[asterisk-bugs] [Asterisk 0010414]: One-way audio (one-way perfect and one-way distorted)

noreply at bugs.digium.com noreply at bugs.digium.com
Thu Aug 9 16:03:37 CDT 2007


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=10414 
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Reported By:                jeanneth
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   10414
Category:                   Channels/chan_h323
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.9  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             08-09-2007 01:03 CDT
Last Modified:              08-09-2007 16:03 CDT
====================================================================== 
Summary:                    One-way audio (one-way perfect and one-way
distorted)
Description: 
I make ---[H323]-----> asterisk --[SIP]----> sip client but is not working
proprely  I have One-way audio (one-way perfect and one-way distorted)


====================================================================== 

---------------------------------------------------------------------- 
 jeanneth - 08-09-07 16:03  
---------------------------------------------------------------------- 
isis*CLI== New H.323 Connection created.
        --Received SETUP message
    -- Setting up Call
    --          Call token:  [ip$200.6.248.126:51600/4096]
    --          Calling party name:  [unknown]
    --          Calling party number:  [24845678]
    --          Called party name:  [51298907]
    --          Called party number:  [51298907]
    --          Calling party IP:  [200.6.248.126]
Setting capabilities to 0x4010f (g723|gsm|ulaw|alaw|g729|h261)
Capabilities in preference order is ()
Allowed Codecs:
         Table:
   G.723.1A <1>
   G.723.1 <2>
   GSM-06.10 <3>
   G.711-uLaw-64k <4>
   G.711-ALaw-64k <5>
   G.729A <6>
   G.729 <7>
   UserInput/hookflash <8>
   UserInput/RFC2833 <9>
   UserInput/dtmf <10>
 Set:
   0:
     0:
       G.723.1A <1>
       G.723.1 <2>
       GSM-06.10 <3>
       G.711-uLaw-64k <4>
       G.711-ALaw-64k <5>
       G.729A <6>
       G.729 <7>
     1:I>
       UserInput/hookflash <8>
     2:
       UserInput/RFC2833 <9>
       UserInput/dtmf <10>

isis*CLI=-= In OnAnswerCall for call 4096
                - Progress Indicator: 0
                - Inserting PI of 0 into ALERTING message
        -- Started logical channel: sending GSM-06.10
                -- channelsOpen = 1
                External RTP Session Starting
                RTP channel id 1 parameters:
                -- remoteIpAddress: 200.6.248.126
                -- remotePort: 49154
                -- ExternalIpAddress: 216.144.238.254
                -- ExternalPort: 5378
        -- Started logical channel: receiving GSM-06.10
                -- channelsOpen = 2
                External RTP Session Starting
                RTP channel id 1 parameters:
                -- remoteIpAddress: 200.6.248.126
                -- remotePort: 49154
                -- ExternalIpAddress: 216.144.238.254
                -- ExternalPort: 5378
        ExternalRTPChannel Destroyed
        ExternalRTPChannel Destroyed
        ExternalRTPChannel Destroyed
        ExternalRTPChannel Destroyed
        -- Transmitting RFC2833 on payload 101
Peer capability is GSM-06.10 <1>
Found peer capability GSM-06.10 <1>, Asterisk code is 2, frame size (in
ms) is 20
Peer capability is G.711-ALaw-64k <4>
Found peer capability G.711-ALaw-64k <4>, Asterisk code is 8, frame size
(in ms) is 20
Peer capability is G.711-uLaw-64k <5>
Found peer capability G.711-uLaw-64k <5>, Asterisk code is 4, frame size
(in ms) is 20
Peer capabilities = 0xe (gsm|ulaw|alaw), ordered list is (gsm|alaw|ulaw)
    -- Executing [51298907 at h323:1]
Dial("H323/ip$200.6.248.126:51600/4096", "SIP/23535970/51298907") in new
stack
    -- Called 23535970/51298907
isis*CLI-- Received Facility message...
Using 216.144.238.254 for outbound H.245 transport
Peer capability is GSM-06.10 <1>
Found peer capability GSM-06.10 <1>, Asterisk code is 2, frame size (in
ms) is 20
Peer capability is G.711-ALaw-64k <4>
Found peer capability G.711-ALaw-64k <4>, Asterisk code is 8, frame size
(in ms) is 20
Peer capability is G.711-uLaw-64k <5>
Found peer capability G.711-uLaw-64k <5>, Asterisk code is 4, frame size
(in ms) is 20
Peer capabilities = 0xe (gsm|ulaw|alaw), ordered list is (gsm|alaw|ulaw)
isis*CLI-- Received Facility message...
   -- SIP/23535970-08f898d0 answered H323/ip$200.6.248.126:51600/4096
        Answering call ip$200.6.248.126:51600/4096
        =-= In OnConnectionEstablished for call 4096
                -- Connection Established with "unknown [200.6.248.126]"
  == Spawn extension (h323, 51298907, 1) exited non-zero on
'H323/ip$200.6.248.126:51600/4096'
        -- Sending RELEASE COMPLETE
        -- ClearCall: Request to clear call with token
ip$200.6.248.126:51600/4096, cause EndedByRemoteUser
isis*CLI>       channelsOpen = 1
                channelsOpen = 0
isis*CLIExternalRTPChannel Destroyed
isis*CLIExternalRTPChannel Destroyed
isis*CLI-- ClearCall: Request to clear call with token
ip$200.6.248.126:51600/4096, cause EndedByTransportFail
-- unknown [200.6.248.126] has cleared the call
        == H.323 Connection deleted. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
08-09-07 16:03  jeanneth       Note Added: 0068684                          
======================================================================




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