[asterisk-users] One way voice with Asterisk

[Silver Thorne]

Hello All;

I have more clues that may assist in resolving this:

If I use the same softphone and dial out with the same Asterisk server. 
The SIP/voice traffic is able to be heard in both directions.

So, anyone have any ideas for me? Still a little clueless.

Glen
On 11/6/2010 13:00, Zuhair Raza wrote:
>
> Hi
> Try Nat=yes in general settings
>
> On 06-Nov-2010 9:57 PM, "Silver Thorne" <zoraxus at gmail.com 
> <mailto: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 
> <http://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
> >
> >
> >
> > --
> > _____________________________________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> > http://www.asterisk.org/hello
> >
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>
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