similar to: Sip registration Asterisk 1.8

Hello, For the moment I have the following settings in my sip.conf. I want to optimize them to archive the following things: - for the moment all my users will re-register too often. I want that only lagged users to re-register quickly. - check from time to time all users but no too often to see if is logged and can be called. Overall i want only lagged users to reregister and users with good

Hello, I want to optimize my registrations and calls of peers to my asterisk with the following options in sip.conf: ---///--- qualify = yes qualify = 500 qualifyfreq=5 registerattempts = 0 registertimeout = 10 maxexpiry = 60 minexpiry = 20 defaultexpiry = 600 ---///--- Can someone more experienced with these settings to help me to optimize connections from peers with mobile phone that using

Hello! Could you help me with Asterisk 11.21.2 and AsteriskNow platform. The problem is: My Asterisk PBX has SIP (chan_sip) trunk to provider. Asterisk periodically loses trunk registratrion: *sip show registry:* /Host??????????????????????????????????? dnsmgr Username?????? Refresh State??????????????? Reg.Time???????????????? // //X.X.X.X:5060??????????????????? N????? <LOGIN>

Hi all, I am using an asterisk as a client to connect to another asterisk server by registering with the register string. Registration is done without any hassel, but after sometime my asterisk loses the registration with the server and the server starts displaying the following msgs repeatedly: [Aug 9 06:37:59] NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale nonce

Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' => 1. Wait(1) [pbx_config] 2.

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

I lost my internet connection today for a short time. During that time 1.2.12.1 stopped talking to my phones. Asterisk was still working as I got 2 voicemails. I have TDM analog cards for incoming calls. Anyway my cisco phones had X's (lost registration) and my uniden phones said "Registration error". Why would phones loose registration to asterisk when the internet connection

Hello again! I just got a SIP account and it seems - from a config on the net -, that I've configured it correctly. But I get no call to the outside. Registration was OK. I tried: channel originate sip/1/echo at iptel.org Application ... I see the channel active for a while, but no call gets established. In my config I have defined the section [iptel] for the outgoing call and I

Hi List, One particular user is kicked quite often (espescially during the usage of Outlook). When she tries to login she isn't very succesfull. The serverlog show this: Jul 13 10:41:14 server smbd[12277]: [2006/07/13 10:41:14, 0] lib/util_sock.c:write_data(554) Jul 13 10:41:14 server smbd[12277]: write_data: write failure in writing to client 192.168.1.101. Error Connection reset by

https://bugzilla.samba.org/show_bug.cgi?id=12030 Bug ID: 12030 Summary: rsync leaves temporary files when transfer is interrupted. Product: rsync Version: 3.1.2 Hardware: All OS: FreeBSD Status: NEW Severity: enhancement Priority: P5 Component: core

Hi there I've just upgraded to openssh-3.2.2p1 from openssh-1.2.3 and am having some difficulties. On one of the platforms I'm using (linux kernel 2.2.19 with glibc 2.1.1) it works fine, but on another (linux kernel 2.2.20 with glibc 2.0.7) I get this in the syslog every time I log in: sshd[12277]: Accepted publickey for root from 212.38.67.158 port 2397 ssh2 PAM_pwdb[12277]: (sshd)

I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146 asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659 openssl-1.0.1e-51.el7_2.2.x86_64 [root at elx4 ~]#

Hello list, I have in sip.conf : /maxexpiry=60 ; Maximum allowed time of incoming registrations ; and subscriptions (seconds) minexpiry=60 ; Minimum length of registrations/subscriptions (default 60) defaultexpiry=120 ; Default length of incoming/outgoing registration ;-----------------------------------------

Hi! I want my PBX to be reachable at my ekiga.net account. It seems I am registered: vajna2*CLI> sip show registry Host Username Refresh State Reg.Time ekiga.net:5060 magwas 585 Registered Sat, 13 Nov 2010 13:48:22 However when others try to call magwas at ekiga.net, they find me unavailable. My asterisk

Hi, I have the panel data of income statement of several banks. The date 9803 means 1998-Q1, 9806 means 1998-Q2, etc. I transform the date code to 1 (for 1998Q1), 2 (for 1998Q2), ...., 16 (for 2011Q4) where 1, 2, .... are placed in Col1. Now the income statement of a specific quarter is actually the accumulation from the beginning of the year. For example, the cost data of 1999Q3 is the

Hi I installed R 2.2.0 from source and want to use install.packages but it doesn't work. http_proxy is set to http://proxy.sun.ac.za:3128 but it still can't connect to the repository. The mirror is available, I can connect to it via the internet. Any help welcome, Rainer -- NEW TELEPHONE NUMBER Tel: +27 - (0)72 808 2975 (w) Rainer M. Krug, Dipl. Phys. (Germany), MSc

Hi, I have data on stomach contents. Possible prey species are in the hundreds, so a list of prey codes has been in used in many labs doing this kind of work. When comes time to do analyses on these data one often wants to regroup prey in broader categories, especially for rare prey. In SAS you can nest a large number of "if-else", or do this more cleanly with "select"

Hello list, perhaps someone can guide me, finding out what's going wrong in the following scenario (Active Directory , Samba 3.0.20b same with 3.0.28a): CHILD1.CONTOSO.COM <-trusts-> CONTOSO.COM <-trusts->CHILD2.CONTOSO.COM | | | User: CHILD1\testtest | Samba

Hi list, This is my domain enviroment and all DC are windows 2008r2 http://i.imgur.com/8cNOtm2.jpeg When I used samba-4.0.5, I join my box to domain "HC1" , I got trusted domain "CHILD2" in "wbinfo -m". [/share/Public] # wbinfo -m BUILTIN MYBOX HC1 CHILD1 TREEROOT HC2 CHILD2 Then I upgraded my box to samba-4.4.4, I lost CHILD2 in "wbinfo -m".

Hi I use the following code and it stores the results of density() in the list dr: dens <- function(run) { density( positions$X[positions$run==run], bw=3, cut=-2 ) } dr <- lapply(1:5, dens) but the results are stored in dr[[i]] and not dr[i], i.e. plot(dr[[1]]) works, but plot([1]) doesn't. Is there any way that I can store them in dr[i]? Thanks a lot, Rainer -- Rainer M. Krug,