similar to: "chan_sip.c: No such host:" but I can resolve it from command line ?

Dear R useRs, I have a vector with positive and negative numbers: A=c(0,1,2,3,0,4,5) Now if i-th element in vector A is > 0, then i-th element in vector B is a+1 else i-th element in vector b=a (or 0) vector A: 0 1 2 3 0 4 5 vector B: 0 2 3 4 0 5 6 What's the right way to do this. I still have some problems with for and if statements... Cheers, Andrej

Dear useRs, I have two different length vectors: one column (1...m) and one row vector (1...n): 20 40 20 60 5 4 2 Now I have to calculate ratios between column vector elements and each row vector elements: 4 5 10 8 10 20 4 5 20 15 12 30 Thank's in advance for any suggestions, Andrej

Hello! I have this situation / interfaces: Dsl0 - internet interface Eth0 - local network I have linux box with shorewall 2.2. And on the local network I also have a hardware router. I have connected WAN port with settings of my linux box and then created one more local network behind hardware router. It works fine. I then wanted to use VPN function of this hardware router, so i created

Dear R useRs, is there any simple, build in function to match specific regular expression in data file and write it to a vector. I have the following text file: *NEW RECORD *ID-001 *AB-text *NEW RECORD *ID-002 *AB-text etc. Now I have to match all ID fields and print them to a vector: 001 002 etc. I know that this is very simple with Perl or R-Perl interface, but if possible, I want to do

I just want to ask how to enlarge the resolution of my plots in R. For ex. for publising I would like a picture of resolution minimal 500 dpi, all I managed was picture of dim 5,28 X 5,83 with 118 pixels/cm. Thank You for the answer! With best regards, Darja Rupnik [[alternative HTML version deleted]]

Dear R useRs, I have big (23000 rows), vertical bar delimited file: e.g. A00001|Text a,Text b, Text c|345 A00002|Text bla|456 ... .. . Try using A <- read.table('filename.txt', header=FALSE,sep='\|') process stop at line 11975 with warning message: number of items read is not a multiple of the number of columns I have no problems with processing similar file, which is

Hi, I'm getting this error when registering with SIP server using Asterisk 1.4.10 and Freepbx... I'm getting this error no matter what I try to setup in sip.conf : - I'm getting confused whether options are maxexpirey=36000 or maxexpiry=36000 ? - Can I solve this with some settings in sip.conf or is this problem harder ? - I've read something about Asterisk's bug on this

how about redirecting the smbpasswd file to the older version (assuming you have one) using smbpasswd file = /file/path/smbpasswd , I replaced my copy of smbpasswd for 3.09 with a 2.216 and the smbpasswd command stopped working, (no new entry added to the smbpasswd file), but when i used that it worked again "Izo" <I@siol.net> wrote in message news:41B8004E.8050807@siol.net...

Hi Ville, Thank you for the patch. On Fri, Apr 03, 2020 at 11:39:54PM +0300, Ville Syrjala wrote: > From: Ville Syrj?l? <ville.syrjala at linux.intel.com> > > Get rid of mode->vrefresh and just calculate it on demand. Saves > a bit of space and avoids the cached value getting out of sync > with reality. > > Mostly done with cocci, with the following manual fixups:

Hi All; Asterisk version is: 1.8.5.0 But I see at the consol the following warning and really I did google but did not understand if it is bug or related to settings: [Sep 13 15:04:56] WARNING[2209]: chan_sip.c:19667 handle_response_invite: just did sched_add waitid(3429468) for sip_reinvite_retry for dialog 3c581fa96f2b-53yysntgjmwb in handle_response_invite But actually, we see some SNOM IP

Hello, I need help for that error message: ?chan_sip.c:14889 handle_response_invite: Failed to authenticate on INVITE to? My network is: Client1-- -----------asterisk1------asterisk2 Client2-- ? With client1, I do a call ? Asterisk1 forward the call to asterisk2 ? Asterisk2 forward the call to asterisk1 ? Asterisk1 forward the call to

I have a situation here where a user has an AAstra 480i phone, which function corectly. The phone is behing a nat-router (a linksys wrv200 for it's VPN point to point facility). The phone is plugued in a port wich has qos enabled. And when the user places a call, sometimes (not always), we get this in the console : [May 8 13:41:55] WARNING[5804]: chan_sip.c:1948 retrans_pkt: Maximum

Hello All This or similar topics have already been mentioned but without any solution yet. I have built a oneway conference system for a client using one caller's input and broadcast it to all the other participants using app_meetme. E.g. one talker multiple listeners. Unfortunately some of the talkers (I have got multiple rooms) are not loud enough (e.g. use just half the amplitude, so

Hello, I?ve been using Asterisk 1.2.18 for a while, and today, with no apparent changes, I started receiving these messages: Aug 31 13:26:57 NOTICE[27528]: chan_sip.c:5495 sip_reg_timeout: -- Registration for 'user at sipserver' timed out, trying again (Attempt #19) All trunks and extensions went to: sipserver:5060 user 120 Request Sent 011

Hello, I'm testing with asterisk 1.8.3.2 and come across this: Call from one extension to another with: [macro-internal-call] ;ARG1=extension to call exten => s,1,Set(TOCALL=${DB(SIP/${ARG1})}) exten => s,2,Dial(SIP/${TOCALL},60,tT) ... As I had no entry in the asteriskdb, so the SIP uri was empty, and asterisk core dumped with: gdb output: #0 0xb7c7db33 in strchr () from

Just put a new phone in place with the latest firmware from Cisco. This is the first SPA501G we have with this firmware. In the Asterisk CLI we are now seeing the error message below about once every second. When we unplug the phone, the messages quit. NOTICE[15539]: chan_sip.c:9602 copy_header: No field 'CSeq' present to copy Thanks in advance for any assistance on this.

Anyone else have issues with T.38 where the call drops after T.38 is attempted to be negotiated, with a message like the below? WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101'

Hi All, I'm running trixbox 2.0. The problem: a remote extension behind a NAT, can call other extensions, can call any other party, can call voicemail, will ring when rung, but when answered there is nothing and the dialling party continues to hear the ring tone. I'm getting this error in the logs: "That's odd... Got a response on a call we dont know about" I see

hi, can i ask what settings do you recommend for a lot(1000-10000) of different sip phones which are behind NAT(many different routers)? i have qualify=5000 nat=yes cli>sip show settings Reg. min duration 60 secs Reg. max duration: 3600 secs Reg. default duration: 120 secs Outbound reg. timeout: 20 secs Outbound reg. attempts: 0 asterisk 1.4 thanks

Hi, I'm using an Asterisk 1.2.18 box with a remote Snom 360. My Snom always rings but sometimes (it happens randomly!) no voice is passing thru (2 ways). Asterisk CLI shows this warning: Jan 10 10:03:26 WARNING[19164] chan_sip.c: Forbidden - wrong password on authentication for INVITE to '"unknown" <sip:unknown at xxx.xxx.xxx.xxx> I have already set localnet and