Hello
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 :
[code][general]
language=fr
allowguest=no
progressinband=yes
language=fr
;qualify=yes
directmedia=no
notifyringing = yes
limitonpeers = yes
canreinvite=no
srvlookup=yes
bindport=5060
bindaddr=0.0.0.0
defaultexpiry = 3600
register => 0033xxxxxxxxx:xxxxxxxxx@sip.ovh.fr/0033xxxxxxxxx
[ovh2]
;disallow=all
username=0033xxxxxxxxx
nat=no
type=friend
secret=xxxxxxx
qualify=yes
insecure=port,invite
host=sip.ovh.fr
fromuser=0033xxxxxxxxx
fromdomain=sip.ovh.fr
dtmfmode=inband
context=ovx
allow=ulaw
allow=alaw
[Maison]
type=friend
username=Maison
secret=xxxxxxxxx
callerid=“Maison” <39>
host=dynamic
call-limit = 100
context=interne
language=fr
;insecure=port
canreinvite=no
dtmfmode=rfc2833
video=no
restrictcid=no
amaflags=default
mailbox=39@default[/code]
here are the sip settings of my asterisk server :
[code]sip show settings
Global Settings:
UDP Bindaddress: 0.0.0.0:5060
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
Store SIP_CAUSE: No
Network QoS Settings:
IP ToS SIP: CS0
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
Externhost:
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
[/code]
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'