vicidial.org

Hi, I just installed new Vicidial with the help of 'The Ray Solomon.' It's installed properly but when I'm testing the inbound, it's not working and call is being disconnected instant. I also tried to investigate but still no success.

So can you please help me out to fix it.

Please note, this dialer putting in LAN network and having IP 172.17.0.34.

Below are the configuration details of dialer.

asterisk-16.17.0-vici
CentOS Linux release 7.9.2009 (Core)

cat /usr/src/astguiclient/trunk/version
2.14b0.5

SIP Peers configuration.

register => FLIGHT-TESTING:ik0XXXXX@149.XX.XXX.XXX

localhost*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
149.XX.XXX.XXX:5060 N FLIGHT-TESTI 45 Registered Wed, 10 Aug 2022 03:30:14

[FLIGHT-TESTING]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=FLIGHT-TESTING
secret=ik04XXXX
host=149.XX.XXX.XXX
dtmfmode=rfc2833
qualify= yes
nat=force_rport,comedia
context=trunkinbound
insecure=invite

CLI Logs
localhost*CLI> sip set debug on
SIP Debugging enabled
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1

<--- SIP read from UDP:149.XX.XXX.XXX:5060 --->
INVITE sip:12222222222@172.17.0.34:5060 SIP/2.0
CSeq: 1 INVITE
Via: SIP/2.0/UDP 149.XX.XXX.XXX:5060;branch=z9hG4bK01052114483110896169171
From: <sip:13232XXXXXX@149.XX.XXX.XXX:5060>;tag=05012196169171
Call-ID: 09d5bb6cd366ce8d9439a337d707c8e0d7346a6 ... XX.XXX.XXX
To: <sip:12222222222@172.17.0.34>
Contact: <sip:149.XX.XXX.XXX:5060;transport=udp>
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, INFO, NOTIFY, MESSAGE, SUBSCRIBE, REFER, PUBLISH
Content-Type: application/sdp
Content-Length: 208
Max-Forwards: 70

v=0
o=- 186593360 96169156 IN IP4 149.XX.XXX.XXX
s=VoipSIP
c=IN IP4 149.XX.XXX.XXX
t=0 0
m=audio 6426 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (11 headers 10 lines) ---
Sending to 149.XX.XXX.XXX:5060 (NAT)
Using INVITE request as basis request - 09d5bb6cd366ce8d9439a337d707c8e0d7346a6 ... XX.XXX.XXX
Found peer 'FLIGHT-TESTING' for '13232XXXXXX' from 149.XX.XXX.XXX:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 149.XX.XXX.XXX:6426
Looking for 12222222222 in trunkinbound (domain 172.17.0.34)
list_route: hop: <sip:149.XX.XXX.XXX:5060;transport=udp>

can you post the asterisk cli log without sip debug .. (sip set debug off)
type asterisk -vvvvvvvvr
and dial your inbound copy paste the log.

striker wrote:can you post the asterisk cli log without sip debug .. (sip set debug off)
type asterisk -vvvvvvvvr
and dial your inbound copy paste the log.

Please find the asterisk CLI logs

localhost*CLI>
== Using SIP RTP CoS mark 5
-- Executing [12222222222@trunkinbound:1] AGI("SIP/FLIGHT-TESTING-00000000", "agi-DID_route.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
-- <SIP/FLIGHT-TESTING-00000000>AGI Script agi-DID_route.agi completed, returning 0
-- Executing [12222222222@trunkinbound:2] Hangup("SIP/FLIGHT-TESTING-00000000", "") in new stack
== Spawn extension (trunkinbound, 12222222222, 2) exited non-zero on 'SIP/FLIGHT-TESTING-00000000'
-- Executing [h@trunkinbound:1] AGI("SIP/FLIGHT-TESTING-00000000", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- <SIP/FLIGHT-TESTING-00000000>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Using SIP RTP CoS mark 5
-- Executing [12222222222@trunkinbound:1] AGI("SIP/FLIGHT-TESTING-00000001", "agi-DID_route.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
-- <SIP/FLIGHT-TESTING-00000001>AGI Script agi-DID_route.agi completed, returning 0
-- Executing [12222222222@trunkinbound:2] Hangup("SIP/FLIGHT-TESTING-00000001", "") in new stack
== Spawn extension (trunkinbound, 12222222222, 2) exited non-zero on 'SIP/FLIGHT-TESTING-00000001'
-- Executing [h@trunkinbound:1] AGI("SIP/FLIGHT-TESTING-00000001", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- <SIP/FLIGHT-TESTING-00000001>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
localhost*CLI>

seems DID lands fine under trunkinbound context.

can share the DID settings in vicidial

what you have set for DID route:?

striker wrote:seems DID lands fine under trunkinbound context.

can share the DID settings in vicidial

what you have set for DID route:?

