Failed to authenticate on INVITE - SIP/2.0 403 Forbidden

I have two telephony servers running on the same subnet, TS1 (172.19.57.42/255.255.255.252.0) and TS2(172.19.57.44/255.255.255.252.0) Asterisk running and a SIP trunk (totelephony) between them.

There are no problems when dialling SIP extensions belonging to the same TS but I cannot call SIP extensions from TS2 to TS1 or in the other way around. During this test I used: 4755 extension registered in TS1 and 4811 registered in TS2 and tried to make a call from 4811 to 4755, here is the outcome:

On both sites, TS1 and TS2 in SIP.conf:
[totelephony]
type=peer
host=172.19.57.44 -> in TS1
host=172.19.57.42 -> in TS2
context=dialler
disallow=all
allow=alaw
canreinvite=no
insecure=port,invite

On TS1 in sip.conf
[4755]
type=friend
secret=4811
port=5060
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
disallow=all
allow=alaw

On TS2 in sip.conf:
[4811]
type=friend
secret=4811
port=5060
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
disallow=all
allow=alaw

On both sites, TS1 and TS2, in extensions.conf
[from-internal]
exten => _554XXX,1,Dial(SIP/totelephony/${EXTEN:2})
exten => _554XXX,2,Hangup()

exten => _4XXX,1,Dial(SIP/${EXTEN})
exten => _4XXX,2,Hangup()

SIP logs:

[Jun 16 11:53:11] VERBOSE[13585] logger.c:
<— SIP read from 172.19.41.195:5060 —>
INVITE sip:554755@172.19.57.44;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.19.41.195:5060;branch=z9hG4bK-d8754z-2463e75d7bf86691-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:4811@172.19.41.195:5060;transport=UDP
To: sip:554755@172.19.57.44;transport=UDP
From: sip:4811@172.19.57.44;transport=UDP;tag=1620a86d
Call-ID: YjQ3OTI4MjIxNmQ5NDA3MmIzZmQ4YzjU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO
Content-Type: application/sdp
User-Agent: Zoiper for Windows rev.2875
Content-Length: 208

[Jun 16 11:53:11] VERBOSE[13585] logger.c:
<— Reliably Transmitting (no NAT) to 172.19.41.195:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.19.41.195:5060;branch=z9hG4bK-d8754z-2463e75d7bf86691-1—d8754z-;received=172.19.41.195;rport=5060
From: sip:4811@172.19.57.44;transport=UDP;tag=1620a86d
To: sip:554755@172.19.57.44;transport=UDP;tag=as65fb9e7d
Call-ID: YjQ3OTI4MjIxNmQ5NDA3MmIzZUzZjU.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4e769d22"
Content-Length: 0

[Jun 16 11:53:11] VERBOSE[13585] logger.c:
<— SIP read from 172.19.41.195:5060 —>
ACK sip:554755@172.19.57.44;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.19.41.195:5060;branch=z9hG4bK-d8754z-275d7bf86691-1—d8754z-;rport
To: sip:554755@172.19.57.44;transport=UDP;tag=as65f9e7d
From: sip:4811@172.19.57.44;transport=UDP;tag=162a86d
Call-ID: YjQ3OTI4MjIxNmQ5NDA3MmIzZmQ4YzMxMjYzMGUzZjU.
CSeq: 1 ACK
Content-Length: 0

[Jun 16 11:53:11] VERBOSE[13585] logger.c:
<— SIP read from 172.19.41.195:5060 —>
INVITE sip:554755@172.19.57.44;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.19.41.195:5060;branch=z9hG4bK-d8754z-96682b7d500ec-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:4811@172.19.41.195:5060;transport=UDP
To: sip:554755@172.19.57.44;transport=UDP
From: sip:4811@172.19.57.44;transport=UDP;tag=1620a86d
Call-ID: YjQ3OTI4MjIxNmQ5NDA3MmIzZmQ4YzMxMjYzMGUzZjU.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username=“4811”,realm=“asterisk”,nonce=“4e769d22”,uri="sip:554755@172.19.57.44;transport=UDP",response=“d203e3bdedc3a37f78ffddd6eb9a225e”,algorithm=MD5
User-Agent: Zoiper for Windows rev.2875
Content-Length: 208

[Jun 16 11:53:11] VERBOSE[13585] logger.c: Looking for 554755 in from-internal (domain 172.19.57.44)
[Jun 16 11:53:11] VERBOSE[13585] logger.c: list_route: hop: sip:4811@172.19.41.195:5060;transport=UDP
[Jun 16 11:53:11] VERBOSE[13585] logger.c:
<— Transmitting (NAT) to 172.19.41.195:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.19.41.195:5060;branch=z9hG4bK-d8754z-96682b32f7d500ec-1—d8754z-;received=172.19.41.195;rport=5060
From: sip:4811@172.19.57.44;transport=UDP;tag=1620a86d
To: sip:554755@172.19.57.44;transport=UDP
Call-ID: YjQ3OTI4MjIxNmQ5NDA3MmIzZmQ4YzMzMGUzZjU.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:554755@172.19.57.44
Content-Length: 0

