vicidial.org

by **dbanks** » Thu Mar 10, 2011 10:39 am

Hello all,

I am not an asterisk or vicidial guru but I "finally" get the job done. I have installed Goautodial for my company. The install went well as did the config. However the strangest thing, the softphone does not show as registered on the screen. I am able to dial another extension or a phone number and hear the person and be heard as well. Below is the "Error" I get when I launch the phone:

[Mar 10 10:01:56] NOTICE[2209]: chan_sip.c:16378 handle_request_register: Registration from '"cc100"<sip:cc100@19 2.168.15.77>' failed for '192.168.15.4' - No matching peer found

I have checked user and pw but no joy. Any ideas ??

Thank You

by **williamconley** » Thu Mar 10, 2011 11:29 am

1) Always post your VICIDIAL Version and Build (bottom left corner of almost every admin screen).

2) I assume you used some sort of installation instruction set with a setup guide. Which one did you use, and where did you stop? (I assume at registration ... but did you skip anything? and which one did you use?)

3) Have you read the Vicidial Manager's Manual (eflo.net, free and paid versions available).

4) "I have checked user and pw" ... POST them (obviously you can change them later, but post the entire setup for user/server/password that you are using in the setup for a single phone, both in the Admin->Phones GUI and on the phone itself).

by **dbanks** » Thu Mar 10, 2011 11:46 am

williamconley wrote:1) Always post your VICIDIAL Version and Build (bottom left corner of almost every admin screen).

2) I assume you used some sort of installation instruction set with a setup guide. Which one did you use, and where did you stop? (I assume at registration ... but did you skip anything? and which one did you use?)

3) Have you read the Vicidial Manager's Manual (eflo.net, free and paid versions available).

4) "I have checked user and pw" ... POST them (obviously you can change them later, but post the entire setup for user/server/password that you are using in the setup for a single phone, both in the Admin->Phones GUI and on the phone itself).

I downloaded the Goautodial ISO and went through the complete install. Yes I have read the manual and I was not able to determine the issue, which is why I posted my question here.

Extension

[cc100]
username=cc100
secret=test
accountcode=cc100
callerid="station 100" <305XXXXXXX>
mailbox=100
context=default
type=friend
host=dynamic

I actually changed the secret in goautodial and I tried nano but it does not seem to update. What setup server file are you asking to see please.

-------------------------------------------------

by **williamconley** » Thu Mar 10, 2011 11:54 am

VICIDIAL Version and Build

(the build makes a difference for troubleshooting issues as they arise in the code, always include it, it helps everyone in the community and can often help YOU if there is a known bug with the version you are using

)

Now that we have the asterisk half of the puzzle, what type of phone are you using and what did you put into that for settings?

by **dbanks** » Thu Mar 10, 2011 12:03 pm

williamconley wrote:
VICIDIAL Version and Build
(the build makes a difference for troubleshooting issues as they arise in the code, always include it, it helps everyone in the community and can often help YOU if there is a known bug with the version you are using )

Now that we have the asterisk half of the puzzle, what type of phone are you using and what did you put into that for settings?

Understood, why didnt I think of that............

Eyebeam is the brand of softphone. Here is the deal, when I did the initial install I tested the phone to see if it would register and it did with the basic and default config. However I had issues with one way audio and I then added "externip" to my sip.conf, one way audio issues resolved. However it was at that point that I noticed the extension would no longer register, but I can certainly dial a phone and have a perfect conversation. The fact that it is not "registered" is a problem for the vicidial side when I would try to login as an agent, it did not ring the phone.

by **williamconley** » Thu Mar 10, 2011 12:14 pm

VICIDIAL Version and Build

by **dbanks** » Thu Mar 10, 2011 12:31 pm

williamconley wrote:VICIDIAL Version and Build

I beleive it is

Goautodial v.2.2.1-237
VICIDIAL Build v.100510-2015

Not 100% Where can I find this ?

It says the Distro is Goautodial 2.0CE

by **williamconley** » Thu Mar 10, 2011 12:51 pm

Sorry about that: but it is NOT goautodial v2.2.1 ... that's misleading. It is VICIDIAL 2.2.1 Build 100510-2015.

The GoAutoDial version is 2.0CE.

Goautodial is ONLY the installer (he has also placed some wrapper code in place which tends to mask the vicidial imaging, but the SOFTWARE you installed is vicidial, the disk you used to install it was GoAutoDial).

