Displaying 13 results from an estimated 13 matches similar to: "stale nonce"
2005 Jul 21
1
account code missing in csv cdr
My cdrs are missing accountcodes for incoming calls from other asterisk
servers..
I've seen a few people mentioning this on the list and the solution
seems to be setting up a dialplan for incoming calls from a particular
sip peer.. in my opinion this does not scale well at all and I am
looking for a solution to correct this problem.
example sip peer:
[asterisk_gw]
type=friend
2005 Mar 16
2
t.38 support news?
Maybe I've missed it but I'm wondering if there has been any movement
towards getting t.38 support into asterisk.. has there been any news?
Where is t.38 support at? will it even happen?
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2005 Aug 01
4
test message - ignore me
Haven't seen email since the 29th.. just testing.
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2003 Apr 07
4
4-stable and C rand()?
Hi everyone, sorry if this has been answered before - I caught a whiff of a
discussion about c's rand() function in a mailing list archive, but couldn't
find a definitive answer.
I'm trying to do a simple CS project on my machine where I generate two sets
of values in parallel using rand() and am running into infinite loops of
values, and couldn't figure out why, so I wrote a test
2008 Oct 13
1
heimdal/AD documentation
as i promise last week, a incomplete documentation about configuring a trust
beetween a heimdal kdc and a windows AD domain
really sorry for non-french speakers
of course, i'm very interresting in any feedback...
Pascal
configuration
- le realm Kerberos est DEMO.LOCAL
- le realm du domaine AD est ad.demo.local
La configuration du KDC lui m?me ne pr?sente pas de difficult?
2014 Mar 13
0
From Harouna Ouedraogo.
Dear Friend,
STRICTLY CONFIDENTIAL,
I am the Manager in charge of Auditing section of Societe Generale de Banques au Burkina (SGBB), I Hoped that you will not expose or betray this trust and confident that I am about to repose on you for the mutual benefit of our both interests. I need your urgent assistance in transferring the sum of Five Million
2010 Nov 03
2
bugs and misfeatures in polr(MASS).... fixed!
In polr.R the (several) functions gmin and fmin contain the code
> theta <- beta[pc + 1L:q]
> gamm <- c(-100, cumsum(c(theta[1L], exp(theta[-1L]))), 100)
That's bad. There's no reason to suppose beta[pc+1L] is larger than
-100 or that the cumulative sum is smaller than 100. For practical
datasets those assumptions are frequently violated, causing the
2004 Apr 29
0
OT: softswitch or otherwise?
Has anyone setup SIP services with ss7 and lis trunks? If so .. what was
used hardware and software.. we're trying to do a SIP -> pstn setup and
not having much luck as QWEST keeps pushing dates off (aka trying to
screw us over) for our pri lines due to the recent court and fcc
activity in regards to unbundled switching and I'm looking for
solutions/ideas involving SS7..
2004 Dec 01
2
voicemail cuts off / hangs up
I'm having a problem with voicemail where the system will allow me to
login to the vm box no problem but when it starts tell tell me the
number of messages I have it hangs up.. I get "you have" and it dies
right there.. I'm running cvs tag v1-0.. what might be causing this?
I looked through my mail list archive and didn't notice anything like this..
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2005 Jan 14
0
app_conference compile?
Has anybody compiled app_conference as of late?
I've already asked on the app_conference devel list but as I'm rather in
a hurry my thinking is somebody here has both run into and found a way
to get this compiled and running.
Using stable asterisk and the most recent app_conference from it's cvs
on sourceforge..
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2005 Jul 13
0
tiny audio drops (blips)
We are receiving multiple audio drop outs on calls .. I've done quite a
bit of troubleshooting and it only involves calls that require the
Dial(SIP/xxx,,t) for transfers.. as long as the media path goes through
the server the audio blips happen.. using ulaw codec, btw. I have been
able to align the blips in audio to a specific point involving
asterisk.. it seems to happen right at about
2005 Aug 02
0
codec question
I'm looking for opinions on g726-32 vs. g711u..
They both have decent audio quality.. and looking at the wiki I get the
impression that g726 is like the little brother to g711. Yet, I've run
into quite a few sip termination vendors who don't support it. Does
anyone on the list actively use g726 for anything and what have those
experiences been?
The g726 codec for me at least
2005 Oct 18
1
sip rfc bye violated?
I have this in sip show history for a particular channel marked as dead
(should be removed) in sip show channels:
1. TxReqRel INVITE / 102 INVITE
2. Rx SIP/2.0 / 102 INVITE
3. CancelDestroy
4. Rx SIP/2.0 / 102 INVITE
5. CancelDestroy
6. Unhold SIP/2.0
7. Rx SIP/2.0 / 102 INVITE
8. CancelDestroy
9. Unhold SIP/2.0
10. Rx