Trunk SIP - Problème One way audio

[ddpetit]

Bonjour à tous,

J'essaie desespérement depuis plusieurs jours de faire fonctionner mon serveur Asterisk avec un trunk sip (il était auparavant branché avec un B410P et j'avais aussi galéré à l'époque pour trouver les bons paramètres). Il s'agit d'un trunk sip pris chez OVH

Malheureusement j'ai un one way audio et je n'arrive pas à trouver la solution tout seul. Le problème est le suivant : lorsque quelqu'un m'appelle, je l'entends mais il ne m'entend pas

Le paramétrage est le suivant :

SIP.conf :

[general]
nat=force_rport,comedia
externip=XXX.XXX.XXX.XXX ; Adresse IP de la ligne SERANVILLE
localnet=192.168.130.0/255.255.255.0
qualify=yes
defaultexpiry=1800 ; Temps de register de la ligne
context=trunk-ovh ; Nom du context pour le trunk dans sip.conf
directmedia=no
bindport=5060 ; Port d'ecoute.
bindaddr=0.0.0.0 
srvlookup=no ;Autoriser les appels via noms DNS
register => XXX@siptrunk.ovh.net ;Autenthification du trunk
disallow=all
allow=ulaw
allow=alaw

[trunk-ovh] 
nat=force_rport,comedia
type=friend
host=siptrunk.ovh.net
context=from-pstn
language=fr
insecure=invite,port
defaultuser=XXX
secret=XXX

[215]
type=friend
defaultuser=215
secret=XXX
callerid="Audrey SER" <215>
qualify=200        
nat=force_rport,comedia       
insecure=port  
host=dynamic                  
directmedia=no
context=appel-sortant-comptoir
language=fr
call-limit=4
busy-level=1
subscribecontext=blf 
callcounter=yes

Firewall (tout est quasiment ouvert) => pour info j'ai laissé la config de base de rtp.conf. Client étant mon réseau interne 192.168.130.X

iptables -A INPUT -p udp -m udp --dport 4569 -j ACCEPT
iptables -A INPUT -p udp -m udp --dport 5060 -j ACCEPT
iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT

iptables -A FORWARD -i client ! -d 10.0.0.0/8 -p udp --dport 5060 -m state --state NEW,ESTABLISHED -j ACCEPT
iptables -A FORWARD -o client ! -s 10.0.0.0/8 -p udp --sport 5060 -m state ! --state NEW -j ACCEPT
iptables -A FORWARD -i client ! -d 10.0.0.0/8 -p tcp --dport 5060 -m state --state NEW,ESTABLISHED -j ACCEPT
iptables -A FORWARD -o client ! -s 10.0.0.0/8 -p tcp --sport 5060 -m state ! --state NEW -j ACCEPT
iptables -A FORWARD -i client ! -d 10.0.0.0/8 -p udp --dport 5962 -m state --state NEW,ESTABLISHED -j ACCEPT
iptables -A FORWARD -o client ! -s 10.0.0.0/8 -p udp --sport 5962 -m state ! --state NEW -j ACCEPT
iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT

Je poste le debug sip en dessous

Merci d'avance pour votre aide

[ddpetit]

<--- SIP read from UDP:91.121.129.23:5060 --->
INVITE sip:s@192.168.2.100:5060;transport=udp SIP/2.0
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
Contact: <sip:10.7.1.65:5060>
Content-Type: application/sdp
CSeq: 198813419 INVITE
From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
Max-Forwards: 29
Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
To: <sip:0383720044@10.7.1.65;user=phone>
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-TKLX-87078c4d-1549530f
Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
User-Agent: Cirpack/v4.76 (gw_sip)
Content-Length: 318

v=0
o=cp10 156526349186 156526349186 IN IP4 91.121.128.136
s=SIP Call
c=IN IP4 91.121.128.136
t=0 0
m=audio 32142 RTP/AVP 8 18 0 101
b=AS:82
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (13 headers 15 lines) ---
Sending to 91.121.129.23:5060 (NAT)
Sending to 91.121.129.23:5060 (NAT)
Using INVITE request as basis request - 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
Found peer 'trunk-ovh' for '0383723596' from 91.121.129.23:5060
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 91.121.128.136:32142
Looking for s in from-pstn (domain 192.168.2.100)
sip_route_dump: route/path hop: <sip:91.121.129.23:5060;lr>

<--- Transmitting (NAT) to 91.121.129.23:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-TKLX-87078c4d-1549530f;received=91.121.129.23;rport=5060
Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
To: <sip:0383720044@10.7.1.65;user=phone>
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
CSeq: 198813419 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:s@185.246.18.202:5060>
Content-Length: 0


