Inbound calls not working

[Bajamark]

Hi I'm new to vicidial and I'm trying to get inbound calls working, I got outbound just fine, any help would be appreciated.

Thank you.

[Bajamark]

This is what im getting with sip debug

[Sep 23 05:06:58] --- (17 headers 16 lines) ---
[Sep 23 05:06:58] Sending to 64.2.142.90:5060 (NAT)
[Sep 23 05:06:58] Using INVITE request as basis request - 10a6fb59438a995e4ed95f8501f60756@64.2.142.229
[Sep 23 05:06:58] No matching peer for '7147844990' from '64.2.142.90:5060'
[Sep 23 05:06:58] NOTICE[2338]: chan_sip.c:23412 handle_request_invite: Failed to authenticate device "7147844990" <sip:7147844990@64.2.142.229>;tag=as656935d5
[Sep 23 05:06:58]
<--- Reliably Transmitting (NAT) to 64.2.142.90:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 64.2.142.90;branch=z9hG4bKc96c.3b7964b6.0;received=64.2.142.90;rport=5060
Via: SIP/2.0/UDP 64.2.142.229:5060;received=64.2.142.229;branch=z9hG4bK116c3a34;rport=5060
From: "7147844990" <sip:7147844990@64.2.142.229>;tag=as656935d5
To: <sip:6192596327@192.168.1.64>;tag=as3dfc0ee5
Call-ID: 10a6fb59438a995e4ed95f8501f60756@64.2.142.229
CSeq: 103 INVITE
Server: Asterisk PBX 1.8.23.0-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

[geoff3dmg]

There's something wrong in your carrier settings. Asterisk isn't matching up the incoming call as coming from them.

[Bajamark]

Hi,

This is my carrier settings

[vitel-inbound]
type=friend
dtmfmode=auto
host=xxxxxx
context=trunkinbound
allow=all
insecure=port,invite
canreinvite=no

[vitel-outbound]
type=friend
dtmfmode=auto
host=xxxxx
allow=all
canreinvite=no

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(SIP/vitel-outbound/${EXTEN:1},,To)
exten => _91NXXNXXXXXX,3,Hangup

[geoff3dmg]

Why two entries?

[Bajamark]

for the inbound and outbound

[geoff3dmg]

There's no need to do that.

[Bajamark]

Ok, So I got the inbound working, if I use context=trunkinbound the call gets recived but the call drops after 6 seconds, this is what I get on cli

WARNING[2335]: chan_sip.c:3983 retrans_pkt: Retransmission timeout reached on transmission 43fd457c5cef0bc645df6c515ca15702@64.2.142.102 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 6400ms with no response
[Oct 1 14:07:53] WARNING[2335]: chan_sip.c:4012 retrans_pkt: Hanging up call 43fd457c5cef0bc645df6c515ca15702@64.2.142.102 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/ ... nsmissions).

But I talked to My VOIP provider and they provided me with this context=from-trunk

With this context the call does not go through at all but I get this on the CLI

NOTICE[2335]: chan_sip.c:23510 handle_request_invite: Call from 'vitel-inbound2' (64.2.142.90:5060) to extension '7142020439' rejected because extension not found in context 'from-trunk'.

With the first context=trunkinbound My VOIP provider says that the call is being rejected, but if I try the context=from-trunk they told me the call was accepted
Any ideas?

[geoff3dmg]

context=trunkinbound is what you need to use

Code: Select all

WARNING[2335]: chan_sip.c:3983 retrans_pkt: Retransmission timeout reached on transmission 43fd457c5cef0bc645df6c515ca15702@64.2.142.102 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions

Ok, this is an issue with the RTP stream (the audio). This is very likely to be a port forwarding issue on your firewall. You did forward all the ports you need for Asterisk to work correctly?

[Bajamark]

Yes, I Have port forwarded all the the UDP ports to my server I have a netgear router R6300, but the thing is that MY VOIP say that with trunkinbound the call gets rejected, and when I try from-trunk the call does go through but get the extension not found error

[geoff3dmg]

Your firewall/router is not letting the RTP packets in and asterisk is terminating the call due to no audio. So yes, your VOIP provider would see rejected calls. If you set your inbound context to 'from-trunk' then yes, you will get an 'extension not found' error, because it does not exist. The audio isn't even sent at this point, so you would never see any RTP errors.

Fix your firewall/portforwarding. I'd try a different router personally. The R6300 is designed for home use.

Also you did change your externip setting in sip.conf didn't you?

[Bajamark]

I have this
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
;t38pt_udptl = yes ; Default false
;register => 1234:password@mysipprovider.com
;registertimeout=20 ; retry registration calls every 20 seconds (default)
;registerattempts=10 ; Number of registration attempts before we give up
externip = 201.170.238.77 ; Address that we're going to put in outbound SIP
;externhost=test.test.com ; Alternatively you can specify a domain
;externrefresh=10 ; How often to refresh externhost if
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


And No I don't think that is my global IP, I thought it changed automatically, Ishould change it to my current IP, correct?

[Bajamark]

geoff3dmg Thank you, very much, I changed the externip to my IP restarted and the call does not get cut, It was the sip.conf all this time.
Thank you