Displaying 20 results from an estimated 22 matches for "timert1".
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timer1
2016 Aug 10
2
chan_pjsip ignoring endpoint device state (qualify) on dial
...ll Setup time of SIP Requests.
> In case of GSM Network with high delay you need to set the T1 timer a
> higher value like 1000ms (500 ms default). Similarly you can reduce the
> Call setup time by configuring 'T2' upto you choice as per you telephony
> network. Configure t1min, timert1 and timerb according to your network.
No, that won't work.
First ? 't1min', 'timert1' and 'timerb' are chan_sip, not pjsip options.
Second ? the 'timer_t1' and 'timer_b' settings available in 'pjsip.conf'
are global and not per-endpoint. I can...
2016 Feb 04
0
AST-2016-002: File descriptor exhaustion in chan_sip
...Last Updated On February 3, 2016
Advisory Contact Richard Mudgett <rmudgett AT digium DOT com>
CVE Name Pending
Description Setting the sip.conf timert1 value to a value higher than
1245 can cause an integer overflow and result in large
retransmit timeout times. These large timeout values hold
system file descriptors hostage and can cause the system to
run out of f...
2016 Aug 09
3
chan_pjsip ignoring endpoint device state (qualify) on dial
Hi,
We have been migrating our PBX system from Asterisk 1.8 and chan_sip to
Asterisk 13 and chan_pjsip. Things are mostly, ok, but now I have
stumbled on a behaviour difference I don't like.
With chan_pjsip when a phone went unexpectedly offline (Ethernet cable
disconnected) Asterisk would detect this quickly (through the 'qualify'
pings), mark the phone as 'Unavailable' and
2013 Jul 17
0
SIP timers
Hello List,
I tried to change the following parameters in sip.conf file, but looks like it cannot be changed,
Defaut values:
;t1min=100
;timert1=500
;timerb=32000
I have changed to:
;t1min=100
timert1=100
timerb=6400
Sometime I can see too many retransmission of BYE to some of the UAs if UA is unreachable. Is there a way that I can reduce the number of retransmission of BYE message?
Regards
Rajib
-------------- next part ----------...
2008 Dec 18
2
Dial timeout with SIP - how to set timeout for INVITE ACK
I have a concern with Dial command, I want to enable a secondary route with a remote partner, if the first route fails then we use the second one :
Solution1: it will try both (there will be 2 simultanious actives calls ringing) this is not clean when calling an endusers
exten => _X.,1,Dial(SIP/${EXTEN}@remote-sip1,5 <mailto:SIP/${EXTEN}@remote-sip1,5> )
exten =>
2008 Mar 02
0
Cisco 7970 - register with NAT phone
...timerRegisterExpires>3600</timerRegisterExpires>
<timerRegisterDelta>5</timerRegisterDelta>
<timerKeepAliveExpires>120</timerKeepAliveExpires>
<timerSubscribeExpires>120</timerSubscribeExpires>
<timerSubscribeDelta>5</timerSubscribeDelta>
<timerT1>500</timerT1>
<timerT2>4000</timerT2>
<maxRedirects>70</maxRedirects>
<remotePartyID>true</remotePartyID>
<userInfo>None</userInfo>
</sipStack>
<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</auto...
2011 Mar 02
1
Registering Cisco 7942G IP phone with Asterisk!.
Hi,
?
We are new to IP phone firmware upgradation (Sorry if it is a re-post of previous question(s)).
?
Recently we have bought a cisco 7942G IP phone.
It currently has SIP 42.9-0-2SR1S firmware loaded on it.
We do not see any option to configure a SIP Proxy where we can provide SIP Server (Asterisk PC/Device)? IP address (with current firmware on it) to register it with Asterisk.
?
Do we need to
2012 Jan 15
0
configuring a Cisco 7961 so that different line appearances register to different SIP proxy addresses
...rExpires>
<timerRegisterDelta>5</timerRegisterDelta>
<timerKeepAliveExpires>120</timerKeepAliveExpires>
<timerSubscribeExpires>120</timerSubscribeExpires>
<timerSubscribeDelta>5</timerSubscribeDelta>
<timerT1>500</timerT1>
<timerT2>4000</timerT2>
<maxRedirects>70</maxRedirects>
<remotePartyID>true</remotePartyID>
<userInfo>None</userInfo>
</sipStack>
<autoAnswerTimer>1</autoAnswerTimer...
2008 Jan 25
2
Unprovisioned 7961
Hi Everyone,
I am having some issues getting my 7961 working with Trixbox. I have loaded
the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an
unprovisioned state. A status message shows up and says ?Error Verifying
Config Info?.
I have read quite a bit on this topic (getting 7961?s to work with Asterisk
and TB) and only came across a few postings where other people
2014 Jul 26
1
Rejecting secure audio stream without encryption details - when using ws clients and Kamailio integration
...session-refresher: NULL
t38pt_usertpsource: NULL
regexten: NULL
fromdomain: testers.com
fromuser: 660
qualify: NULL
defaultip: NULL
rtptimeout: NULL
rtpholdtimeout: NULL
sendrpid: NULL
outboundproxy: PU.BL.IC.IP
timert1: NULL
timerb: NULL
qualifyfreq: NULL
constantssrc: NULL
contactpermit: NULL
contactdeny: NULL
usereqphone: NULL
textsupport: NULL
faxdetect: NULL
buggymwi: NULL
auth: NULL
fullname: NULL
trunkname...
