Only one way sound on the phone?


I have installed my asterisk 12 server on a dedicated server. Which mean that there is no nat on the server and it uses a public ip address.

i have connected the server to my sip provider (ovh) and i have connected a sip phone to asterisk.

When i call outside with my sip phone, i can the other partie can hear what i say on the sip phone, but i can not hear what the other partie says.

i have made test after disabling seslinux and the firewall on the asterisk server and i still have the same problem.

As my sip phone is behind a nat, i have tryed with putting my sip phone on dmz and i still have the problem.

Do you know what can be the problem ?

Here is part of my sip.conf :

notifyringing = yes
limitonpeers = yes
defaultexpiry = 3600
register =>


callerid=“Maison” <39>
call-limit = 100

here are the sip settings of my asterisk server :

[code]sip show settings

Global Settings:

UDP Bindaddress:
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: No
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Path support : No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: Asterisk PBX 12.4.0
SDP Session Name: Asterisk PBX 12.4.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: 4294967295
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No

Network QoS Settings:

IP ToS RTP audio: CS0
IP ToS RTP video: CS0
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No

Network Settings:

SIP address remapping: Disabled, no localnet list
Externaddr: (null)
Externrefresh: 10

Global Signalling Settings:

Codecs: (gsm|ulaw|alaw|h263|testlaw)
Codec Order: none
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 3600 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy:
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70

Default Settings:

Allowed transports: UDP
Outbound transport: UDP
Context: default
Record on feature: automon
Record off feature: automon
Force rport: Auto (No)
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: Yes
Language: fr
Tone zone:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk


And here is a call log :

== Using SIP RTP CoS mark 5 -- Executing [06xxxxxxxx@interne:1] Set("SIP/Maison-00000020", "DIRNAME=Sortant/30/201407/23") in new stack -- Executing [06xxxxxxxx@interne:2] Dial("SIP/Maison-00000020", "SIP/06xxxxxxxx@ovh2") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/06xxxxxxxx@ovh2 -- SIP/ovh2-00000021 is ringing -- SIP/ovh2-00000021 is making progress passing it to SIP/Maison-00000020 -- SIP/ovh2-00000021 answered SIP/Maison-00000020 -- Channel SIP/Maison-00000020 joined 'simple_bridge' basic-bridge <48c8a0bf-5321-40c5-bd78-e2abb106905a> -- Channel SIP/ovh2-00000021 joined 'simple_bridge' basic-bridge <48c8a0bf-5321-40c5-bd78-e2abb106905a> -- Channel SIP/Maison-00000020 left 'simple_bridge' basic-bridge <48c8a0bf-5321-40c5-bd78-e2abb106905a> -- Channel SIP/ovh2-00000021 left 'simple_bridge' basic-bridge <48c8a0bf-5321-40c5-bd78-e2abb106905a> == Spawn extension (interne, 06xxxxxxxx, 2) exited non-zero on 'SIP/Maison-00000020'

If it can help, here is the sip show channel log : is the provider ip ( ) is the sip phone public ip
yy.yy.yy.yy is asterisk’s server public ip
06xxxxxxxx is the number i am calling is the sip phone ip on my network

i don’t know what is

[code]ns878958CLI> sip show channels
Peer User/ANR Call ID Format Hold Last Message Expiry Peer 06xxxxxxxx 035e79a37c60dfc (ulaw) No Tx: ACK ovh2 Maison 3842088367@192_ (ulaw) No Rx: ACK Maison (None) 00-07974-025894 (nothing) No Rx: OPTIONS
3 active SIP dialogs
CLI> sip show channel 3842088367@192_168_0_28
ns878958*CLI> sip show channel