<------------>

<--- Transmitting (NAT) to 91.121.129.23:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-TKLX-87078c4d-1549530f;received=91.121.129.23;rport=5060
Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
CSeq: 198813419 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:s@185.246.18.202:5060>
Content-Length: 0


<------------>
Audio is at 11998
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.130.49:5060:
INVITE sip:214@192.168.130.49:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.130.254:5060;branch=z9hG4bK4374ac72;rport
Max-Forwards: 70
From: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
To: <sip:214@192.168.130.49:5060>
Contact: <sip:0383723596@192.168.130.254:5060>
Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Date: Thu, 08 Aug 2019 11:24:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 294

v=0
o=root 638829780 638829780 IN IP4 192.168.130.254
s=Asterisk PBX 13.14.1~dfsg-2+deb9u4
c=IN IP4 192.168.130.254
t=0 0
m=audio 11998 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:192.168.130.49:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.130.254:5060;branch=z9hG4bK4374ac72;rport=5060
From: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
To: <sip:214@192.168.130.49:5060>
Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
CSeq: 102 INVITE
User-Agent: Yealink SIP-T41S 66.84.0.15
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.130.49:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.130.254:5060;branch=z9hG4bK4374ac72;rport=5060
From: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
To: <sip:214@192.168.130.49:5060>;tag=2669272485
Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
CSeq: 102 INVITE
Contact: <sip:214@192.168.130.49:5060>
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-T41S 66.84.0.15
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:214@192.168.130.49:5060>

<--- Transmitting (NAT) to 91.121.129.23:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-TKLX-87078c4d-1549530f;received=91.121.129.23;rport=5060
Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
CSeq: 198813419 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:s@185.246.18.202:5060>
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.130.49:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.130.254:5060;branch=z9hG4bK4374ac72;rport=5060
From: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
To: <sip:214@192.168.130.49:5060>;tag=2669272485
Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
CSeq: 102 INVITE
Contact: <sip:214@192.168.130.49:5060>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-T41S 66.84.0.15
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 215

v=0
o=- 20037 20037 IN IP4 192.168.130.49
s=SDP data
c=IN IP4 192.168.130.49
t=0 0
m=audio 12244 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.130.49:12244
sip_route_dump: route/path hop: <sip:214@192.168.130.49:5060>
Transmitting (NAT) to 192.168.130.49:5060:
ACK sip:214@192.168.130.49:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.130.254:5060;branch=z9hG4bK1c2f34b1;rport
Max-Forwards: 70
From: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
To: <sip:214@192.168.130.49:5060>;tag=2669272485
Contact: <sip:0383723596@192.168.130.254:5060>
Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Content-Length: 0

[ddpetit]

---
Audio is at 10246
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 91.121.129.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-TKLX-87078c4d-1549530f;received=91.121.129.23;rport=5060
Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
CSeq: 198813419 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:s@185.246.18.202:5060>
Content-Type: application/sdp
Content-Length: 292

v=0
o=root 665561663 665561663 IN IP4 185.246.18.202
s=Asterisk PBX 13.14.1~dfsg-2+deb9u4
c=IN IP4 185.246.18.202
t=0 0
m=audio 10246 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>

<--- SIP read from UDP:91.121.129.23:5060 --->
ACK sip:s@192.168.2.100:5060 SIP/2.0
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
Contact: <sip:10.7.1.65:5060>
CSeq: 198813419 ACK
From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
Max-Forwards: 29
To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-LHLE-8707918f-3db175f4
User-Agent: Cirpack/v4.76 (gw_sip)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:91.121.129.23:5060 --->
INVITE sip:s@192.168.2.100:5060 SIP/2.0
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
Contact: <sip:10.7.1.65:5060>
Content-Type: application/sdp
CSeq: 198813420 INVITE
From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
Max-Forwards: 29
Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-BPIK-87079191-6277c4a6
Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
User-Agent: Cirpack/v4.76 (gw_sip)
Content-Length: 318

v=0
o=cp10 156526349186 156526349187 IN IP4 91.121.128.136
s=SIP Call
c=IN IP4 91.121.128.136
t=0 0
m=audio 32142 RTP/AVP 0 8 18 101
b=AS:82
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (13 headers 15 lines) ---
Sending to 91.121.129.23:5060 (NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 91.121.128.136:32142

<--- Transmitting (NAT) to 91.121.129.23:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-BPIK-87079191-6277c4a6;received=91.121.129.23;rport=5060
Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
CSeq: 198813420 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:s@185.246.18.202:5060>
Content-Length: 0