If this is in any way annoying, you also have the option of using the Vicibox Redux installer CD (vicibox.com) which does not "mask" vicidial at all (vanilla install, but on OpenSuSE instead of CentOS and has only the STOCK agent screen). On the upside, Vicibox can install the latest SVN code, which is at version 2.4 instead of 2.2.1, without having to upgrade after, it's already there

You likely have an issue with what Asterisk considers "local". Have another look at the sip.conf file and make sure that your local network is considered "local" so that local eyebeam is treated differently than a non-local carrier (ie: no NAT and no externip)

by **dbanks** » Thu Mar 10, 2011 12:56 pm

Thanks for the clarification. I have looked but cant see were there is an issue.

[GENERAL]

context=default ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
externip = xxx.xxx.xx.xxx ; Public IP Address
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
mohinterpret=default
mohsuggest=default
language=en ; Default language setting for all users/peers
relaxdtmf=yes ; Relax dtmf handling
trustrpid = no ; If Remote-Party-ID should be trusted
sendrpid = yes ; If Remote-Party-ID should be sent
progressinband=no ; If we should generate in-band ringing always
dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
videosupport=no ; Turn on support for SIP video. You need to turn this on
callevents=yes ; generate manager events when sip ua
rtpkeepalive=60 ; Send keepalives in the RTP stream to keep NAT open
notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
limitonpeers = yes ; Apply call limits on peers only. This will improve
localnet=192.168.0.0/255.255.0.0 ; All RFC 1918 addresses are local networks
localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
nat=yes ; Global NAT settings (Affects all peers and users)
canreinvite=no ; Asterisk by default tries to redirect the
jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
jbmaxsize = 100 ; Max length of the jitterbuffer in milliseconds.
jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
qualify=yes ; By default, qualify all peers at 2000ms
limitonpeer = yes ; enable call limit on a per peer basis, different from limitonpeers

by **williamconley** » Thu Mar 10, 2011 1:46 pm

localnet=192.168.0.0/255.255.0.0 ; All RFC 1918 addresses are local networks
localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
localnet=169.254.0.0/255.255.0.0 ;Zero conf local network

are your local computers (and your vicidial) on any of these "local" networks?[/code]

by **dbanks** » Thu Mar 10, 2011 1:50 pm

williamconley wrote:
localnet=192.168.0.0/255.255.0.0 ; All RFC 1918 addresses are local networks
localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
are your local computers (and your vicidial) on any of these "local" networks?[/code]

No they are not. But when I tested the registration before I put the externip, it registered. Should I change these accordingly ?

by **williamconley** » Thu Mar 10, 2011 1:59 pm

your localnet should be listed in this so it will NOT use the externip with these machines.

by **dbanks** » Thu Mar 10, 2011 2:32 pm

williamconley wrote:your localnet should be listed in this so it will NOT use the externip with these machines.

OK I did that and still no registration and the other line does not hear the touch tones.

by **williamconley** » Thu Mar 10, 2011 2:35 pm

have you reloaded asterisk (or rebooted to be safer?)

what is the ip of your vici server (internal ip & external ip & ip listed in astguiclient.conf would all be nice, xx out the last two of the external ...) and the ip of the local computer with the "issue"

by **dbanks** » Thu Mar 10, 2011 2:38 pm

williamconley wrote:have you reloaded asterisk (or rebooted to be safer?)

Always a reboot.

by **dbanks** » Thu Mar 10, 2011 4:38 pm

williamconley wrote:have you reloaded asterisk (or rebooted to be safer?)

what is the ip of your vici server (internal ip & external ip & ip listed in astguiclient.conf would all be nice, xx out the last two of the external ...) and the ip of the local computer with the "issue"

That didnt do it? An other ideas William ?

Thanks

by **williamconley** » Thu Mar 10, 2011 5:12 pm

dbanks wrote:
williamconley wrote:what is the ip of your vici server (internal ip & external ip & ip listed in astguiclient.conf would all be nice, xx out the last two of the external ...) and the ip of the local computer with the "issue"

That didnt do it? An other ideas William ?

