No ringing and both side voice issue on vicidial

[abhiupadhyayy]

Hi Everyone,

I am new on vicidial and I want your help guys I am not very good on vici dial I have started to learn vicidial.

I have Installed vicidial v9 on the virtual machine the problem I am facing "when I make (manual call) call I am not able to hear a ring and once call picked up not able to hear other people voice even he is not able to hear my voice" I have already asked with my sip provider he said that must be a dialer problem. I have used xlite and zoiper as my softphone still problem is there I have already checked for headphone all ok no problem with that.

I can hear vicidial standard messages "you are the only person in this conference" Once I logged in.

Please help me looking forward
I am new on vicidial

[sipprovider]
type=friend
disallow=all
allow=ulaw
allow=alaw
allow=g729
fromuser=rxxxxx
secret=98xxxxxx
username=rmango
host=xxx.99.36.xx
canreinvite=no
insecure=very
qualify=yes
nat=yes



exten => _52X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _52X.,2,Dial(sip/${EXTEN:2}@sipprovider,55,o)
exten => _52X.,3,Hangup
.

[Kabis]

Hi,

In /etc/asterisk/sip.conf, Please confirm
nat=yes.
If not available, please uncomment and reload asterisk and check.

Regards
KABIS

[abhiupadhyayy]

thanks for support really appreciate your reply but "sip.conf" nat=yes it is already applied and already uncomment.

what else I should do?

Regards
Abhi

[Kabis]

Try Following Options:

exten => _52X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _52X.,2,Dial(SIP/sipprovider/${EXTEN:2},,To)
exten => _52X.,3,Hangup()

Try this dialplan.

1. If not ring, try to add 'r' in Dial command at last like 'Tor'.

2. Are you using your extensions using public IP, please add nat=yes in respective extensions.

If This also not works, we need to check configuration directly in server.


Regards
KABIS

[carpenox]

it sounds like a dahdi issue, in the shell type: dadhi_cfg -vvv

Show me the results

[abhiupadhyayy]

it sounds like a dahdi issue, in the shell type: dadhi_cfg -vvv

Show me the results

Thanks for the reply I really appreciate

as I mentioned above I am new on vici dial

I checked for file in dadhi_cfg I didn't find anything so I checked for dadhi directory and I got some list

assigned-spans.conf modules system.conf.bak
assigned-spans.conf.sample modules.sample system.conf.sample
genconf_parameters span-types.conf.sample
init.conf system.conf

will you please tell me where I can find it?

again thanks a lot

Regards
Abhi

[carpenox]

yea no prob, ssh to your shell and make sure you have root privileges,you can do this by typing "whoami" and if it doesnt say "root" then type "sudo -i" which should bring you into root

Then type the follwoing:

Code: Select all

mkdir /usr/src/asterisk
cd /usr/src/asterisk
wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz
tar -xzvf dahdi-linux-complete-current.tar.gz
cd /usr/src/asterisk/dahdi-linux-complete-3.1.0+3.1.0/
make && make install
modprobe dahdi
dahdi_cfg -vvvvv


This should get dahdi working for ya and then try whatever wasn't working for you before. Good luck,

-Nox

[abhiupadhyayy]

yea no prob, ssh to your shell and make sure you have root privileges,you can do this by typing "whoami" and if it doesnt say "root" then type "sudo -i" which should bring you into root

Then type the follwoing:

Code: Select all

mkdir /usr/src/asterisk
cd /usr/src/asterisk
wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz
tar -xzvf dahdi-linux-complete-current.tar.gz
cd /usr/src/asterisk/dahdi-linux-complete-3.1.0+3.1.0/
make && make install
modprobe dahdi
dahdi_cfg -vvvvv


This should get dahdi working for ya and then try whatever wasn't working for you before. Good luck,

-Nox

Thanks a lot
I have used the above code this is what I am getting now after that

Code: Select all

DAHDI Tools Version - 3.1.0

DAHDI Version: 3.1.0
Echo Canceller(s):
Configuration
======================


Channel map:


0 channels to configure.



still, problem persists

Regards
Abhi

[carpenox]

ok run these commands:

Code: Select all

screen -list
asterisk -r
core set verbose 25
sip set debug on


Try to make a call after you enter that last line, and show me the output from just before, during and after the call attempt.