  • SIP Call
    Curr. trans. direction: Outgoing
    Owner channel ID: SIP/ovh2-0000002f
    Our Codec Capability: (gsm|ulaw|alaw|h263|testlaw)
    Non-Codec Capability (DTMF): 0
    Their Codec Capability: (ulaw|alaw)
    Joint Codec Capability: (ulaw|alaw)
    Format: (ulaw)
    T.38 support No
    Video support No
    MaxCallBR: 384 kbps
    Theoretical Address:
    Received Address:
    SIP Transfer mode: open
    Force rport: No
    Audio IP: yy.yy.yy.yy (local)
    Our Tag: as1e158be1
    Their Tag: 00-07765-08c789af-3c381af94
    SIP User agent:
    Username: 06xxxxxxxx
    Peername: ovh2
    Original uri: sip:
    Need Destroy: No
    Last Message: Tx: ACK
    Promiscuous Redir: No
    Route: sip:;transport=udp;lr
    DTMF Mode: inband
    SIP Options: (none)
    Session-Timer: Inactive
    Transport: UDP
    Media: RTP

ns878958*CLI> sip show channel 3842088367@192_168_0_28

  • SIP Call
    Curr. trans. direction: Incoming
    Call-ID: 3842088367@192_168_0_28
    Owner channel ID: SIP/Maison-0000002e
    Our Codec Capability: (gsm|ulaw|alaw|h263|testlaw)
    Non-Codec Capability (DTMF): 1
    Their Codec Capability: (ulaw|alaw|g726|g729|g726aal2|g722)
    Joint Codec Capability: (ulaw|alaw)
    Format: (ulaw)
    T.38 support No
    Video support No
    MaxCallBR: 384 kbps
    Theoretical Address:
    Received Address:
    SIP Transfer mode: open
    Force rport: No
    Audio IP: yy.yy.yy.yy (local)
    Our Tag: as308b0699
    Their Tag: 3258810094
    SIP User agent: A510 IP/
    Username: Maison
    Peername: Maison
    Original uri: sip:Maison@
    Caller-ID: 30
    Need Destroy: No
    Last Message: Rx: ACK
    Promiscuous Redir: No
    Route: sip:Maison@
    DTMF Mode: rfc2833
    SIP Options: replaces replace
    Session-Timer: Inactive
    Transport: UDP
    Media: RTP

and show peer (it is very odd as i have always internal ip on this and never the external ip)

[code]sip show peer Maison

  • Name : Maison
    Description :
    Secret :
    MD5Secret :
    Remote Secret:
    Context : interne
    Record On feature : automon
    Record Off feature : automon
    Subscr.Cont. :
    Language : fr
    Tonezone :
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup :
    Pickupgroup :
    Named Callgr :
    Nam. Pickupgr:
    MOH Suggest :
    Mailbox : 39@default
    VM Extension : asterisk
    LastMsgsSent : 0/0
    Call limit : 0
    Max forwards : 0
    Dynamic : Yes
    Callerid : “Maison” <39>
    MaxCallBR : 384 kbps
    Expire : 98
    Insecure : port,invite
    Force rport : No
    Symmetric RTP: No
    ACL : No
    DirectMedACL : No
    T.38 support : No
    T.38 EC mode : Unknown
    T.38 MaxDtgrm: 4294967295
    DirectMedia : No
    PromiscRedir : No
    User=Phone : No
    Video Support: No
    Text Support : No
    Ign SDP ver : No
    Trust RPID : No
    Send RPID : No
    Path support : No
    Path : N/A
    TrustIDOutbnd: Legacy
    Subscriptions: Yes
    Overlap dial : Yes
    DTMFmode : auto
    Timer T1 : 500
    Timer B : 32000
    ToHost :
    Addr->IP :
    Defaddr->IP : (null)
    Prim.Transp. : UDP
    Allowed.Trsp : UDP
    Def. Username: Maison
    SIP Options : (none)
    Codecs : (gsm|ulaw|alaw|h263|testlaw)
    Codec Order : (none)
    Auto-Framing : No
    Status : Unmonitored
    Useragent : A510 IP/
    Reg. Contact : sip:Maison@
    Qualify Freq : 60000 ms
    Keepalive : 0 ms
    Sess-Timers : Accept
    Sess-Refresh : uas
    Sess-Expires : 1800 secs
    Min-Sess : 90 secs
    RTP Engine : asterisk
    Parkinglot :
    Use Reason : No
    Encryption : No


As my sip phone is behind a nat …


I would bet your problem is here.