SIP/5554442222,60,") in new stack
[May 30 20:53:09] WARNING[2779]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[May 30 20:53:09] == Everyone is busy/congested at this time (1:0/0/1)
[May 30 20:53:09] -- Executing [5554442222@default:2] Goto("SIP/8001-0000000f", "default,850266666666665554442222,1") in new stack
[May 30 20:53:09] -- Goto (default,850266666666665554442222,1)
[May 30 20:53:09] -- Executing [850266666666665554442222@default:1] Wait("SIP/8001-0000000f", "1") in new stack
[May 30 20:53:10] -- Executing [850266666666665554442222@default:2] VoiceMail("SIP/8001-0000000f", "5554442222,u") in new stack
[May 30 20:53:10] -- <SIP/8001-0000000f> Playing 'vm-theperson.gsm' (language 'en')
[May 30 20:53:12] -- <SIP/8001-0000000f> Playing 'digits/5.gsm' (language 'en')
[May 30 20:53:12] -- <SIP/8001-0000000f> Playing 'digits/5.gsm' (language 'en')
[May 30 20:53:13] -- <SIP/8001-0000000f> Playing 'digits/5.gsm' (language 'en')
[May 30 20:53:14] -- <SIP/8001-0000000f> Playing 'digits/4.gsm' (language 'en')
[May 30 20:53:15] -- <SIP/8001-0000000f> Playing 'digits/4.gsm' (language 'en')
[May 30 20:53:15] -- <SIP/8001-0000000f> Playing 'digits/4.gsm' (language 'en')
[May 30 20:53:16] -- <SIP/8001-0000000f> Playing 'digits/2.gsm' (language 'en')
[May 30 20:53:16] -- <SIP/8001-0000000f> Playing 'digits/2.gsm' (language 'en')
[May 30 20:53:17] -- <SIP/8001-0000000f> Playing 'digits/2.gsm' (language 'en')
[May 30 20:53:18] -- <SIP/8001-0000000f> Playing 'digits/2.gsm' (language 'en')
[May 30 20:53:18] -- <SIP/8001-0000000f> Playing 'vm-isunavail.gsm' (language 'en')
[May 30 20:53:19] -- <SIP/8001-0000000f> Playing 'vm-intro.gsm' (language 'en')
[May 30 20:53:22] DTMF[2779]: channel.c:4151 __ast_read: DTMF begin '#' received on SIP/8001-0000000f
[May 30 20:53:22] DTMF[2779]: channel.c:4155 __ast_read: DTMF begin ignored '#' on SIP/8001-0000000f
[May 30 20:53:22] DTMF[2779]: channel.c:4066 __ast_read: DTMF end '#' received on SIP/8001-0000000f, duration 80 ms
[May 30 20:53:22] DTMF[2779]: channel.c:4135 __ast_read: DTMF end passthrough '#' on SIP/8001-0000000f
[May 30 20:53:22] -- <SIP/8001-0000000f> Playing 'beep.gsm' (language 'en')
[May 30 20:53:22] -- Recording the message
[May 30 20:53:22] -- x=0, open writing: /var/spool/asterisk/voicemail/default/5554442222/tmp/CJ5TzY format: wav49, 0x893d788
[May 30 20:53:22] -- x=1, open writing: /var/spool/asterisk/voicemail/default/5554442222/tmp/CJ5TzY format: gsm, 0x8970018
[May 30 20:53:22] -- x=2, open writing: /var/spool/asterisk/voicemail/default/5554442222/tmp/CJ5TzY format: wav, 0x88fa928



VERSION: 2.9-441a
BUILD: 140612-1628
© 2014 ViciDial Group

go*CLI> core show version
Asterisk 1.8.23.0-1_centos5.go RPM by demian@goautodial.com built by root @ go.goautodial.org on a x86_64 running Linux on 2013-12-06 23:34:26 UTC

[wwwanaya]

go*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
5554442222/5554442222 (Unspecified) D N 0 UNKNOWN
8001/8001 192.168.1.2 D N

[williamconley]

Executing [5554442222@default:1] Dial("SIP/8001-0000000f", "SIP/5554442222,60,") in new stack

This shows the call being executed in "default" not "trunkinbound". Perhaps you forgot to change the context of the phone to trunkinbound (which was the entire concept).

Note that there is more than one "context" setting for the phone. Change both to be safe.

[boybawang]

set context=trunkinbound on your carrier that has your DID

[wwwanaya]

Executing [5554442222@default:1] Dial("SIP/8001-0000000f", "SIP/5554442222,60,") in new stack

This shows the call being executed in "default" not "trunkinbound". Perhaps you forgot to change the context of the phone to trunkinbound (which was the entire concept).

Note that there is more than one "context" setting for the phone. Change both to be safe.

Yes is the phone context and extension context right? I change it for both.

[williamconley]

I think you missed what I said. The output shows "default" not "trunkinbound". So ... no, your context entry for the phone is not correct (or that entry was not saved and/or activated for some reason).