<------------>
Audio is at 10246
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 91.121.129.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-BPIK-87079191-6277c4a6;received=91.121.129.23;rport=5060
Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
CSeq: 198813420 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:s@185.246.18.202:5060>
Content-Type: application/sdp
Content-Length: 292

v=0
o=root 665561663 665561664 IN IP4 185.246.18.202
s=Asterisk PBX 13.14.1~dfsg-2+deb9u4
c=IN IP4 185.246.18.202
t=0 0
m=audio 10246 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>

<--- SIP read from UDP:91.121.129.23:5060 --->
ACK sip:s@192.168.2.100:5060 SIP/2.0
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
Contact: <sip:10.7.1.65:5060>
CSeq: 198813420 ACK
From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
Max-Forwards: 29
To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-YYHU-87079197-2495cec1
User-Agent: Cirpack/v4.76 (gw_sip)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:91.121.129.23:5060 --->
INVITE sip:s@192.168.2.100:5060 SIP/2.0
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
Contact: <sip:10.7.1.65:5060>
Content-Type: application/sdp
CSeq: 198813421 INVITE
From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
Max-Forwards: 29
Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-GJGF-8707921a-3fcd69fb
Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
User-Agent: Cirpack/v4.76 (gw_sip)
Content-Length: 243

v=0
o=cp10 156526349186 156526349188 IN IP4 91.121.128.136
s=SIP Call
c=IN IP4 91.121.128.136
t=0 0
m=audio 32142 RTP/AVP 8 101
b=AS:82
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (13 headers 12 lines) ---
Sending to 91.121.129.23:5060 (NAT)
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 91.121.128.136:32142

<--- Transmitting (NAT) to 91.121.129.23:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-GJGF-8707921a-3fcd69fb;received=91.121.129.23;rport=5060
Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
CSeq: 198813421 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:s@185.246.18.202:5060>
Content-Length: 0

[ddpetit]

<------------>
Audio is at 10246
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 91.121.129.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-GJGF-8707921a-3fcd69fb;received=91.121.129.23;rport=5060
Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
CSeq: 198813421 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:s@185.246.18.202:5060>
Content-Type: application/sdp
Content-Length: 268

v=0
o=root 665561663 665561665 IN IP4 185.246.18.202
s=Asterisk PBX 13.14.1~dfsg-2+deb9u4
c=IN IP4 185.246.18.202
t=0 0
m=audio 10246 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>

<--- SIP read from UDP:91.121.129.23:5060 --->
ACK sip:s@192.168.2.100:5060 SIP/2.0
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
Contact: <sip:10.7.1.65:5060>
CSeq: 198813421 ACK
From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
Max-Forwards: 29
To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-SAQW-8707921f-562ea85e
User-Agent: Cirpack/v4.76 (gw_sip)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:192.168.130.49:5060 --->
BYE sip:0383723596@192.168.130.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.130.49:5060;branch=z9hG4bK144884565
From: <sip:214@192.168.130.49:5060>;tag=2669272485
To: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
CSeq: 2 BYE
Contact: <sip:214@192.168.130.49:5060>
Max-Forwards: 70
User-Agent: Yealink SIP-T41S 66.84.0.15
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.130.49:5060 (NAT)
Scheduling destruction of SIP dialog '56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060' in 6400 ms (Method: BYE)

<--- Transmitting (NAT) to 192.168.130.49:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.130.49:5060;branch=z9hG4bK144884565;received=192.168.130.49;rport=5060
From: <sip:214@192.168.130.49:5060>;tag=2669272485
To: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
CSeq: 2 BYE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '19122-IS-0c93dd04-597706255@siptrunk.ovh.net' in 6400 ms (Method: ACK)
Reliably Transmitting (NAT) to 91.121.129.23:5060:
BYE sip:10.7.1.65:5060 SIP/2.0
Via: SIP/2.0/UDP 185.246.18.202:5060;branch=z9hG4bK26126eb9;rport
Route: <sip:91.121.129.23:5060;lr>
Max-Forwards: 70
From: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
To: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u4
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:91.121.129.23:5060 --->
SIP/2.0 200 OK
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
CSeq: 102 BYE
From: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
Record-Route: <sip:91.121.129.23:5060;transport=udp;lr>;session=443934
To: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
Via: SIP/2.0/UDP 192.168.2.100:5060;received=192.168.2.100;rport=5060;branch=z9hG4bK26126eb9
Server: Cirpack/v4.76 (gw_sip)
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '19122-IS-0c93dd04-597706255@siptrunk.ovh.net' Method: ACK

[bwen]

Bonjour,

J'ai déjà rencontré ce problème plusieurs, c'est généralement toujours un problème "réseau" dans 99% des cas, une fois c’était effectivement un blocage niveau firewall et l'autre cas que j'ai rencontré était un peu plus "tordu", c’était une route de retour qui n'empruntait pas le même chemin que la requête initiale.

Cordialement,