2016 Mar 29
3
Asterisk 11.22.0 Now Available
...fs in
udptl_rx_packet cause ast_frdup crash (Reported by Walter
Doekes)
* ASTERISK-25742 - Secondary IFP Packets can result in accessing
uninitialized pointers and a crash (Reported by Torrey Searle)
* ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
non-default timert1 (Reported by Alexander Traud)
* ASTERISK-25730 - build: make uninstall after make distclean
tries to remove root (Reported by George Joseph)
* ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in
sip_sipredirect (Reported by Badalian Vyacheslav)
* ASTERISK-25714 - ASAN:hea...
2014 Dec 15
1
T.38 not working - help needed with log interpretation
On 10.12.2014 11:42, Frederic Van Espen wrote:
> Hi,
>
> - Could you share the details of the SDP in each INVITE and OK packet?
> - How are your SIP endpoints configured in asterisk sip.conf? (the SIP
> trunk provider and the local endpoint)
> - What type is the local endpoint?
>
> Cheers,
>
> Frederic
>
Frederic, I now have tried to describe the situation
2015 Mar 31
0
How does chan_sip match an ACK?
...ng system
tried sending the 200 OK to Answer:
exten => _X.,1,Ringing
exten => _X.,n,Wait(1)
exten => _X.,n,Answer
exten => _X.,n,Goto(wherever)
On further reading, I would think I could also solve it by setting the
T1 values in sip.conf, instead of doing the above:
Should I set "timert1=500", or "t1min=500" or both, or what?
Cheers
Tony
--
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org
2011 Aug 08
0
Timer B in sip.conf cannot be changed
I am using 1.8. I need to change timerb to 6500, that is, if there is
no response of some sort in 6.5 seconds, consider the call failed and
try another route. It does not matter what do I set for the other
timers:
T1min=100
timert1=100
Timerb=6500
The command "sip show settings" always shows Timer B=32000. Any ideas
how can I reduce Timer B?
2016 Feb 04
0
Asterisk 11.6-cert12, 11.21.1, 13.1-cert3, 13.7.1 Now Available (Security Release)
...TP server currently has a default configuration which allows
the BEAST vulnerability to be exploited if the TLS functionality is enabled.
This can allow a man-in-the-middle attack to decrypt data passing through it.
* AST-2016-002: File descriptor exhaustion in chan_sip
Setting the sip.conf timert1 value to a value higher than 1245 can cause an
integer overflow and result in large retransmit timeout times. These large
timeout values hold system file descriptors hostage and can cause the system
to run out of file descriptors.
* AST-2016-003: Remote crash vulnerability receiving UDPTL FA...
2016 Mar 29
0
Asterisk 11.22.0 Now Available
...fs in
udptl_rx_packet cause ast_frdup crash (Reported by Walter
Doekes)
* ASTERISK-25742 - Secondary IFP Packets can result in accessing
uninitialized pointers and a crash (Reported by Torrey Searle)
* ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
non-default timert1 (Reported by Alexander Traud)
* ASTERISK-25730 - build: make uninstall after make distclean
tries to remove root (Reported by George Joseph)
* ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in
sip_sipredirect (Reported by Badalian Vyacheslav)
* ASTERISK-25714 - ASAN:hea...
2016 Mar 29
5
Asterisk 13.8.0 Now Available
...alized pointers and a crash (Reported by Torrey Searle)
* ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
Vulnerability - Investigate vulnerability of HTTP server
(Reported by Alex A. Welzl)
* ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
non-default timert1 (Reported by Alexander Traud)
* ASTERISK-25702 - PjSip realtime DB and Cache Errors since
upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by
Nic Colledge)
* ASTERISK-25730 - build: make uninstall after make distclean
tries to remove root (Reported by George Joseph...
2016 Mar 29
0
Asterisk 13.8.0 Now Available
...alized pointers and a crash (Reported by Torrey Searle)
* ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
Vulnerability - Investigate vulnerability of HTTP server
(Reported by Alex A. Welzl)
* ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
non-default timert1 (Reported by Alexander Traud)
* ASTERISK-25702 - PjSip realtime DB and Cache Errors since
upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by
Nic Colledge)
* ASTERISK-25730 - build: make uninstall after make distclean
tries to remove root (Reported by George Joseph...
2016 Jul 13
0
Certified Asterisk 13.8-cert1 Now Available
...alized pointers and a crash (Reported by Torrey Searle)
* ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
Vulnerability - Investigate vulnerability of HTTP server
(Reported by Alex A. Welzl)
* ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
non-default timert1 (Reported by Alexander Traud)
* ASTERISK-25702 - PjSip realtime DB and Cache Errors since
upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by
Nic Colledge)
* ASTERISK-25730 - build: make uninstall after make distclean
tries to remove root (Reported by George Joseph...