Weird no-audio problem

[akabaka]

hey everyone,
i’m currently deploying a second asterisk server at my office and try to connect them together.
the network look something like that:

I’ve a weird no-audio problem only in some cases.
situations with no problems (whichever initiate the call in every case):

• phone A call phone B
• phone A or B call C or D
• phone A or B call outbound via trunk provider

but whenever sip peers on server B try to call each others or to place an outboud call to sip trunk provider or to get a incoming call from it, even if the dial process look alright (phones ringing and so on), there’s no audio.
i’m not sure what to do here as the logs show no errors whatsoever.
any help would be greatly appreciated.
tell me if you need some config or logs
Cheers
jc

[akabaka]

After some ore log reading, I may have found something, her are the logs:

= Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
       > 0x7f19b807bfc0 -- Strict RTP learning after remote address set to: 10.173.39.2:42766
    -- Executing [06XXXXXXXX@phones:1] Dial("SIP/jc-00000177", "SIP/06XXXXXXXX@forfait-ovh") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/06XXXXXXXXX@forfait-ovh
    -- SIP/forfait-ovh-00000178 is ringing
       > 0x7f19ac0227b0 -- Strict RTP learning after remote address set to: 91.121.129.138:30150
    -- SIP/forfait-ovh-00000178 is making progress passing it to SIP/jc-00000177
       > 0x7f19ac0227b0 -- Strict RTP learning after remote address set to: 91.121.129.138:30150
    -- SIP/forfait-ovh-00000178 answered SIP/jc-00000177
    -- Channel SIP/forfait-ovh-00000178 joined 'simple_bridge' basic-bridge <5199b488-ddd7-4e33-a7bc-dbcb4f7d1b9a>
    -- Channel SIP/jc-00000177 joined 'simple_bridge' basic-bridge <5199b488-ddd7-4e33-a7bc-dbcb4f7d1b9a>
    -- Channel SIP/jc-00000177 left 'simple_bridge' basic-bridge <5199b488-ddd7-4e33-a7bc-dbcb4f7d1b9a>
    -- Channel SIP/forfait-ovh-00000178 left 'simple_bridge' basic-bridge <5199b488-ddd7-4e33-a7bc-dbcb4f7d1b9a>
  == Spawn extension (phones, 06XXXXXXXX, 1) exited non-zero on 'SIP/jc-00000177'

I can see only:
Strict RTP learning after remote address set to: 10.173.39.2:42766
but never switch to and locking to that I see on working calls like here:

 Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
       > 0x7f19b8071aa0 -- Strict RTP learning after remote address set to: 91.121.129.139:33910
    -- Executing [s@incoming:1] Dial("SIP/forfait-ovh-00000127", "IAX2/rincewind-to-office/99999s") in new stack
    -- Called IAX2/rincewind-to-office/99999s
    -- Call accepted by 217.181.187.73:46569 (format gsm)
    -- Format for call is (gsm)
    -- IAX2/rincewind-to-office-3357 is ringing
    -- IAX2/rincewind-to-office-3357 answered SIP/forfait-ovh-00000127
    -- Channel IAX2/rincewind-to-office-3357 joined 'simple_bridge' basic-bridge <4bab775e-cd70-446d-8427-8d7830f74a1c>
    -- Channel SIP/forfait-ovh-00000127 joined 'simple_bridge' basic-bridge <4bab775e-cd70-446d-8427-8d7830f74a1c>
       > 0x7f19b8071aa0 -- Strict RTP learning after remote address set to: 91.121.129.139:33910
       > 0x7f19b8071aa0 -- Strict RTP switching to RTP target address 91.121.129.139:33910 as source
[Sep  7 09:55:49] WARNING[17718][C-0000014f]: chan_iax2.c:1239 jb_warning_output: Resyncing the jb. last_delay 0, this delay -77372538, threshold 400, new offset 77372538
       > 0x7f19b8071aa0 -- Strict RTP learning complete - Locking on source address 91.121.129.139:3391

I’ve googled around but not found anything so far.
here are my sip.conf:

[general]
register => 00XXXXXXXXXX:XXXXXXXX@sip.ovh.fr
context=default
transport=udp
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
localnet=10.173.39.0/255.255.255.0 ;connected via openvpn
;externip=46.XX.XX.XX
externhost=sip-rincewind.XXXXX.XX
externrefresh=120
svrlookup=yes
limitonpeer=yes
alwaysauthreject=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
qualify=yes
disallow=g729
allow=g722
allow=ulaw
canreinvite=no
jbenable=yes
jbmaxsize=300
jbresyncthreshold=400
jbimpl=adaptive
jbtargetextra=60

[authentication]

[forfait-ovh]
type=peer
host=sip.ovh.fr
context=incoming
language=fr
defaultuser=00XXXXXXXXXXX
secret=pass
insecure=port,invite
dtmfmode=auto
video=no
restrictcid=no
amaflags=default
qualify=yes

[jc]
type=friend
host=dynamic
context=phones
qualify=yes
secret=secret
notifyringing=yes
canreinvite=no

if anyone got any advice on how to debug that
thanks

[akabaka]

ok for the record, I’ve forgotten to put

exten => _X,1,Answer

in my dialplan, just that.