|Summary:||ASTERISK-15661: One way audio after placing call on hold and resuming|
|Reporter:||Jordan Kirby (jordankirby)||Labels:|
|Date Opened:||2010-02-19 09:21:25.000-0600||Date Closed:||2011-06-07 14:07:24|
|Environment:||Attachments:||( 0) full.txt|
( 1) OnHoldTest.pcap
( 2) sip_debug.txt
( 3) sip_show_channels.txt
( 4) sip_show_history.txt
|Description:||This fault occurs on 220.127.116.11, 18.104.22.168, 22.214.171.124, 126.96.36.199-RC2, SVN-247894.|
sip.conf: directrtpsetup=yes, nat=yes
Both extensions: canreinvite=yes, nat=yes
Phones are on the same LAN and behind NAT.
Server is in a separate location also behind NAT. All standard internal and external calls work fine.
Extension A calls extension B. Extension A puts the call on hold, extension B gets played music as expected. When extension A resumes the call extension B can't hear extension A.
This seems to be because Asterisk sends the external IP address of extension B to extension A in the SDP when the call is resumed.
****** ADDITIONAL INFORMATION ******
I will attach a wireshark trace and copies of the "sip show channel xxx" results for each leg of the call.
|Comments:||By: Jordan Kirby (jordankirby) 2010-02-19 09:22:57.000-0600|
The same problem occurs when the server is not behind NAT (phones still NATd).
Attached traces are from a server not behind NAT.
By: Jordan Kirby (jordankirby) 2010-02-19 09:27:25.000-0600
The incorrect Audio IP seems to sometimes be the IP of the server and sometimes the external IP of phone A.
By: Leif Madsen (lmadsen) 2010-02-19 12:36:43.000-0600
Can you also provide a SIP trace from the Asterisk console, along with SIP history and debug level logging (as set from logger.conf)?
By: David Woolley (davidw) 2010-02-22 05:59:39.000-0600
I wonder if this is related to ASTERISK-1345545. The fix for that was to prevent partial implementation of directrtpsetup when directrtpsetup wasn't enabled. However, I believe it would still be a problem if directrtpsetup were enabled.
By: Jordan Kirby (jordankirby) 2010-02-22 07:58:46.000-0600
I have uploaded a copy of the full log (with debug on), the asterisk sip debug and the history from the channels I could see after the hold and resume.
By: Russell Bryant (russell) 2010-09-16 11:41:57
We were able to reproduce this in an older version but not in current versions, so I think this is fixed. Can you please confirm?
By: Jordan Kirby (jordankirby) 2010-10-28 03:32:53
I have retested this issue on 188.8.131.52 and 1.8.0 and I can confirm the problem still exists.
By: Leif Madsen (lmadsen) 2010-11-04 14:03:04
Do you have externip set? It sounds like you might.
Do you have localnet set? If not, that is likely the problem.
Sounds like it might be a configuration issue in sip.conf.