[asterisk-users] One way voice with Asterisk

[Silver Thorne]

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