similar to: IAX calls dropout on button press

Dear R users,   Please assist me with the following problem. I have a dataset that looks like the following:   dat<-data.frame(   'id'=rep(c(1,2,3),each=3),   'time'=rep(c(1,2,3),3),   'y'= c(2,2,NA,2,NA,NA,2,5,7) )   I wish to create a variable for dropout time in dataframe 'dat' such that the dropout time is the time to drop out by the subject as follows:    

Hi, I need some help again. :-) I have "Too much recursion error" prototype.js line 1288, I heve this error when I try to use Effect.dropout (script.aculo.us) if I remove table ID, there is no error, and the effect works fine (remove the tr from table) But I need this ID assign to this table, as table ID is used by other effect (Builder.node) Please help .... Thanks YUAN Here is

The problem is some calls from the PSTN have hidden caller id so if you want to change it to something else then modify chan_sip.c #define CALLERID_UNKNOWN "Asterisk" I've changed mine to: #define CALLERID_UNKNOWN "Unknown" -----Original Message----- From: Shaun Ewing [mailto:sewing@gmail.com] Sent: 22 September 2004 14:16 To: Asterisk Users Mailing List

I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make outgoing calls via it fine, however incoming calls are dropped after a few seconds ( or as soon as a command like Playback, or the call is picked up if forwarded to a SIP

Hi, I have an MGE Ellipse 1000 connected to a FreeBSD 7.1 system and it works well except that if I yank the cable it doesn't detect a problem.. It seems to quite merrily read the old data (upsc reports the same values). There is nothing logged by NUT to indicate comms is lost (usbhid-ups is still running). In the attached log I yanked the able at the 28 second mark and plugged it back

Checking the earlier mails, it stated that the phones were $75 (100) & $85 (102) ref :- http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the person said there was no price change. Anyone on this list actually bought them at the $75 & $85 rate ??? Regards...Martin -- Too much is just enough.

Hello everybody. I have an customer asterisk 1.0.5 running well since 3 monthes, 2 TDM cards 4 FXO, 4 FXS. Since one week, unable to pass call between Zap and Sip getting the "exception on 15, channel 1" The * box is connected to an eads PBX and it seems that failure started when they make some changes on the PBX. Have someone an idea and what is causisng this failure? Here are the

Hi - I've recently bought a mediatrix 1204 and have had a complete nightmare getting it up and running with an asterisk@home setup. I know this isn't a mediatrix list but I'm at my wits end and the support with this product is atrocious. (mine was even shipped with firmware that was incompatible with the win32 software it came with so I wasted a day trying to work out why the SNMP

Hi, ( As a total newbie to R ) I get this error when attempting to use read.table to read a data frame: "Error in type.convert(x, na.strings, as.is) : null string encountered" (Is there help documentation for the type.convert function ( I could not find it in the usual places) This does not always happen, but seems to occur when I am attempting to read data frames with more than a

Hello ppl, The scenario : I restart asterisk, sip show peers shows nothing. I make a call from 7013 to 7011. I get the following o/p : SIP Seeding peer from astdb: '7013' at 7013@192.168.10.53 for 3600 SIP Seeding peer from astdb: '7011' at 7011@192.168.10.45 for 240 And then the call goes thru. So, does 'Seeding', means * registers both users?? But a subsequent

my monthly archive script does: echo `date` start ${i} doveadm mailbox create \#ARCHIVE/${YEAR_LAST_MONTH}/${i} doveadm -f tab mailbox status messages ${i} doveadm move \#ARCHIVE/${YEAR_LAST_MONTH}/${i} mailbox \ ${i} BEFORE ${TODAY} SINCE ${FIRST_LAST_MONTH} doveadm -f tab mailbox status messages ${i} echo `date` done ${i} for each mailbox that has >= 1 message in

Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The

I've installed Asterisk from the Asterisk @home distribution. Ultimately I will be using Asterisk for voicemail for about 150 users. Calls are (mostly) handled by a legacy PBX although we do have a couple of Cisco 1760 routers that connect a remote office. I've setup a SIP trunk that routes calls from Asterisk to the 1760, and that works fine. I've also configured one of the 1760s to

I'm trying to get working a spa3000 with asterisk. My problem is I cant get wroking the incomming calls (I installed the lastest firmware). My problem is asterisk is rejecting the authentication from the spa3000. Asterisk answers forbidden (SIP/2.0 403 Forbidden) and I think I placed the username and password correctly... Sip.conf says this: [linea2] username=linea2 type=peer secret=1111

Hello, I got Quintum A800 with the P5-2-1 firmware. I configure my asterisk trunk as followed: [SIP_BD1] type=peer qualify=yes host=192.168.0.254 disallow=all context=from-pstn allow=h723 and inside the quantum I change the config sip useragent to 5060. Up to this part if I run sip show peers, I got: asterisk1*CLI> sip show peers Name/username????????????? Host??????????? Dyn Nat ACL

It works for me. I don't know why it wouldn't work for you. Looking at the autoindexing code I don't see how it could be possible that it works for saving but not copying. > On 10 Sep 2015, at 21:05, Larry Rosenman <larryrtx at gmail.com> wrote: > > Is there a fix coming for this, Timo? Or is it a longer term issue? > > On Mon, Sep 7, 2015 at 5:23 PM, Larry

Is there a fix coming for this, Timo? Or is it a longer term issue? On Mon, Sep 7, 2015 at 5:23 PM, Larry Rosenman <larryrtx at gmail.com> wrote: > It doesn't in my current 2.2.18 setup with the config I posted. > > > On Mon, Sep 7, 2015 at 5:22 PM, Timo Sirainen <tss at iki.fi> wrote: > >> It should. >> >> On 08 Sep 2015, at 01:01, Larry

On Tue, 2003-12-23 at 19:22, Brian West wrote: > I'm just double checking.. I was told it wasn't possible but i'm going to > ask just in case. > > Can you set outbound callerid on a channelized T1? > >I think there is a way to do something like DID with the 4 digits of >DTMF passed before the call. It is unlikely though that you will find >someone interested

Hello, I have read the information on echo and SIP in the FAQ and I have scoured the mailing list for possible solutions, but as yet I have not been able to get rid of this echo. I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed into an asterisk server. If I call between the Sip Phone (Budgettone-100) and the 4 FXS ports everything sounds great. If I call out to the PSTN

It doesn't in my current 2.2.18 setup with the config I posted. On Mon, Sep 7, 2015 at 5:22 PM, Timo Sirainen <tss at iki.fi> wrote: > It should. > > On 08 Sep 2015, at 01:01, Larry Rosenman <larryrtx at gmail.com> wrote: > > should fts_autoindex handle that case? > > > On Mon, Sep 7, 2015 at 5:00 PM, Timo Sirainen <tss at iki.fi> wrote: >