sip attacks

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My asterisk server is getting bogged down every 5 minutes. My ping time is
going from 60ms to 800 ms and the call quality is bad. I have fail2ban running and I am using iptables. I have two ip connections
to the box. How can I tell if the poor performance is due to sip attacks? I don't see
any reg attempts in my asterisk cli. I use to get frequent attacks but
fail2ban seems to be taking care of that. See how ping time gets worst in a short space of time and server performance

Asterisk Users 4.1 years ago 2 Answers

asterisk + sccp-b problem

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Dear,
with asterisk 1.6.2.18 and sccp-bv3stable on two servers, we tried to
register about 1200 cisco phones, for a company.
in out of official hours, all 1200 phones registered and the cpu and ram was
below 5%. H323 is the protocol for incoming calls, and SIP for outgoing ones. in official hours, with only 10 calls, the cpu went more than 100% , and
crashed.
the bt full result of gdb was attached I have some questions now,
1-is any problem in the attached report.
2-does asterisk 1.4 more stable than 1.6…

Asterisk Users 4.2 years ago 1 Answer

Tutorial on the Asterisk Manager Interface

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I've used the manager interface to make calls successfully, now I'd like
a look at some of he other ways it can be used. I've seen references to its use to perform call cut off and rate CDRs. Is anyone aware of a reference or tutorial I could look at? Bruce Ferrell

Asterisk Users 4.2 years ago 0 Answers

Accept the dtmf input in call patch

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Hi team, Is it possible to capture dtmf input once call is patched between a-party and b-party? Also on dtmf input issue hangup request to b-party with out disconnecting A-party. How is this scenario implemented in dialplan?
Thanks
Vinod Dharashive
Sent from BlackBerry® on Airtel

Asterisk Users 4.2 years ago 0 Answers

How to use these feature of Asterisk

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Use This Information. You can customize the prompt a bit, if the default prompt is too dull for
you. First add these lines to */etc/asterisk/extensions.conf* in the
[globals] section: ${ENV(UNIX)}
${ENV(ASTERISK_PROMPT)} Then in */etc/profile* on the Asterisk server, set the ASTERISK_PROMPT
values: ASTERISK_PROMPT='%t, %l2, %h*> '
export PATH USER LOGNAME MAIL HOSTNAME HISTSIZE INPUTRC ASTERISK_PROMPT Your *export* variables will probably be different; just tack
ASTERISK_PROMPT on at the end. Reboot, run *asterisk -r* from your X
terminal, and voilá! The prompt is customized and your colors do not change:
*17:51:30, 0.54,…

Asterisk Users 4.2 years ago 0 Answers

X86_64 Compilation Issue

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Hi, compiling up a new installation of Asterisk 1.8.5 on CentOS 6 X86_64 and am seeing the following when running the make: /usr/bin/ld: skipping incompatible /usr/lib/libpam.so when searching for -lpam
/usr/bin/ld: skipping incompatible /usr/lib/libssl.so when searching for -lssl
/usr/bin/ld: skipping incompatible /usr/lib/libssl.a when searching for -lssl
/usr/bin/ld: skipping incompatible /usr/lib/libcrypto.so when searching for -lcrypto
/usr/bin/ld: skipping incompatible /usr/lib/libcrypto.a when searching for -lcrypto How can I get Asterisk to pick up the 64 bit version of the libraries instead of the 32 bit ones ? Is it just a case of updating LD_LIBRARY_PATH ?

Asterisk Users 4.2 years ago 1 Answer

instead of username

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Hello,
Asterisk seems to try to authenticate incoming INVITE based on the [section]
in sip.conf and not the username specified. I just removed the "insecure" option from my sip.conf requesting every
connection to be authenticated. I added the match_auth_username=yes in the
[general] section for extra security. To make it work, I have to use the
same [section] identifier as username. This is really bad because if
multiple provider are giving me the same username, it doesn't work. If I put the following data in sip.conf, it doesn't work. Asterisk return
the following error:…

Asterisk Users 4.2 years ago 0 Answers