Who do you have your SIP trunk thru?

[abhiupadhyayy]

As you asked I have used all command sharing output with you

screen -list

There are screens on:
2256.ASTemail (Detached)
2252.ASTVDadFILL (Detached)
2249.ASTfastlog (Detached)
2246.ASTVDadapt (Detached)
2243.ASTVDremote (Detached)
2240.ASTVDauto (Detached)
2237.ASTlisten (Detached)
2234.ASTsend (Detached)
2231.ASTupdate (Detached)
2041.asterisk (Detached)
2036.astshell20200503203747 (Detached)
11 Sockets in /run/screens/S-root.

core set verbose 25

it was 21 now it is 25

thanks a lot for support
I have changed the dialplan as below mentioned now I can hear ring but the voice is not going through now what should I try to do

exten => _52X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _52X.,2,Dial(SIP/sipprovider/${EXTEN:2},,Tor)
exten => _52X.,3,Hangup()

thanks for the help

Regard
Abhi

[carpenox]

i need to see the SIP debug, once you do "asterisk -r" and your at the CLI, type "core set debug on" and then login, make a call, then show me the debug output, be sure to take out your IP and mark a few with XX like 1XX.34.6X.254 so u dont show everyone your ip

[abhiupadhyayy]

debug output

Usage: core set debug [atleast] <level> [module]
core set debug off

Sets level of debug messages to be displayed or
sets a module name to display debug messages from.
0 or off means no messages should be displayed.
Equivalent to -d[d[...]] on startup
[May 4 03:19:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 4 03:19:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 4 03:19:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 4 03:19:07] == Manager 'sendcron' logged off from 127.0.0.1
[May 4 03:19:11] == Manager 'sendcron' logged on from 127.0.0.1
[May 4 03:19:11] == Manager 'sendcron' logged off from 127.0.0.1
[May 4 03:19:43] == Manager 'sendcron' logged on from 127.0.0.1
[May 4 03:19:43] == Using SIP RTP CoS mark 5
[May 4 03:19:43] -- Called 1001
[May 4 03:19:43] -- SIP/1001-00000000 is ringing
[May 4 03:19:43] > 0x561fce8b2c10 -- Strict RTP learning after remote address set to: xx2.x68.0.xxx:1370
[May 4 03:19:43] -- SIP/1001-00000000 answered
[May 4 03:19:43] -- Executing [8600051@default:1] MeetMe("SIP/1001-00000000", "8600051,F") in new stack
[May 4 03:19:43] -- Created MeetMe conference 1023 for conference '8600051'
[May 4 03:19:43] -- <SIP/1001-00000000> Playing 'conf-onlyperson.gsm' (language 'en')
[May 4 03:19:43] > 0x561fce8b2c10 -- Strict RTP switching to RTP target address xx2.x68.0.xxx:1370 as source
[May 4 03:19:44] == Manager 'sendcron' logged off from 127.0.0.1
[May 4 03:19:48] > 0x561fce8b2c10 -- Strict RTP learning complete - Locking on source address xx2.x68.0.xxx:1370
[May 4 03:20:05] == Manager 'sendcron' logged on from 127.0.0.1
[May 4 03:20:05] == Manager 'sendcron' logged off from 127.0.0.1
[May 4 03:20:05] == Manager 'sendcron' logged on from 127.0.0.1
[May 4 03:20:05] == Manager 'sendcron' logged off from 127.0.0.1
[May 4 03:20:10] == Manager 'sendcron' logged on from 127.0.0.1
[May 4 03:20:10] == Manager 'sendcron' logged off from 127.0.0.1
[May 4 03:20:30] == Manager 'sendcron' logged on from 127.0.0.1
[May 4 03:20:30] -- Called 8600051@default
[May 4 03:20:30] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000000;2", "8600051,F") in new stack
[May 4 03:20:30] -- Local/8600051@default-00000000;1 answered
[May 4 03:20:30] -- Executing [5219403xx0252@default:1] AGI("Local/8600051@default-00000000;1", "agi://127.0.0.1:4577/call_log") in new stack
[May 4 03:20:30] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=0001))
