similar to: Asterisk-1.2.10 not releasing SIP sessions

Displaying 20 results from an estimated 10000 matches similar to: "Asterisk-1.2.10 not releasing SIP sessions"

2010 Apr 02
1
Extremely weired Thunderbird OpenSSH interaction
Dear OpenSSH developers, first thank you for this great tool! Me and a friend have experienced some seriously crazy interaction between Thunderbird and OpenSSH, the problem is it's not reproducable but as it left definite traces on the server and it could be a serious security problem I still want to report it. so the following happened: My friend is running Ubuntu 9.10 with the new
2010 Feb 25
3
why mtd root device number is __makedev(0,254)
Hi, I'm wondering when /proc/cmdline has "root=/dev/mtdblockX", name_to_dev_t_real will return __makedev(0,254) for root device's device number which is weired. I think we need to create the device on the fly from information in sysfs. Maybe I have mistaken something. Is there any clue? -- Regards, Shizheng -------------- next part -------------- An HTML attachment was
2017 Nov 17
2
semi-OT:apcupsd
John R Pierce wrote: > On 11/17/2017 7:16 AM, m.roth at 5-cent.us wrote: >> I can't seem to find apcupsd for C 6. Just went to epel's website, and >> not >> visible. Anyone have a clue? >> > > suggestion, use NUT instead, the Network UPS Tools....? works for all > sorts of UPS's, not just APC, and supports a master/slave sort of > network control
2006 Dec 29
2
chan_sip loading delay in Asterisk 1.2.10
Hi, I'm running Asterisk 1.2.10 on gentoo linux and facing strange kind of issue. 1. chan_sip.so takes about 10 secs to load up when asterisk starts. 2. When I dialout using SIP it takes 20 secs to output " -- Called SIP 123@1.1.1.1" and get ring back from B party... Is there any config that I can check to reduce both delays? -ag
2020 Apr 30
2
dreplsrv memory
Hi ! Since few months, dreplsrv is eating lots of memory in few days. Within 2 weeks, the process is generaly OOM killed and replication become weired. I need to restart samba-ad-dc. Here is my config : - Samba 4.11.6 / 4 vcpu / 2Go Ram - 10215 objects (ldbs file = 300MB) This memory problem only happen on my headbridge DC (star topolgy with 20 DC) If I add some ram, the
2005 Jun 01
1
A problem on sink() and format,suggestions appreciated
Dear R users I get a weired problem when use sink: since the data set pretty big, I sink intermediate result for further use,following lines are consistently used when write data ########################### sink("dataname.txt") data sink() ########################## at first couples of run, all 10 variables are wrote to a file in following format: V1 V2 ....... V10 1
2008 Apr 18
2
R CMD check <PACKAGE> Error
Hi, Can anyone give me a hint on what's wrong if "R CMD check <PACKAGE>" gives me this error: Rd files with unknown encoding: <FILENAME>.Rd I have checked that file over and over and can't find any weired characters or some such. How can I fix this? Thanks for your time, Joh
2010 Jun 07
1
fit data with y = x^-1
Dear list, I am getting weired with fitting data with a 1/x-polynomial. Suggest I have the following data: x <- c(1,2,3,4,5,6,7) y <- c(100,20,4,2,1,.3,.1) I may fit this with a linear model fit1 = lm(y ~ I(x)) Getting plot out of this model I applied library(polynom) pol1 = polynomial(fit1$coefficients) f1 = as.function(pol1) plot(x,y) lines(x, f1(x), col = 2) Clearly, this model
2006 Sep 28
2
RHTML page not rendering properly
I''ve got a weired problem. I got a RHTML page which is not rendered properly in the browser all other pages in my app do renders correctly. Only this one page returns the page''s HTML with the following header Content-Type: text/html Set-Cookie: _session_id=cb3c140fd4ef907c99ef203a053fba1a; path=/ Status: 200 OK Cache-Control: no-cache <html> .....................
2010 Apr 21
3
Asterisk choking on voice messages announcements
Hi Everyone, I have a weired situation where calls in and out are proceessed all right but when I dial *97 Asterisk is literally choking when it comes to announcements like "Password" or "Call from 205-456-2222". Each one of those announcements can take like 10+ seconds to finish with most of it not even compoundable. I run "top" and there is no heavy load on CPU or
2011 Apr 09
1
Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the "h" extension?