Thanks

by **dbanks** » Thu Mar 10, 2011 5:25 pm

This is what comes up when I nano

# astguiclient.conf - configuration elements for the astguiclient package
# this is the astguiclient configuration file
# all comments will be lost if you run install.pl again

# Paths used by astGUIclient
PATHhome => /usr/share/astguiclient
PATHlogs => /var/log/astguiclient
PATHagi => /var/lib/asterisk/agi-bin
PATHweb => /var/www/html
PATHsounds => /var/lib/asterisk/sounds
PATHmonitor => /var/spool/asterisk/monitor
PATHDONEmonitor => /var/spool/asterisk/monitorDONE

# The IP address of this machine
VARserver_ip => 192.168.15.77

# Database connection information
VARDB_server => localhost
VARDB_database => asterisk
VARDB_user => cron
VARDB_pass => 1234
VARDB_port => 3306

# Alpha-Numeric list of the astGUIclient processes to be kept running
# (value should be listing of characters with no spaces: 123456)
# X - NO KEEPALIVE PROCESSES (use only if you want none to be keepalive)
# 1 - AST_update
# 2 - AST_send_listen
# 3 - AST_VDauto_dial
# 4 - AST_VDremote_agents
# 5 - AST_VDadapt (If multi-server system, this must only be on one server)
# 6 - FastAGI_log
# 7 - AST_VDauto_dial_FILL (only for multi-server, this must only be on one server)
# 8 - ip_relay (used for blind agent monitoring)
# 9 - Timeclock auto logout
VARactive_keepalives => 12345689

# Asterisk version VICIDIAL is installed for
VARasterisk_version => 1.4

# FTP recording archive connection information
VARFTP_host => 192.168.1.2

by **williamconley** » Thu Mar 10, 2011 5:43 pm

what is the ip of your vici server ( & external ip & , xx out the last two of the external ...) and the ip of the local computer with the "issue"

by **dbanks** » Thu Mar 10, 2011 5:54 pm

williamconley wrote:what is the ip of your vici server ( & external ip & , xx out the last two of the external ...) and the ip of the local computer with the "issue"

internal -192.168.15.77

External - 190.195.xx.xxx

The computer that I am working from is 192.168.15.4

by **williamconley** » Thu Mar 10, 2011 6:07 pm

whew.

now: before we go nuts ... verify by taking externip OUT (just add a ; to the beginning of the line!) and see if that causes it to register again (to verify that your assertion is correct)

and sip debug to watch the registration attempt with and without the externip in place.

by **dbanks** » Thu Mar 10, 2011 6:47 pm

williamconley wrote:whew.

now: before we go nuts ... verify by taking externip OUT (just add a ; to the beginning of the line!) and see if that causes it to register again (to verify that your assertion is correct)

and sip debug to watch the registration attempt with and without the externip in place.

did that, rebooted, still no registration. Strange.

by **williamconley** » Thu Mar 10, 2011 6:51 pm

any traffic on the asterisk sip debug from the ip of your soft phone?

what settings did you put in the soft phone? (be specific, fields and values ...)

by **dbanks** » Thu Mar 10, 2011 8:02 pm

Sorry for the delay, power outage down here.

by **dbanks** » Thu Mar 10, 2011 8:15 pm

Display Name - cc100
User Name - cc100
PW - test (default)
Auth. User Name - cc100
Domain - 192.168.15.77

I will try to figure out what is going on in debug. I have never had to use it before..

by **dbanks** » Thu Mar 10, 2011 8:19 pm

Getting this in the CLI

Verbosity is at least 21
[Mar 10 20:18:55] NOTICE[2176]: chan_sip.c:16378 handle_request_register: Registration from '"cc100"<sip:cc100@192.168.15.77>' failed for '192.168.15.3' - No matching peer found

by **dbanks** » Thu Mar 10, 2011 8:24 pm

Got this in "agi debug" Hope that is the right command......

Verbosity is at least 21
go*CLI> agi debug
AGI Debugging Enabled
[Mar 10 20:23:02] == Parsing '/etc/asterisk/manager.conf': [Mar 10 20:23:02] Found
[Mar 10 20:23:02] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 10 20:23:02] == Parsing '/etc/asterisk/manager.conf': [Mar 10 20:23:02] Found
[Mar 10 20:23:02] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 10 20:23:02] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 10 20:23:02] NOTICE[2176]: chan_sip.c:16378 handle_request_register: Registration from '"cc100"<sip:cc100@192.168.15.77>' failed for '192.168.15.3' - No matching peer found
[Mar 10 20:23:07] == Parsing '/etc/asterisk/manager.conf': [Mar 10 20:23:07] Found
[Mar 10 20:23:07] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 10 20:23:07] ERROR[17420]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
[Mar 10 20:23:07] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 10 20:23:08] == Manager 'sendcron' logged off from 127.0.0.1
go*CLI>

by **williamconley** » Thu Mar 10, 2011 8:36 pm

and does cc100 exist in the sip.conf include files? (ie: starting with sip.conf, can you trace #includes to a context named [cc100]?)

another method, of course, is "sip show peers"

by **dbanks** » Thu Mar 10, 2011 9:07 pm

williamconley wrote:and does cc100 exist in the sip.conf include files? (ie: starting with sip.conf, can you trace #includes to a context named [cc100]?)

another method, of course, is "sip show peers"

My apologies, I am not sure I follow. In the sip.conf there is no reference to any extension.