[May 4 03:20:30] -- <Local/8600051@default-00000000;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 4 03:20:30] -- Executing [5219403xx0252@default:2] Dial("Local/8600051@default-00000000;1", "SIP/SipProviderAC/19403xx0252,,Tor") in new stack
[May 4 03:20:30] == Using SIP RTP CoS mark 5
[May 4 03:20:30] -- Called SIP/SipProviderAC/19403xx0252
[May 4 03:20:31] == Manager 'sendcron' logged off from 127.0.0.1
[May 4 03:20:32] > 0x7fcfb8042150 -- Strict RTP learning after remote address set to: 1xx.16.1xx.2xx:34066
[May 4 03:20:32] -- SIP/SipProviderAC-00000001 is making progress passing it to Local/8600051@default-00000000;1
[May 4 03:20:40] -- SIP/SipProviderAC-00000001 answered Local/8600051@default-00000000;1
[May 4 03:20:40] -- Channel SIP/SipProviderAC-00000001 joined 'simple_bridge' basic-bridge <513ebe5a-9e1c-496c-85ae-4211ea73ddcc>
[May 4 03:20:40] -- Channel Local/8600051@default-00000000;1 joined 'simple_bridge' basic-bridge <513ebe5a-9e1c-496c-85ae-4211ea73ddcc>
[May 4 03:21:02] -- Channel SIP/SipProviderAC-00000001 left 'simple_bridge' basic-bridge <513ebe5a-9e1c-496c-85ae-4211ea73ddcc>
[May 4 03:21:02] -- Channel Local/8600051@default-00000000;1 left 'simple_bridge' basic-bridge <513ebe5a-9e1c-496c-85ae-4211ea73ddcc>
[May 4 03:21:02] == Spawn extension (default, 5219403xx0252, 2) exited non-zero on 'Local/8600051@default-00000000;1'
[May 4 03:21:02] -- Executing [h@default:1] AGI("Local/8600051@default-00000000;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----31-----31-----SIP 200 OK)") in new stack
[May 4 03:21:02] -- <Local/8600051@default-00000000;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... 31-----SIP 200 OK) completed, returning 0
[May 4 03:21:02] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000000;2'
[May 4 03:21:02] WARNING[2731][C-00000001]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[May 4 03:21:02] -- Executing [h@default:1] AGI("Local/8600051@default-00000000;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[May 4 03:21:02] -- <Local/8600051@default-00000000;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0

[carpenox]

ok so its making the call to the number but u cant hear it, i would guess ti has something to do with the RTP which is how the sound goes back and forth. in that output its using a very high udp port 34066, do u have that port open for udp? what softphone are you using? try opening UDP ports 8000-35000 and see if that helps

[Kabis]

What is your RTPstart and rtpend value in /etc/asterisk/rtp.conf

[carpenox]

yea tru u can set it to 7000 and 7100 and then just open those ports UDP

[abhiupadhyayy]

What is your RTPstart and rtpend value in /etc/asterisk/rtp.conf

this rtp.conf output

; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
; Defaults are rtpstart=5000 and rtpend=31000
;
rtpstart=10000
rtpend=20000
;
; Whether to enable or disable UDP checksums on RTP traffic
;
;rtpchecksums=no
;
; The amount of time a DTMF digit with no 'end' marker should be
; allowed to continue (in 'samples', 1/8000 of a second)
;
;dtmftimeout=3000
; rtcpinterval = 5000 ; Milliseconds between rtcp reports
;(min 500, max 60000, default 5000)
;
"rtp.conf" 141L, 5660C

Thanks for support


Regards
Abhi

[abhiupadhyayy]

yea tru u can set it to 7000 and 7100 and then just open those ports UDP

Do you mean rtpstart and rtpend ?

[Kabis]

Do you use firewall? If yes, what are the ports you opened? Do you use agent screen via network or with vpn or public ip?

[carpenox]

yes rtpstart and end