Can't stablish pjsip trunk without auth

I’m trying to move a chan_sip trunk to pjsip without auth.
I’ve tryed with auth too but without lucky.

I can’t figure out what’s wrong once 488 msg didn’t point me to the right direction.

old sip.conf

[general]
bindport=5060
bindaddr=10.10.0.105
tcpenable=yes
tcpbindaddr=10.10.0.105
dtmfmode=rfc2833
disallow=all
allow=alaw
nat=force_rport,comedia
qualify=no
reinvite=no
canreinvite=no
insecure=port,invite
transport=udp,tcp
allowguest=no
autocreatepeer=no
srvlookup=yes
;rtptimeout=30
useragent="NodeGw - v0.0.1"
callcounter=yes
localnet=10.10.0.0/255.255.255.0

[FREESWITCH](!)
type=friend
insecure=port,invite
nat=force_rport,comedia
disallow=all
allow=alaw,ulaw
canreinvite=no
qualify=yes
context=FROM-DISCADOR
deny=0.0.0.0/0.0.0.0
permit=10.10.0.0/255.255.255.0

[SRV1](FREESWITCH)
host=10.10.0.103

pjsip.conf

[global]
type = global
user_agent = "NodeGw - v0.0.1"

[transport-udp]
type = transport
protocol = udp
bind = 10.10.0.105:5060
local_net = 10.10.0.0/255.255.255.0

[transport-tcp]
type = transport
protocol = tcp
bind = 10.10.0.105
local_net = 10.10.0.0/255.255.255.0

[SRV1]
type = aor
contact = sip:SRV1@10.10.0.103
max_contacts=10

[SRV1]
type = identify
endpoint = SRV1
match = 10.10.0.0/24
srv_lookups = false

[SRV1]
type = endpoint
transport = transport-tcp
context = FROM-DISCADOR
disallow = all
allow = ulaw
allow = alaw
aors = SRV1
direct_media = no

[acl]
type = acl
permit = 10.10.0.0/255.255.255.0
deny = 0.0.0.0/0.0.0.0

pcap

INVITE sip:00171908338397@10.10.0.105;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.0.103:5080;branch=z9hG4bK5pceN8S8Z6m4S
Max-Forwards: 70
From: <sip:SRV1@10.10.0.105>;tag=56QjejZKZy0Bc
To: <sip:00171908338397@10.10.0.105>
Call-ID: 7f2b18f7-eb7f-123b-5199-eab49573deaf
CSeq: 60362634 INVITE
Contact: <sip:gw+5cd97303-fa1f-427b-a65f-bd28dd2b4d26@10.10.0.103:5080;transport=tcp;gw=5cd97303-fa1f-427b-a65f-bd28dd2b4d26>
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 266
X-Maestro-CustomerId: 21154104
X-Maestro-CustomerPhoneNumber: 71908338397
X-Maestro-CustomerExtraInfo: {"valor":53204.93
X-Maestro-MailingId: 1
X-FS-Support: update_display,send_info
Remote-Party-ID: <sip:0000000000@10.10.0.105>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1669814294 1669814295 IN IP4 10.10.0.103
s=FreeSWITCH
c=IN IP4 10.10.0.103
t=0 0
m=audio 18558 RTP/AVP 0 8 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:13 CN/8000
a=ptime:20

SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.10.0.103:5080;received=10.10.0.103;branch=z9hG4bK5pceN8S8Z6m4S
Call-ID: 7f2b18f7-eb7f-123b-5199-eab49573deaf
From: <sip:SRV1@10.10.0.105>;tag=56QjejZKZy0Bc
To: <sip:00171908338397@10.10.0.105>
CSeq: 60362634 INVITE
Server: "NodeGw - v0.0.1"
Content-Length:  0

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/TCP 10.10.0.103:5080;received=10.10.0.103;branch=z9hG4bK5pceN8S8Z6m4S
Call-ID: 7f2b18f7-eb7f-123b-5199-eab49573deaf
From: <sip:SRV1@10.10.0.105>;tag=56QjejZKZy0Bc
To: <sip:00171908338397@10.10.0.105>;tag=c86e9806-09e9-47f0-89a6-8b7036e094dc
CSeq: 60362634 INVITE
Server: "NodeGw - v0.0.1"
Content-Length:  0

ACK sip:00171908338397@10.10.0.105;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.0.103:5080;branch=z9hG4bK5pceN8S8Z6m4S
Max-Forwards: 70
From: <sip:SRV1@10.10.0.105>;tag=56QjejZKZy0Bc
To: <sip:00171908338397@10.10.0.105>;tag=c86e9806-09e9-47f0-89a6-8b7036e094dc
Call-ID: 7f2b18f7-eb7f-123b-5199-eab49573deaf
CSeq: 60362634 ACK
Content-Length: 0