Doing sip show peers I get ZERO across the board

by **williamconley** » Thu Mar 10, 2011 9:14 pm

sip.conf never has anything in it "directly". there are "#include" statements which include other files. in one of those (likely sip-vicidial.conf) we would expect to find [cc100], but just finding it in sip-vicidial without ensuring that sip-vicidial was "#include"d in sip.conf would be poor troubleshooting ...

however: if sip show peers does not list any peers ...

what do you have in vicidial admin under "Admin->Phones"?

by **dbanks** » Thu Mar 10, 2011 9:21 pm

williamconley wrote:sip.conf never has anything in it "directly". there are "#include" statements which include other files. in one of those (likely sip-vicidial.conf) we would expect to find [cc100], but just finding it in sip-vicidial without ensuring that sip-vicidial was "#include"d in sip.conf would be poor troubleshooting ...

however: if sip show peers does not list any peers ...

what do you have in vicidial admin under "Admin->Phones"?

In the sip-vicidial.conf, yes the extensions are listed there.

I do not see the #includes statements in the sip.conf

In Admin.Phones, I show all of the extensions and they are active.

by **williamconley** » Thu Mar 10, 2011 9:27 pm

Code: Select all: #include sip-vicidial.conf

does not appear in your /etc/asterisk/sip.conf file?

by **dbanks** » Thu Mar 10, 2011 9:31 pm

williamconley wrote:
Code: Select all
#include sip-vicidial.conf
does not appear in your /etc/asterisk/sip.conf file?

No sir. The sip.conf is pasted in the 9th post from the top.

by **williamconley** » Thu Mar 10, 2011 9:37 pm

1)

put it in at the bottom

Code: Select all: qualify=yes ; By default, qualify all peers at 2000ms limitonpeer = yes ; enable call limit on a per peer basis, different from limitonpeers #include sip-vicidial.conf

do not forget to "reload" asterisk (asterisk -rx "reload")
2)

Where'd it go? (if you registered once ... it WAS there ... so ... What Else Changed When That Changed?)

by **dbanks** » Thu Mar 10, 2011 9:50 pm

You nailed it. Registers and I am able to get the ring as an agent. I sincerely appreciate you time and effort.

by **williamconley** » Thu Mar 10, 2011 10:02 pm

now find out who deleted that line from the conf (some clean freak, of course

)

by **dbanks** » Thu Mar 10, 2011 10:10 pm

williamconley wrote:now find out who deleted that line from the conf (some clean freak, of course )

Honestly, I am the only one that touched this box. Here is the kicker, I installed it and was trying to test the dialer and the extension connected and I was able to dial a second extension I had setup as a test. Then while trying to dial out to the world I got one way audio. Got that fixed and then the phone would not register. So I reformatted and reinstalled and got the same issue, the phone would not register. I never delete or change configurations before I make a copy and looking back, it was never in the original sip.conf. Very strange..

by **williamconley** » Thu Mar 10, 2011 10:30 pm

odd. but normal users never touch the sip.conf (not an accident there, externip i believe is expected to be put in the carrier setup for any carrier requiring it) to avoid ... issues. same with extension.conf and any .conf for that matter.

ESPECIALLY for "Asterisk Professionals"

by **dbanks** » Fri Mar 11, 2011 10:48 am

Good Morning and thanks again.

One question I have is, is it possible to change the field names in the
new_listloader_superL.php and the agent screen to reflect the specific data titles we need to have displayed without breaking Vici ??

by **williamconley** » Fri Mar 11, 2011 11:09 am

vicidial 2.4 has that, i dunno about 2.2.

admin->system settings in 2.4 allows specifying new labels for the standard fields (ie: changing first name to company and last name to contact, that sort of thing), which changes the field labels on the agent screens. i expect that also changes it everywhere else the labels are used.

vicidial.org

Extension does not register but I can dial numbers ?????????

Extension does not register but I can dial numbers ?????????

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