[asterisk-users] One way voice with Asterisk

[Zuhair Raza]

Hi
Try Nat=yes in general settings

On 06-Nov-2010 9:57 PM, "Silver Thorne" <zoraxus at gmail.com> wrote:
> Let me explain:
>
> When I dial into Asterisk ( I have a SIP trunk - which I need to make
> sure is not faulty), I only get one-way voice communication.
> The calling party, from the SIP trunk hears nothing - the extension
> rings on the Asterisk server (you can see it in the CLI and hear it at
> the computer), and the softphone rings
>
> However, when you answer the SIP softphone , you can only hear the voice
> FROM the softphone out.
>
> Where would I start to troubleshoot this? I am a little clueless!
>
> Thanks for all of your help.
>
> Asterisk 1.4.31 built by root @ some_server.foo.net on a x86_64 running
> Linux on 2010-06-10 14:32:34 UTC
>
> Sip Settings:
>
> Global Settings:
> ----------------
> SIP Port: 5060
> Bindaddress: 0.0.0.0
> Videosupport: No
> AutoCreatePeer: No
> Allow unknown access: Yes
> Allow subscriptions: Yes
> Allow overlap dialing: Yes
> Promsic. redir: No
> SIP domain support: No
> Call to non-local dom.: Yes
> URI user is phone no: No
> Our auth realm asterisk
> Realm. auth: No
> Always auth rejects: No
> Call limit peers only: No
> Direct RTP setup: No
> User Agent: Asterisk PBX
> MWI checking interval: 10 secs
> Reg. context: (not set)
> Caller ID: asterisk
> From: Domain:
> Record SIP history: Off
> Call Events: Off
> IP ToS SIP: none
> IP ToS RTP audio: none
> IP ToS RTP video: none
> T38 fax pt UDPTL: No
> RFC2833 Compensation: No
> SIP realtime: Disabled
>
> Global Signalling Settings:
> ---------------------------
> Codecs: 0x8000e (gsm|ulaw|alaw|h263)
> Codec Order: none
> T1 minimum: 100
> No premature media: No
> Relax DTMF: 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: No
> Reg. min duration 60 secs
> Reg. max duration: 3600 secs
> Reg. default duration: 120 secs
> Outbound reg. timeout: 20 secs
> Outbound reg. attempts: 0
> Notify ringing state: Yes
> Notify hold state: No
> SIP Transfer mode: open
> Max Call Bitrate: 384 kbps
> Auto-Framing: No
>
> Default Settings:
> -----------------
> Context: default
> Nat: RFC3581
> DTMF: rfc2833
> Qualify: 0
> Use ClientCode: No
> Progress inband: Never
> Language: (Defaults to English)
> MOH Interpret: default
> MOH Suggest:
> Voice Mail Extension: asterisk
>
> ----
> Parsing /etc/asterisk/extconfig.conf
>
> sip show peer
>
> * Name : 155
> Secret :<Set>
> MD5Secret :<Not set>
> Context : extern
> Language : en
> AMA flags : Unknown
> Transfer mode: open
> MaxCallBR : 384 kbps
> CallingPres : Presentation Allowed, Not Screened
> Call limit : 0
> Callgroup :
> Pickupgroup :
> Callerid : "Glen's Sysadmin Test Line"<200111222>
> ACL : No
> Codec Order : (none)
> Auto-Framing: No
>
>
>
> --
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