write() returned error

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On Fri, 11 Mar 2011, satish patel wrote: > We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi script
> doesn't working We have allpage.agi script for paging system on all
> polycom 501 but after upgrade it broke. Any idea what is this error ? > [Mar 11 15:40:46] ERROR[3140]: utils.c:1130 ast_carefulwrite: write()
> returned error: Broken pipe Without source code, I'd guess you are violation the AGI protocol. What language are you using? which AGI library are you using? Can you reduce your source code to a simple application that reliably reproduces

Asterisk Users 4.5 years ago 4 Answers

Multiple array overflow and crash vulnerabilities in UDPTL code

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Asterisk Project Security Advisory - AST-2011-002 Product Asterisk
Summary Multiple array overflow and crash vulnerabilities in
UDPTL code
Nature of Advisory Exploitable Stack and Heap Array Overflows
Susceptibility Remote Unauthenticated Sessions
Severity Critical
Exploits Known No
Reported On January 27, 2011
Reported By Matthew Nicholson
Posted On February 21, 2011
Last Updated On February 21, 2011
Advisory Contact Matthew Nicholson
CVE Name Description When decoding UDPTL packets, multiple stack and heap based
arrays can be made to overflow by specially crafted packets.
Systems doing T.38…

Asterisk Users 4.6 years ago 0 Answers

Statistics for Asterisk CDR

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Hi everybody, I am looking for CDR stats, which can provide functionality:
- call statistics
- agent statistics Basically Call Center statistics. Can you recommend me some Open Source project ? Thanks in advance,
Albert

Asterisk Users 4.6 years ago 0 Answers

CDR with unix time.

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Good morning everyone. I wonder if it is possible, without touching the source code, to Asterisk
save the cdr with date in unix time instead of the default date. It's
possible?
Thanks in advance,

Asterisk Users 4.6 years ago 6 Answers

asterisk18 rpm issues

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Hi there, Per the instruction from http://www.asterisk.org/downloads/yum , I
setup the yum repository on my Centos 5 x86_64 machine and did a yum install asterisk18 asterisk18-configs then I startup the asterisk (with no changes to config) just to see if
it runs, but see below errors in the /var/log/asterisk/messages: [Jan 31 11:41:40] WARNING[2899] loader.c: Error loading module
'res_pktccops': /usr/lib/asterisk/modules/res_pktccops.so: cannot open
shared object file: No such file or directory
[Jan 31 11:41:40] WARNING[2899] loader.c: Error loading module
'chan_mgcp.so': /usr/lib/asterisk/modules/chan_mgcp.so: undefined
symbol: ast_pktccops_gate_alloc I checked the system and can't find the file

Asterisk Users 4.6 years ago 3 Answers

Return variables from func_odbc calls?

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This is primarily aimed at Sir Lesher, whose name graces the source
code for func_odbc that I'm currently trying to read to answer this
question. Tilghman (or anyone else who has determined the answer to this query), I have googled, searched wikis, and I'm currently perusing the source
code, but the long and short of it is that I cannot seem to find any
reference to variables set by func_odbc calls such as something that
would indicate if a query worked so that I can (in the dialplan)
handle errors on the fly. Another…

Asterisk Users 4.7 years ago 13 Answers

Stack buffer overflow in SIP channel driver

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Asterisk Project Security Advisory - AST-2011-001 Product Asterisk
Summary Stack buffer overflow in SIP channel driver
Nature of Advisory Exploitable Stack Buffer Overflow
Susceptibility Remote Authenticated Sessions
Severity Moderate
Exploits Known No
Reported On January 11, 2011
Reported By Matthew Nicholson
Posted On January 18, 2011
Last Updated On January 18, 2011
Advisory Contact Matthew Nicholson
CVE Name Description When forming an outgoing SIP request while in pedantic mode, a
stack buffer can be made to overflow if supplied with
carefully crafted caller ID information. This…

Asterisk Users 4.7 years ago 3 Answers

DTMF not being heard correctly by far end conference system

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Am 12.01.2011 11:37, schrieb Duncan Turnbull:
> Hi there
>
> I have two different asterisk systems (both 1.4) whose dtmf tones are not being picked up by a particular conference system users are dialling into. I can call myself with the phones and hear the tones, but I am guessing perhaps they are too short or somehow different. I have looked and looked but can't nail down the reason. I don't believe this is a general issue, rather some specific conference systems that they need.
>
> I am sure I saw this covered…

Asterisk Users 4.7 years ago 1 Answer