Loaded Modules ommiting funcs

Module                         Description                              Use Count  Status      Support Level
acl                            Named ACL system                         2          Running              core
app_stack.so                   Dialplan subroutines (Gosub, Return, etc 0          Running              core
bridge_builtin_features.so     Built in bridging features               1          Running              core
bridge_builtin_interval_features.so Built in bridging interval features      0          Running              core
bridge_holding.so              Holding bridge module                    0          Running              core
bridge_native_rtp.so           Native RTP bridging module               0          Running              core
bridge_simple.so               Simple two channel bridging module       0          Running              core
bridge_softmix.so              Multi-party software based channel mixin 0          Running              core
ccss                           Call Completion Supplementary Services   2          Running              core
cdr                            CDR Engine                               2          Running              core
cel                            CEL Engine                               1          Running              core
chan_bridge_media.so           Bridge Media Channel Driver              0          Running              core
chan_pjsip.so                  PJSIP Channel Driver                     0          Running              core
codec_a_mu.so                  A-law and Mulaw direct Coder/Decoder     0          Running              core
codec_adpcm.so                 Adaptive Differential PCM Coder/Decoder  0          Running              core
codec_alaw.so                  A-law Coder/Decoder                      0          Running              core
codec_g722.so                  ITU G.722-64kbps G722 Transcoder         0          Running              core
codec_g726.so                  ITU G.726-32kbps G726 Transcoder         0          Running              core
codec_gsm.so                   GSM Coder/Decoder                        0          Running              core
codec_ilbc.so                  iLBC Coder/Decoder                       0          Running              core
codec_lpc10.so                 LPC10 2.4kbps Coder/Decoder              0          Running              core
codec_resample.so              SLIN Resampling Codec                    0          Running              core
codec_speex.so                 Speex Coder/Decoder                      0          Running              core
codec_ulaw.so                  mu-Law Coder/Decoder                     0          Running              core
dnsmgr                         DNS Manager                              2          Running              core
dsp                            DSP                                      1          Running              core
enum                           ENUM Support                             2          Running              core
extconfig                      Configuration                            13         Running              core
features                       Call Features                            1          Running              core
format_g719.so                 ITU G.719                                0          Running              core
format_g723.so                 G.723.1 Simple Timestamp File Format     0          Running              core
format_g726.so                 Raw G.726 (16/24/32/40kbps) data         0          Running              core
format_g729.so                 Raw G.729 data                           0          Running              core
format_gsm.so                  Raw GSM data                             0          Running              core
format_h263.so                 Raw H.263 data                           0          Running              core
format_h264.so                 Raw H.264 data                           0          Running              core
format_ilbc.so                 Raw iLBC data                            0          Running              core
format_ogg_speex.so            OGG/Speex audio                          0          Running          extended
format_ogg_vorbis.so           OGG/Vorbis audio                         0          Running              core
format_pcm.so                  Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0          Running              core
format_siren14.so              ITU G.722.1 Annex C (Siren14, licensed f 0          Running              core
format_siren7.so               ITU G.722.1 (Siren7, licensed from Polyc 0          Running              core
format_sln.so                  Raw Signed Linear Audio support (SLN) 8k 0          Running              core
format_vox.so                  Dialogic VOX (ADPCM) File Format         0          Running          extended
format_wav.so                  Microsoft WAV/WAV16 format (8kHz/16kHz S 0          Running              core
format_wav_gsm.so              Microsoft WAV format (Proprietary GSM)   0          Running              core
http                           Built-in HTTP Server                     2          Running              core
indications                    Indication Tone Handling                 1          Running              core
logger                         Logger                                   1          Running              core
manager                        Asterisk Manager Interface               1          Running              core
pbx_config.so                  Text Extension Configuration             0          Running              core
pbx_loopback.so                Loopback Switch                          0          Running              core
pbx_spool.so                   Outgoing Spool Support                   0          Running              core
plc                            PLC                                      1          Running              core
res_pjproject.so               PJPROJECT Log and Utility Support        2          Running              core
res_pjsip.so                   Basic SIP resource                       5          Running              core
res_pjsip_endpoint_identifier_ip.so PJSIP IP endpoint identifier             0          Running              core
res_pjsip_logger.so            PJSIP Packet Logger                      0          Running              core
res_pjsip_pubsub.so            PJSIP event resource                     2          Running              core
res_pjsip_session.so           PJSIP Session resource                   2          Running              core
res_rtp_asterisk.so            Asterisk RTP Stack                       0          Running              core
res_sorcery_astdb.so           Sorcery Astdb Object Wizard              3          Running              core
res_sorcery_config.so          Sorcery Configuration File Object Wizard 13         Running              core
res_sorcery_memory.so          Sorcery In-Memory Object Wizard          2          Running              core
res_sorcery_memory_cache.so    Sorcery Memory Cache Object Wizard       0          Running              core
res_sorcery_realtime.so        Sorcery Realtime Object Wizard           0          Running              core
res_statsd.so                  StatsD client support                    1          Running          extended
sounds                         Sounds Index                             1          Running              core
udptl                          UDPTL                                    1          Running              core

You’d need res_pjsip_sdp_rtp.so loaded to support RTP negotiation in SDP. There may be other modules not loaded that are required for your specific usage.

That’s it!
Thank you @jcolp

I’m working to load the minimum modules as possible. And excluded some ones, but isn’t easy do discover what i’ve need reinclude.

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