Juniper JUNOS OS 10.4 - RELEASE NOTES Release Note page 163

Copyright © 2010, Juniper Networks, Inc.
Issues in Junos OS Release 10.4 for SRX Series Services Gateways and J Series Services Routers
On SRX3600 devices, under certain conditions TCP window information was not
updating properly, causing TCP sequence check errors. [PR/530399: This issue has
been resolved.]
On SRX650 devices, Dot1p bits of Level 2 packet traffic across XPIMs changed.
[PR/534064: This issue has been resolved.]
On J2320 devices, a PFE core heap memory leak caused by a MAC move led to high
flowd CPU utilization. [PR/541185: This issue has been resolved.]
On SRX240 High Memory devices, when users committed a configuration change that
added a security policy containing a DNS address object, the device generated a core
file if the DNS address object was unresolved to an IP address. [PR/542175: This issue
has been resolved.]
On SRX5800 devices, a core file was generated as a result of IKE IPsec passthrough.
[PR/551631: This issue has been resolved.]
Hardware
On SRX650 devices, when the 2-Port 10-Gigabit Ethernet XPIM was operating in fiber
mode and the link speed was set to 1000 megabits (Mb), the data transmission did
not take place properly. [PR/498016: This issue has been resolved.]
Infrastructure
On SRX650 devices, the SNMP walk to the device timed out randomly. [PR/524629:
This issue has been resolved.]
On SRX Series devices, changing the firewall filter policer configuration resulted in the
SNMP MIBs not being updated correctly and therefore the counters not being accessible.
[PR/555719: This issue has been resolved.]
Integrated Convergence Services
On SRX210 devices, J-Web did not provide support for the SIP template extension
inheritance feature. [PR/455787: This issue has been resolved.]
On SRX240 devices with Integrated Convergence Services, when you call the Sip2
phone from the Sip1 phone and then press the
call the e911 number, the calls are sent successfully. When you press the
button on the Sip2 phone, the Sip2-e911 calls were disconnected, but the Sip1 phone
call remaind active. [PR/489227: This issue has been resolved.]
On SRX210 devices with Integrated Convergence Services, if you had a call established
between two SIP stations and you made a call transfer from either of the SIP stations
to an analog FXS station, the call goes through but there could be no voice.
[PR/504269: This issue has been resolved.]
On SRX240 devices with Integrated Convergence Services, when all lines were busy
in a hunt group, you might receive a busy signal and the call might not be forwarded
to a voice-mail server. [PR/516691: This issue has been resolved.]
On SRX240 devices with voice capability and Avaya ASM set up, the DTMF tone was
not heard when the last added party in a 3-way conference call hangs up. [PR/529115:
This issue has been resolved.]
button on the Sip2 phone and
Transfer
Complete
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