I have newly set up Asterisk 13.0.1 with PJSIP.
The pbx box is located behind a firewalling router, it is a DS-Lite setup (means the NAT-ing is done on providerside) additionally I have set up iptables with a strict policy of not allowing unknown hosts, with the exemption of some open port for incoming RTP packets.
I managed at last to have configuration where I can dial out to my SIP provider (at least to one).
The call is set up and seems to be ok.
I receive the following messages when setting up the call:
[Dec 18 12:15:31] WARNING[9533] pjsip: tsx0x1237c2c .......Temporary failure in sending Request msg INVITE/cseq=18615 (tdta0x1207af8), will try next server: Unsuitable transport selected (PJSIP_ETPNOTSUITABLE)
[Dec 18 12:48:46] WARNING[12553] pjsip: tsx0x1230644 .......Temporary failure in sending Request msg INVITE/cseq=3665 (tdta0xb3378e90), will try next server: Unsuitable transport selected (PJSIP_ETPNOTSUITABLE)
[Dec 18 12:49:12] WARNING[12581][C-00000013] app_dial.c: Unable to write frametype: 2
The transport and endpoint settings look like this:
[udp-nat]
type=transport
protocol=udp
bind=0.0.0.0:5060
local_net=192.168.178.0/24
[code]endpoint
type=endpoint
transport=udp-nat
context=ExternalSets
disallow=all
allow=alaw
allow=ulaw
callerid=
direct_media=no
disable_direct_media_on_nat=yes
force_rport=yes
ice_support=yes
rewrite_contact=yes
provider
;context=ExternalSets
outbound_auth=provider
aors=provider[/code]
Any idea what the reason for that message(s) might be and how I could prevent them?
Addendum:
When turning on debug 5 and verbose 5 this message shows up:
[Dec 19 19:42:23] DEBUG[9533] pjsip: endpoint .......Request msg INVITE/cseq=14948 (tdta0x1293448) exceeds UDP size threshold (1300), sending with TCP
[Dec 19 19:42:23] WARNING[9533] pjsip: tsx0x12944b4 .......Temporary failure in sending Request msg INVITE/cseq=14948 (tdta0x1293448), will try next server: Unsuitable transport selected (PJSIP_ETPNOTSUITABLE)
I am even more confused, as I have TCP not switched on as a transport (at last I did not intentionally) and it seems that still UDP is used, the next message is:
[code][Dec 19 19:42:23] VERBOSE[9533] res_pjsip_logger.c: <— Transmitting SIP request (1397 bytes) to UDP:aaa.bbb.cc.1dd:5060 —>
INVITE sip:01776804900@:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.99:5060;rport;branch=z9hG4bKPj5c0db9ad-d1ba-4b8f-93be-34a03436a98e
From: <sip:@127.0.1.1>;tag=f67de967-d35f-4cfc-ba83-bd18843fca16
To: <sip:@>
Contact: sip:df160b99-adfc-4035-8409-d55adf993a2a@192.168.178.99:5060
Call-ID: 124f371f-8b7f-4479-83d0-2c6333a7223e
CSeq: 14948 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Proxy-Authorization: Digest username="", realm=“127.0.1.1”, nonce=“VJRyulSUcY5Vs6gaPIrnFL4O6Chat49R”, uri=“sip::5060”, response="cc09fdc905cac13ab80a373f84e08e97"
Content-Type: application/sdp
Content-Length: 519
v=0
o=- 1614771262 1614771262 IN IP4 192.168.178.99
s=Asterisk
c=IN IP4 192.168.178.99
t=0 0
m=audio 14114 RTP/AVP 9 8 0 101
a=ice-ufrag:76d6baa44170f54f0d1db2b74f85e137
a=ice-pwd:03403bf12fafbccc3e573bff17a1573d
a=candidate:Hc0a8b263 1 UDP 2130706431 192.168.178.99 14114 typ host
a=candidate:Hc0a8b263 2 UDP 2130706430 192.168.178.99 14115 typ host
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv[/code]
