# Only one way sound on the phone? **URL:** \ **Category:** Asterisk Support\ **Created:** [July 23, 2014, 11:13am UTC](https://community.asterisk.org/t/only-one-way-sound-on-the-phone/46729 "2014-07-23T11:13:51Z")\ **Posts on this page:** 3\ **Page:** 1
**Author:** ![casp](https://avatars.discourse-cdn.com/v4/letter/c/7c8e57/32.png) [@casp](https://community.asterisk.org/u/casp)\ **Post date:** [July 23, 2014, 11:13am UTC](https://community.asterisk.org/t/only-one-way-sound-on-the-phone/46729/1 "2014-07-23T11:13:51Z")
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](mailto: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](http://host=sip.ovh.fr) fromuser=0033xxxxxxxxx [fromdomain=sip.ovh.fr](http://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'` ---
**Author:** ![casp](https://avatars.discourse-cdn.com/v4/letter/c/7c8e57/32.png) [@casp](https://community.asterisk.org/u/casp)\ **Post date:** [July 23, 2014, 12:43pm UTC](https://community.asterisk.org/t/only-one-way-sound-on-the-phone/46729/2 "2014-07-23T12:43:43Z")
If it can help, here is the sip show channel log : 91.121.129.20 is the provider ip ( [sip.ovh.fr](http://sip.ovh.fr) ) [xx.xxx.xxx.xxx](http://xx.xxx.xxx.xxx) is the sip phone public ip yy.yy.yy.yy is asterisk’s server public ip 06xxxxxxxx is the number i am calling 192.168.0.28 is the sip phone ip on my network i don’t know what is 10.7.1.65 [code]ns878958_CLI\> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry Peer 91.121.129.20 06xxxxxxxx 035e79a37c60dfc (ulaw) No Tx: ACK ovh2 [xx.xxx.xxx.xxx](http://xx.xxx.xxx.xxx) Maison 3842088367@192\_ (ulaw) No Rx: ACK Maison 91.121.129.20 (None) 00-07974-025894 (nothing) No Rx: OPTIONS 3 active SIP dialogs ns878958_CLI\> sip show channel [035e79a37c60dfce0955be883b7d7013@sip.ovh.fr](mailto:035e79a37c60dfce0955be883b7d7013@sip.ovh.fr) 3842088367@192\_168\_0\_28 ns878958\*CLI\> sip show channel [035e79a37c60dfce0955be883b7d7013@sip.ovh.fr](mailto:035e79a37c60dfce0955be883b7d7013@sip.ovh.fr) - SIP Call Curr. trans. direction: Outgoing Call-ID: [035e79a37c60dfce0955be883b7d7013@sip.ovh.fr](mailto:035e79a37c60dfce0955be883b7d7013@sip.ovh.fr) 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: 91.121.129.20:5060 Received Address: 91.121.129.20:5060 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:10.7.1.65:5060 Need Destroy: No Last Message: Tx: ACK Promiscuous Redir: No Route: sip:91.121.129.20:5060;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: [xx.xxx.xxx.xxx:5060](http://xx.xxx.xxx.xxx:5060) Received Address: [xx.xxx.xxx.xxx:5060](http://xx.xxx.xxx.xxx:5060) 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/42.075.00.000.000 Username: Maison Peername: Maison Original uri: sip:Maison@192.168.0.28:5060 Caller-ID: 30 Need Destroy: No Last Message: Rx: ACK Promiscuous Redir: No Route: sip:Maison@192.168.0.28:5060 DTMF Mode: rfc2833 SIP Options: replaces replace Session-Timer: Inactive Transport: UDP Media: RTP [/code] 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 : AMA flags : DOCUMENTATION 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 : 192.168.0.28:5060 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/42.075.00.000.000 Reg. Contact : sip:Maison@192.168.0.28:5060 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 [/code] ---
**Author:** ![dejanst](https://avatars.discourse-cdn.com/v4/letter/d/9fc348/32.png) [@dejanst](https://community.asterisk.org/u/dejanst)\ **Post date:** [July 24, 2014, 9:29am UTC](https://community.asterisk.org/t/only-one-way-sound-on-the-phone/46729/3 "2014-07-24T09:29:00Z")
[quote=“casp”]Hello As my sip phone is behind a nat … [/quote] I would bet your problem is here.