GOautodial Open Source Omni-channel Contact Center Suite

Hi,

Please post following information.

1. Complete Carrier Settings (Registration String, Account Entry and Dialplan Entry)
2. Asterisk Cli logs during call.
3. sip show registry

Thanks

Wittie Manansala wrote:

Hi,

Please post following information.

1. Complete Carrier Settings (Registration String, Account Entry and Dialplan Entry)
2. Asterisk Cli logs during call.
3. sip show registry

Thanks

thank you for reply

1. this is my Carrier Settings for GOAV3 and i test it in VICIBOX9 it work.

Call In

* Registration String:* 
is empty
*Account Entry*
[AppelEntrant]
disallow=all
allow=alaw
allow=ulaw
allow=g729
type=friend
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=port,invite
nat=yes
host=[providerIPIn]

* Globals String:*
LYNIN=SIP/AppelEntrant
* Dialplan Entry:* 
exten => 33[numPhone],1,AGI(agi://127.0.0.1:4577/call_log)
exten => 33[numPhone],2,Dial(SIP/${EXTEN:10}@LYNIN,,tTo)
exten => 33[numPhone],3,Hangup

call Out

* Registration String:* 
is empty
*Account Entry*
[AppelSortant]
type=friend
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=very
nat=yes
host=[providerIPOut]
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=g729

* Globals String:* 
TSIPLYN=SIP/AppelSortant
exten => _0033[1-9]XXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _0033[1-9]XXXXXXXX,2,Dial(${TSIPLYN}/${EXTEN},,tTo)
exten => _0033[1-9]XXXXXXXX,3,Hangup

exten => _00330[1-9]XXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _00330[1-9]XXXXXXXX,2,Dial(${TSIPLYN}/0033${EXTEN:5},,tTo)
exten => _00330[1-9]XXXXXXXX,3,Hangup

1. Asterisk Cli logs during call.

Connected to Asterisk 13.17.2-vici currently running on goa (pid = 2753)
[Nov  1 16:06:45] NOTICE[5237][C-00000000]: chan_sip.c:23990 handle_response_invite: Failed to authenticate on INVITE to '"S1911011606458600051" <sip:5164536886@IPServer:5070>;tag=as2ed4aa4c'
[Nov  1 16:07:20] WARNING[5667][C-00000001]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Nov  1 16:07:20] WARNING[5666][C-00000002]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel

1. sip show registry

Host dnsmgr Username Refresh State Reg.Time
0 SIP registrations.

my service provider

Hi,

Can you post your sip show peers output from your asterisk CLI. Can you also verify to your VoIP carrier if your account is on IP Authentication or SIP registration.

Failed to authenticate on INVITE to '"S1911011606458600051" <sip:5164536886@IPServer:5070>;tag=as2ed4aa4c'

1. Check if its IP authenticated or registration.
2. Make sure not to use the default caller ID 5164536886 as it has been blocked on most carriers.

Levy Ryan Nolasco wrote:

Hi,

Can you post your sip show peers output from your asterisk CLI. Can you also verify to your VoIP carrier if your account is on IP Authentication or SIP registration.

[...]

1. Check if its IP authenticated or registration.
2. Make sure not to use the default caller ID 5164536886 as it has been blocked on most carriers.

thank you for reply
1. Check if its IP authenticated or registration.
it is IP authenticated.

2. Make sure not to use the default caller ID 5164536886 as it has been blocked on most carriers.
i did modify caller ID and it same problem.

chan_sip.c:23990 handle_response_invite: Failed to authenticate on INVITE to ....

> systemctl status kamailio

Nov 06 10:46:21 xxx.xxxxxxxxxxxxx.xxx /usr/sbin/kamailio[7197]: ERROR: <core> [core/parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0)
Nov 06 10:46:21 xxx.xxxxxxxxxxxxx.xxx /usr/sbin/kamailio[7197]: ERROR: <core> [core/parser/msg_parser.c:671]: parse_msg(): ERROR: parse_msg: message=<PING>
Nov 06 10:46:21 xxx.xxxxxxxxxxxxx.xxx /usr/sbin/kamailio[7197]: ERROR: <core> [core/parser/msg_parser.c:330]: parse_headers(): bad header field [(null)]

Hey, have same problem as people above.

sip show peers:

Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
USERNAME               IP                              Yes        Yes         A  5060     OK (3 ms)
USERNAME             IP                              Yes        Yes            5060     OK (17 ms)
kamailio                  MY SERVER IP                               Yes        Yes            5060     OK (1 ms)

sip show registry:

Host                                    dnsmgr Username       Refresh State                Reg.Time
sip host:5060                        Y      my username        105 Registered           Wed, 06 Nov 2019 11:00:06

Here is log from asterisk when I try to login into dialer:

    -- Called 99998585510911@default
    -- Executing [99998585510911@default:1] Dial("Local/99998585510911@default-00000000;2", "SIP/8585510911@kamailio,,tTo") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/8585510911@kamailio
[Nov  6 10:52:04] NOTICE[29358][C-00000000]: chan_sip.c:23990 handle_response_invite: Failed to authenticate on INVITE to '"S1911061052048600051" <sip:1464536321@My server IP:5070>;tag=as33e9f500'
    -- SIP/kamailio-00000000 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [99998585510911@default:2] Hangup("Local/99998585510911@default-00000000;2", "") in new stack
  == Spawn extension (default, 99998585510911, 2) exited non-zero on 'Local/99998585510911@default-00000000;2'
    -- Executing [h@default:1] AGI("Local/99998585510911@default-00000000;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CONGESTION---------------SIP 407 Proxy Authentication Required)") in new stack
  == Manager 'listencron' logged on from 127.0.0.1
    -- <Local/99998585510911@default-00000000;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CONGESTION---------------SIP 407 Proxy Authentication Required) completed, returning 0

Pawel Duda wrote:

Here is log from asterisk when I try to login into dialer:

[...]

Your issue is not the same. It is on your kamailio settings. Please search the forum or make a new post.