No one is in your session

[paruchurup]

Hi

when I am logging into agent gui, I am getting this message, "Now one is in your session: 8600052".

For this conf_exten(8600052), phone extension is 201 (SIP PHONE) which I created and loggged in with. I thought ok as the phone user on extension is not available, it is showing this message, then I tried logging with phone credentials I created from XLite., not able to login.

with the phone details I created, I should be able to login from XLite? It is not showing in asterisk CLI as well when I did > sip show users

Please correct my understanding on this so that I can understand how it should work

Thanks

[williamconley]

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

3) I believe that if you start in the Free version of the Vicidial Manager's Manual at page One and don't skip any pages, you will likely resolve your issue (keep going without skipping any pages until your agent has logged in!). If not, please come back and answer this question: Does your X-Lite phone Register? If not, where did you get the information for the Xlite configuration? (Which field names in Vicidial Admin->Phones correspond to which fields in the Xlite setup?)

4) Happy Hunting!

[paruchurup]

Thanks william for your suggestions. I will follow your suggestions for posting.

Vicidial 2.7 rc1 | Asterisk 1.4.44 | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | XLite | RedHat Linux |Scratch Install

I got the soft phone registered. The problem was with two sip conf files. I have included the sip-vicidial.conf in sip.conf by

sip.conf
#include sip-vicidial.conf

I am still getting this error because the lead phone number is a PSTN phone number which came by loading example leads text file. My requirement is to call a customer who is on soft phone.
How can I add a lead with sip phone number?

[paruchurup]

Hi

The phones I have added in Admin gui, I am able to register via softphones but cannot call each other. Because the context is not updated correctly in the configuration files, as the vicidialer does this updation of the conf files, I cannot change it also.

sip-vicidial.conf

Code: Select all

[201]
username=201
secret=<201pswd>
accountcode=201
callerid="sip201" <7275551212>
mailbox=201
context=default
type=friend
host=dynamic

[601]
username=601
secret=<601pswd>
accountcode=601
callerid="601_praveena" <>
mailbox=601
context=default
type=friend
host=dynamic



extensions-vicidial.conf

Code: Select all


[vicidial-auto-phones]
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

; Phones direct dial extensions:
exten => 201,1,Dial(SIP/201|60|)
exten => 201,2,Goto(default,85026666666666201,1)
exten => 601,1,Dial(SIP/601|60|)
exten => 601,2,Goto(default,85026666666666601,1)



context is 'default' in sip.conf but in extensions it is 'vicidial-auto-phones'. Whyt it is not updating same context in both sip.conf and extensions.conf ?
Because of this 201 cannot call 601 on softphones.

I have updated the table vicidial_list in database to change the phone_number field to 601 for the lead. As an agent I have logged in with phone login 201, so I want to call person on extension 601 logged in on soft phone.

Could any one tell me how to proceed from here...

[paruchurup]

I am able to call 601 and 201 via soft phones by inlcuding the [vicidial-auto-phones] context in [default] context.

Code: Select all

[default]
inlcude => vicidial-auto-phones



Please let me know how do I call customer with sip phone number 601 from Agent gui? It is saying wiating for ring. and call is not happening... and I am thinking this is the reason it is giving no one in session.

[williamconley]

There is no "stock" method to call local phones from a vicidial logged in agent.

If you want a "test scenario", you could create a special carrier that would route all calls through that carrier to a local phone. We create the "fake" carrier for new vicidial users to play with their system quite regularly.

The carrier is basically normal except the dial command is modified to force the call to SIP/ccXXX instead of sending to a real carrier.

Then you can load real leads with real names and numbers and as long as the dial prefix in the campaign sends the calls to the fake carrier, all calls will go to the chosen SIP phone.

[paruchurup]

Thanks Wiliam.

I have created the carrier as given in the manager manual. I understand that in vicidial all call will be routed through this carrier.

Could you please make your sentence more clear,I am not getting it. I understand that I need to change dialplan entry while creating the carrier but not clear how exactly it should look like

Code: Select all

The carrier is basically normal except the dial command is modified to force the call to SIP/ccXXX instead of sending to a real carrier.


The phone_number filed in vicidial_list table which I give for the lead will be 601 only or do I need to prefix anything here..?

Thanks

[williamconley]

Since you asked:

Code: Select all

exten => _01NXXNXXXXXX,1,Set(CALLERID(num)=408122866)
exten => _01NXXNXXXXXX,n,AGI(agi://127.0.0.1:4577/call_log)
exten => _01NXXNXXXXXX,n,Dial(SIP/cc150,,tTor)
exten => _01NXXNXXXXXX,n,Hangup

obviously you should replace 408122866 with a real callerID, but it would be overridden anyway by the second line unless the call is not from a Campaign (this will be the callerid used for manually dialed calls from soft phones).

also obviously you would replace cc150 with the actual sip phone extension you want to get the calls.

Dial prefix of "0" in the campaign will activate it. As will dialing 0+1+ 10digit US phone number.

This will convert any 10 digit US phone number to a call to cc150 on any campaign with a dial prefix of "0". Just remember that the dial code on all US phones is "1" (entered on each record when loading leads, or use the dial code override and choose USA to force a dial code of "1"). If you need this for another country, show your dial pattern for your country and I'm sure someone can rewrite it to work easily.

[NelsonA]

Good evening, Im new to VICI dial and very little knowledge, we have been working just fine with it but my IT joinned the marines, but yesterday I did a reset on it and for some reason IM getting no one is in your session 8600051, the dialer is working fine but I cant connect the calls, I need someone to help me please configure it to the right setting and take me step by step to check on it, please help I need to get it back on line

[carpenox]

welcome to the community Nelson. There is many things that it can be but I can definitely help you debug it. Please start by typing this for me to see if dahdi is running.

dahdi_cfg -v

Let me know what it comes back with. This is typed on the ssh or shell area of your server in the Linux command line interface. If you need live help feel free to reach out to me through any of the methods in my signature. Good luck

Chris