DID: 12222222222>>>DID Route: IN_GROUP>>>User Route Settings In-Group:/In-Group ID: Ingroup created with same DID extension.

M actually unable to attach the screenshot else i can show you.

Please advise...!!!

use imgbb.com to upload images to and paste here

carpenox wrote:use imgbb.com to upload images to and paste here

DID Configuration
https://ibb.co/M13Cwb7

SIP Response 603-declined from dialer.
https://ibb.co/Pr6kdvJ

Please advise, it's too urgent for me.

Ikram_Ali wrote:Please advise, it's too urgent for me.

@williamconley@mflorell

did you add the carrier IP to the trusted zone for the firewall? or add it to the ViciWhite Ip list in the GUI?

carpenox wrote:did you add the carrier IP to the trusted zone for the firewall? or add it to the ViciWhite Ip list in the GUI?

Thanks for your reply Carpenox.

Yes, i created the IP peer in sip.conf and also created the carrier, admin-->carrier.

Should i add the ip in somewhere else in vicidial.

add it in the firewall

firewall-cmd --zone=trusted --permanent --add-source=2.2.2.2 ( <---put carrier ip there)

carpenox wrote:add it in the firewall

firewall-cmd --zone=trusted --permanent --add-source=2.2.2.2 ( <---put carrier ip there)

[root@R2-U24-SRV ~]# firewall-cmd --zone=trusted --permanent --add-source=149.xx.xxx.xxx
FirewallD is not running

Got this because FirewallD is not running.

Anything else, this is so weird issue, can't understand what is the issue.

Ikram_Ali wrote:

carpenox wrote:add it in the firewall

firewall-cmd --zone=trusted --permanent --add-source=2.2.2.2 ( <---put carrier ip there)

[root@R2-U24-SRV ~]# firewall-cmd --zone=trusted --permanent --add-source=149.xx.xxx.xxx
FirewallD is not running

Got this because FirewallD is not running.

Anything else, this is so weird issue, can't understand what is the issue.

Please advise...If I missed something or did a mistake.

ok so we can eliminate that as a possible cause, back to what striker said, the CLI outout. Once you go into "asterisk -r" just take a screen shot of that area while you test a call and show us here

https://ibb.co/GPFm7Yb

CLI Logs:

== Using SIP RTP CoS mark 5
> 0x7f527000e0d0 -- Strict RTP learning after remote address set to: 149.xx.xx.xx:6314
-- Executing [12222222222@trunkinbound:1] AGI("SIP/149.xx.xx.xx-00000003", "agi-DID_route.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
-- <SIP/149.xx.xx.xx-00000003>AGI Script agi-DID_route.agi completed, returning 0
-- Executing [12222222222@trunkinbound:2] Hangup("SIP/149.xx.xx.xx-00000003", "") in new stack
== Spawn extension (trunkinbound, 12222222222, 2) exited non-zero on 'SIP/149.xx.xx.xx-00000003'
[Aug 25 09:02:41] WARNING[8886][C-00000005]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
-- Executing [h@trunkinbound:1] AGI("SIP/149.xx.xx.xx-00000003", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
-- <SIP/149.xx.xx.xx-00000003>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0
R2-U24-SRV*CLI>

Please note, there are 9 sockets not 10. I tried a lot to get 10 sockets but "ASTconf3way" socket not coming anyhow, any advise for this too, although, i know, this is not the issue for inbound calls.

screen -ls
There are screens on:
2318.ASTfastlog (Detached)
2315.ASTVDadapt (Detached)
2312.ASTVDremote (Detached)
2309.ASTVDauto (Detached)
2306.ASTlisten (Detached)
2303.ASTsend (Detached)
2300.ASTupdate (Detached)
2151.asterisk (Detached)
2146.astshell20220825075718 (Detached)
9 Sockets in /var/run/screen/S-root.

seems all good in your setup.

as per log calls landing to Trunkinbound contetxt and getting hangup

Try this
IN DID set the DID Route to PHone, and enter any one of the extension registered in softphone and dial in and check.

Thanks for your reply Striker,
I'll check and let you know.

Ikram_Ali wrote:Thanks for your reply Striker,
I'll check and let you know.

I had to re-installed from scratch again and issue resolved, so i missed something earlier.