Sorry, there are no available sessions

[chrisnetronix]

i cant get the user to login correclty via x-lite or agent interface>

cli:

NOTICE[3703]: chan_sip.c:16378 handle_request_register: Registration from '"7777"<sip:7777@ 66.71.249.170>' failed for '71.203.248.167' - No matching peer found
[Aug 24 00:02:35] NOTICE[3703]: chan_sip.c:16378 handle_request_register: Registration from '"7777"<sip:7777@ 66.71.249.170>' failed for '71.203.248.167' - No matching peer found
[Aug 24 00:02:35] NOTICE[3703]: chan_sip.c:16378 handle_request_register: Registration from '"7777"<sip:7777@ 66.71.249.170>' failed for '71.203.248.167' - No matching peer found
[Aug 24 00:02:35] NOTICE[3703]: chan_sip.c:16378 handle_request_register: Registration from '"7777"<sip:7777@ 66.71.249.170>' failed for '71.203.248.167' - No matching peer found

here is my carrier config:

ViciBox_Redux.i686-2.0.2 from .iso | Vicidial 2.2.1-237 Build 100510-2015 | Asterisk 1.4.21.2 | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation |

I got a flowroute account put in this information like this:

Carrier ID: Flowroute
Carrier Name: Flowroute
Carrier Description: none
Registration String: register => 14512814:xxxxxxxxxx@sip.flowroute.com
Template ID:
Account Entry:

[flowroute]
type=friend
secret=xxxxxxxxxx
username=14512814
host=sip.flowroute.com
dtmfmode=inband
rfc2833compensate=yes
context=inbound
canreinvite=no
allow=ulaw
allow=g729
insecure=port,invite
fromdomain=sip.flowroute.com

Protocol: SIP

Globals String:TESTSIPTRUNK = SIP/testcarrier

Dialplan Entry:

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(sip/${EXTEN:1}@flowroute,55,tTo)
exten => _91NXXNXXXXXX,3,Hangup


I went to phones and put in the server ip and username and passoerd of the 7777 agent and now it shows

there are no available sessions

cli

[Aug 24 00:52:16] NOTICE[3936]: chan_sip.c:16378 handle_request_register: Registration from '"7777"<sip:7777@66.71.249.170>' failed for '71.203.248.167' - Wrong password


the above is from trying to login via xlite

if i try to login via vicidial agent i get in cli:

[Aug 24 00:53:52] NOTICE[7503]: channel.c:3271 __ast_request_and_dial: Unable to request channel SIP/7777


Heres the sip debug cli:


SIP Debugging re-enabled
[Aug 24 02:14:19]
<--- SIP read from 70.167.153.130:5060 --->
OPTIONS sip:66.71.249.170:5060 SIP/2.0
Max-Forwards: 10
Record-Route: <sip:70.167.153.130;lr>
Via: SIP/2.0/UDP 70.167.153.130;branch=z9hG4bKd14f.8ef2bbd501a51aaf2c3592ef000f9621.0
Via: SIP/2.0/UDP 216.115.69.131:5060;branch=0
Route: <sip:70.167.153.130;lr;received="sip:66.71.249.170:5060">
From: sip:ping@invalid;tag=f549cba6
To: sip:66.71.249.170:5060
Call-ID: 887d2e83-118d3166-22d46a4@216.115.69.131
CSeq: 1 OPTIONS
Content-Length: 0


<------------->
[Aug 24 02:14:19] --- (11 headers 0 lines) ---
[Aug 24 02:14:19] Looking for s in default (domain 66.71.249.170)
[Aug 24 02:14:19]
<--- Transmitting (NAT) to 70.167.153.130:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 70.167.153.130;branch=z9hG4bKd14f.8ef2bbd501a51aaf2c3592ef000f9621.0;received=70.167.153.130
Via: SIP/2.0/UDP 216.115.69.131:5060;branch=0
From: sip:ping@invalid;tag=f549cba6
To: sip:66.71.249.170:5060;tag=as7f13dfe6
Call-ID: 887d2e83-118d3166-22d46a4@216.115.69.131
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------>
[Aug 24 02:14:19] Scheduling destruction of SIP dialog '887d2e83-118d3166-22d46a4@216.115.69.131' in 32000 ms (Method: OPTIONS)
[Aug 24 02:14:24] Really destroying SIP dialog 'MTQwNmFjMWRjZGNiNDY5ZDM3Mzk3ZjI2NThlN2UxNTg.' Method: REGISTER



now when trying to login via xlite i get:

[Aug 24 02:41:00] NOTICE[3936]: chan_sip.c:16378 handle_request_register: Registration from '"7777"<sip:7777@66.71.249.170>' failed for '71.203.248.167' - No matching peer found


Any Help?

[chrisnetronix]

Any help would be great

[Kumba]

Your phone is not configured with the proper registration credentials, or is not set-up properly in ViciDial. Double check that they are correct before continuing.

When you have your phone properly connected you should be able to dial '8302' on it and hear music.

[chrisnetronix]

I read all through tutorial A, and it says this about setting up phones at the beginning:

This tutorial assumes that your System Administrator has already set up
your phone entries for your agents' phones.

on page 66 is:

ADMIN - PHONES
Add a New Phone


but it just shows definitions to the fields, it does not have a step by step to setup, I just need to know what to put in the fields correctly

[chrisnetronix]

I had 3 managers manuals and i was reading an old one, now I am reading version 220.pdf

5/10/2010

I see where you are talking about now, now I can bet it to login to xlite correctly but when I go to the vicidial agent login it tells me:

Sorry, there are no available sessions

and cli says:

[Aug 24 21:23:04] NOTICE[3751]: chan_sip.c:3115 auto_congest: Auto-congesting SIP/102-0000000c
[Aug 24 21:23:04] > Channel SIP/102-0000000c was never answered.