Hi Everyone, Trying to run a php script after DeadAGI for A2Billing does it's magic. This is the dialplan: [a2billing] exten => _X.,1,System(php pre-call.php ${CALLERID(num)} ${EXTEN} ${UNIQUEID}) exten => _X.,n,AGI(a2billing.php,1) exten => _X.,n,Hangup() *exten => h,1,Wait(5)* *exten => h,n,System(php post-call.php ${CALLERID(num)} ${UNIQUEID})* As you can see above, I even
2003 Oct 18
1
why does data frame subset return vector
Hello, I've a weired problem with a data frame. Basically it should be just one column with specific names coming from a data file (the file contains 2 rows, one should be the for the rownames of the data frame the other contains numeric values). > df.rr <- read.table("RR_anova.txt", header=T, comment.char="", row.names=1) > df.rr[c(1,2,3),] [1] 1.11e-16 1.11e-16
2002 Jul 31
8
Desktop.ini and samba - Please comment
I was just wondering why does samba look for Desktop.ini when i am mapping to my home directory from w2k running samba 2.2.5 on sol2.6 , client is w2k am troubleshooting an weird issue where am not able to see all my unix home directory files on w2k log level10 is showing me that samba is looking for Desktop.ini could this be cousing something weired. thanks in advance
2006 Oct 21
1
zaptel 1.2.10 make problem
Hi iam installing zaptel 1.2.10 on my FC5 when i make iam getting following error any one suggest me whats wrong, i have installed source also in the same server. grep: /lib/modules/2.6.15-1.2054_FC5/build/include/linux/autoconf.h: No such file or directory ZAPTELVERSION="1.2.10" build_tools/make_version_h > version.h.tmp if cmp -s version.h.tmp version.h ; then echo; else \
2006 Aug 12
3
Problems with x86_64 kickstart
I have successfully set up a very nice kickstart install with the i386 version of CentOS 4.3 and it works great. I am using a custom kickstart script and the kernel and initrd from disc1/images. But when I try to duplicate this for a 64 bit setup (changing paths in my tftp server, ks.cfg, etc to point to the 64 bit stuff) it successfully tftp's everything just as it should (tethereal
2005 Oct 08
0
Re: Asterisk-Users Digest, Vol 15, Issue 28
Hello All Anybody had used ooH323 for asterisk i using ooH323-0.7.2 and asterisk CVS may 2005. OpenH323 1.17.1 and pwlib 1.9.0 and GNUGK 2.0.2 audio is very good, better than SIP and IAX, but i have problem. how to router call from openh323 to outside PSTN. my h323.conf setting ; Objective System's H323 Configuration example for tvcti ; ooh323c driver configuration ; ; [general]
2012 May 10
2
Xapian 1.2.10 released
I've uploaded Xapian 1.2.10 (including Search::Xapian 1.2.10.0). Unsurprisingly, you can download it from: http://xapian.org/download You can read an overview of the release here: http://trac.xapian.org/wiki/ReleaseOverview/1.2.10 The full lists of user-visible changes are linked to from there, and also from the "[news]" links on the download page. As always, if you encounter
2011 Jun 27
0
rtptimeout on 1.8.4
Hi Since switching from 1.6.x to 1.8.4 I have noticed the following 1. When you do a 'core show channel <channel name>' the resulting information only shows data for "Frames In" , "Frames out" is always 0. 2. The rtptimeout option in the sip.conf no longer seems to work. I have this set to 60 seconds but have had channels which have not timeout when the rtp
2007 Mar 28
3
Multi-line phones - Asterisk uses wrong callerid
I have some phones (and an ATA) that are shared between two users who each have separate voicemail but they are not behaving as desired nor expected. Incoming calls show up on the correct lines. Calls originating from the device are seen, at the terminating device, as coming from the account listed last in sip.conf, regardless of the line selected. This creates three main issues I would like
2013 Feb 03
2
RTP timeout if the asterisk box behind NAT
Dears; I am facing a problem in disconnecting the calls, it is related to the rtptimeout (disconnecting if there is no RTP packets from both sides). My Asterisk Box is behind NAT but there is a static real IP address at the ADSL router. We call from the Mobile to the PSTN analogue numbers which are connected to Asterisk Analogue card (the telephone lines are analoge), and then we dial the