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