[Jun 16 11:53:11] VERBOSE[19077] logger.c: – Executing [554755@from-internal:1] Dial(“SIP/4811-00000f40”, “SIP/totelephony/4755”) in new stack
[Jun 16 11:53:11] VERBOSE[19077] logger.c: Audio is at 172.19.57.44 port 14942
[Jun 16 11:53:11] VERBOSE[19077] logger.c: Adding codec 0x8 (alaw) to SDP
[Jun 16 11:53:11] VERBOSE[19077] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Jun 16 11:53:11] VERBOSE[19077] logger.c: Reliably Transmitting (no NAT) to 172.19.57.42:5060:
INVITE sip:4755@172.19.57.42 SIP/2.0
Via: SIP/2.0/UDP 172.19.57.44:5060;branch=z9hG4bK4fe655ba;rport
From: “device” sip:4811@172.19.57.44;tag=as54fef7f4
To: sip:4755@172.19.57.42
Contact: sip:4811@172.19.57.44
Call-ID: 1609872a73662819562bafc20875be5a@172.19.57.44
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 16 Jun 2014 09:53:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 213

[Jun 16 11:53:11] VERBOSE[13585] logger.c:
<— SIP read from 172.19.57.42:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.19.57.44:5060;branch=z9hG4bK4fe655ba;received=172.19.57.44;rport=5060
From: “device” sip:4811@172.19.57.44;tag=as54fef7f4
To: sip:4755@172.19.57.42;tag=as1c71ac32
Call-ID: 1609872a73662819562bafc20875be5a@172.19.57.44
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="290a924f"
Content-Length: 0

<------------->
[Jun 16 11:53:11] VERBOSE[13585] logger.c: — (11 headers 0 lines) —
[Jun 16 11:53:11] VERBOSE[13585] logger.c: Transmitting (no NAT) to 172.19.57.42:5060:
ACK sip:4755@172.19.57.42 SIP/2.0
Via: SIP/2.0/UDP 172.19.57.44:5060;branch=z9hG4bK4fe655ba;rport
From: “device” sip:4811@172.19.57.44;tag=as54fef7f4
To: sip:4755@172.19.57.42;tag=as1c71ac32
Contact: sip:4811@172.19.57.44
Call-ID: 1609872a73662819562bafc20875be5a@172.19.57.44
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


[Jun 16 11:53:11] NOTICE[13585] chan_sip.c: Failed to authenticate on INVITE to ‘“device” sip:4811@172.19.57.44;tag=as54fef7f4’
[Jun 16 11:53:11] VERBOSE[19077] logger.c: – SIP/totelephony-00000f41 is circuit-busy
[Jun 16 11:53:11] VERBOSE[19077] logger.c: == Everyone is busy/congested at this time (1:0/1/0)
[Jun 16 11:53:11] VERBOSE[19077] logger.c: – Executing [554755@from-internal:2] Hangup(“SIP/4811-00000f40”, “”) in new stack
[Jun 16 11:53:11] VERBOSE[13585] logger.c: Really destroying SIP dialog ‘1609872a73662819562bafc20875be5a@172.19.57.44’ Method: INVITE
[Jun 16 11:53:11] VERBOSE[19077] logger.c: == Spawn extension (from-internal, 554755, 2) exited non-zero on ‘SIP/4811-00000f40’
[Jun 16 11:53:11] VERBOSE[19077] logger.c: Scheduling destruction of SIP dialog ‘YjQ3OTI4MjIxNmQ5NDA3MmIzZmQ4YzYzMGUzZjU.’ in 32000 ms (Method: INVITE)
[Jun 16 11:53:11] VERBOSE[19077] logger.c:

<— Reliably Transmitting (NAT) to 172.19.41.195:5060 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.19.41.195:5060;branch=z9hG4bK-d8754z-96682b32f7d500ec-1—d8754z-;received=172.19.41.195;rport=5060
From: sip:4811@172.19.57.44;transport=UDP;tag=1620a86d
To: sip:554755@172.19.57.44;transport=UDP;tag=as7111b7f7
Call-ID: YjQ3OTI4MjIxNmQ5NDA3ZmQ4YzMxMjYzMGUzZjU.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

I know I am missing something but I cannot find what and right now I am a bit loss. Any help would be greatly appreciated. Thanks.