[chrisnetronix]

When you have your phone properly connected you should be able to dial '8302' on it and hear music.

I do hear music on xlite

But : per agent login i still get:

Sorry, there are no available sessions

[williamconley]

try "qualify=yes" on the phone -> change to "qualify=no"

this will remove the auto-congesting, but it indicates a severe time delay when communicating with your x-lite from the server.

[chrisnetronix]

where is : qualify> yes at, I have no idea where this is

[williamconley]

depends on your setup. could be in admin -> templates (sip)

but it ends up in sip-vicidial.conf or sip.conf

you can change it directly in sip-vicidial.conf if you can find it there and it will be "overwritten" if you make any more changes in vicidial.

but if you go back into asterisk after making the change, and enter "reload", the new setting will take effect and will survive until you make a change in vicidial (making it a perfect test, since it is quite temporary).

if you must make the change in sip.conf, however, be sure to change it back.

and DO remember that you must enter "reload" in asterisk for it to take effect.

[chrisnetronix]

changed qualify to no

still
NOTICE[4196]: channel.c:3271 __ast_request_and_dial: Unable to request channel SIP/102

[williamconley]

do you still get

[Aug 24 21:23:04] NOTICE[3751]: chan_sip.c:3115 auto_congest: Auto-congesting SIP/102-0000000c

and did you

Code: Select all

reload

in asterisk?

[chrisnetronix]

here is the cli when trying to log in the agent interface:

this is with qualify no


[Aug 25 17:15:02] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 25 17:15:02] == Parsing '/etc/asterisk/manager.conf': [Aug 25 17:15:02] Found
[Aug 25 17:15:02] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 25 17:15:02] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 25 17:15:02] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 25 17:15:02] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 25 17:15:06] == Parsing '/etc/asterisk/manager.conf': [Aug 25 17:15:06] Found
[Aug 25 17:15:06] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 25 17:15:06] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 25 17:15:06] == Parsing '/etc/asterisk/manager.conf': [Aug 25 17:15:06] Found
[Aug 25 17:15:06] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 25 17:15:06] == Parsing '/etc/asterisk/manager.conf': [Aug 25 17:15:06] Found
[Aug 25 17:15:06] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 25 17:15:06] == Parsing '/etc/asterisk/manager.conf': [Aug 25 17:15:06] Found
[Aug 25 17:15:06] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 25 17:15:07] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 25 17:15:07] ERROR[7170]: utils.c:966 ast_carefulwrite: write() returned error: Connection reset by peer
[Aug 25 17:15:07] ERROR[7170]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
[Aug 25 17:15:07] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 25 17:15:07] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 25 17:15:07] == Parsing '/etc/asterisk/manager.conf': [Aug 25 17:15:07] Found
[Aug 25 17:15:07] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 25 17:15:07] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 25 17:15:13] == Parsing '/etc/asterisk/manager.conf': [Aug 25 17:15:13] Found
[Aug 25 17:15:13] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 25 17:15:31] ERROR[7020]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
[Aug 25 17:15:31] ERROR[7020]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
[Aug 25 17:15:31] ERROR[7020]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
[Aug 25 17:15:31] == Manager 'sendcron' logged off from 127.0.0.1

[chrisnetronix]

with qulify no i do not get the error you quoted and yes I used reload

but the above is what i see in cli and still on agent login i get the sessions error

[chrisnetronix]

Any Help?

[chrisnetronix]

need some help here guys

[williamconley]

that output does not appear to be from the same machine during a login attempt (ie: you posted the cli output from a moment when there was no login attempt, the earlier post had an attempt but this one didn't)

if this is really from the moment of login attempt, you may have been editing things that caused the system to lose track of the server's ip address or other records (because it didn't even attempt to dial, this often happens if the command is issue to the wrong server ip, it's merely ignored)

[striker]

hi

i think you have given the permit or deny option for registration.


check in your phone field wether you have given any ip address in the field of Computer IP address:



for your advice:
dont post your ip address anywere in the internet , your system can be easily hacked.

[rohit raj]

Re: Goautodial 2.1 - Sorry, there are no available sessions

Postby rohit raj » Sat Jun 22, 2013 11:04 am
hello everyone

i have also faced same problem and finally problem got solved

session cannot create is just due to wrong time set on ur linux machine. i used centos and after so many changes it wasnt solved then i change time by running this command on superuser cmd-->date today date

here is the explanation to set date in centos
root@servername ~]# date

Output will be like Tue Aug 23 12:32:22 EDT 2011

Now to set the desired date/time use the same date command with an argument of your required date/time as following format;
date MMDDhhmmYYYY

MM – Two digits of month (e.g. 08)
DD – Two digits of day of month (e.g. 24)
hh – Two digits of hour 24 hours format (e.g. 22)
mm – Two digits of minutes (e.g. 40)
YYYY – Four digits of year (e.g. 2011)

So, to set a date/time for 24th August 2011, 10:40 PM you may run the command as below

[root@servername ~]# date 082422402011

finally i restart everything and then i log in and wipeeeeeeeeeeeeee session available and agent successfully log in

any query send me mail -->rrrohitraj0 on my gmail id




nd also make new vicidial conference which u fill find at admin-conference-create new vicidial conference id-->8600051...add ur server ip...then add extension in that way sip/extension no ex sip/8001. finally save ur vicidial conference .. now see all conference and delete that conference with same id u create which is 8600051. similarly u can make many one