similar to: Group Dial

Displaying 20 results from an estimated 700 matches similar to: "Group Dial"

2004 Feb 02
7
cdr mysql problem
Can someone tell me what is wrong here: Feb 2 19:45:44 ERROR[1074441696]: cdr_addon_mysql.c:381 my_load_module: Failed to connect to mysql database asteriskcdrdb on localhost. The database is created, cdr table also, the username and password is right. I have tried configuring cdr_mysql.conf to connect via localhost mysql.sock or via tcp port, but in both cases I got this error. Thanks!
2003 Oct 15
2
skinny problem
has anyone seen this? -- Starting Skinny session from 192.168.13.102 -- Starting Skinny session from 192.168.13.102 triton*CLI> Oct 15 13:44:05 WARNING[213019]: File chan_skinny.c, Line 2243 (get_input): Skinny Client sent less data than expected. Oct 15 13:44:05 NOTICE[213019]: File chan_skinny.c, Line 2301 (skinny_session): Skinny Session returned: Success Oct 15 13:44:05
2004 Jan 31
2
TE410P E1 PRI problem
Hi everyone! Here is my configuration and messages taken from Asterisk startup. The E1 PRI trunk is connected to our national telecom company here in Croatia. When I call from outside over this trunk to my company I get 'error in connection' respnse. In the same moment I can't see anything in Asterisk, nothing that will tell me that the call reached Asterisk. I think there is a
2004 May 05
2
BUSY tone
Hi everyone, Maybe someone could help me. I have Asterisk in production with TE410P connected to PSTN. When I call from internal phones, either voip or connected via other PRI trunk, to PSTN and if the called phone is busy I don't hear anything!?! I should hear tone indicating that called number is busy. I have played with busydetect and callprogress in zapata.conf, but I didn't find
2003 Oct 18
6
x-lite
Hi everyone, Has anyone experience with xten.net's X-Lite SIP softphone and asterisk? I have a problem and I think X-Lite is not even trying to contact SIP proxy while dialing. Tomica ---- This mail was sent thru ZGWireless free network - www.zgwireless.net, Internet connection sponsored by Iskon Internet d.o.o. - www.iskon.hr
2003 Oct 20
4
SIP Nat Issue
Hi All Has anything been done to fix the issue where the * box is sat behind a nat firewall? Regards Mark
2003 Aug 25
13
SIP phones
Hi, I wonder if you guys can recomend a good SIP phone. A phone thats works great with * has a lot of features, and is cheap. Actually all kind pf VoIP hardware is of interesst. Is there a really good site for VoIP harware ? /Mike
2004 Jan 30
7
Calls dropping off
Hi, I've got a fairly working Asterisk setup, with a few minor glitches, one of which is very very irritating. Sometimes, during a call, the remote end just drops off. We're using software SIP phones (SJPhone) connecting to * then out through analogue lines with X100P cards. There is nothing in the logs and nothing on the console, the call just seems to 'go away'! Can anyone
2007 Dec 26
2
Two lines for outgoing calls
Dear All, I'm using Asterisk 1.4.16.2 with Zaptel 1.4.7 on Debian with kernel 2.6.18. I have two analog lines Zap/1 and Zap/2 as group 1 in zapata.conf. I'm using below context for dialing out. [outbound-local] exten => _9XXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten => _9XXXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten => _9ZXXXXXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten =>
2006 May 29
2
Asterisk Internal sip calls I can´t send/recive
When i made internal call into my LAN using x-lite sip phone client I retrive in askterisk CLI : ----------- ERROR ---------- Verbosity is at least 6 -- Remote UNIX connection -- Executing Dial("SIP/201-979d", "SIP/201|60|t") in new stack -- Called 201 May 29 18:09:28 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call
2010 Aug 30
2
help with dialplan
Todd How do you have the context in the phones sip configs set? Bryant From: "Todd Reese" treese65 at gmail.com Hi all, I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. My current problem is that the phones won't dialout.on the VOIP lines listed as dialout1, dialout2, dialout3. This version of asterisk
2006 May 12
3
Dial Command Reference for SIP channel
Hi all. I was reading a sample config someone had posted relating to call forwarding, and in it, they use a Dial command with components that I cannot find any reference to. Can someone point me to a reference which could explain the difference between Dial(SIP/100|20|Ttr,,wW) and Dial(SIP/100,,wW) Specifically, what is the |20|Ttr ? I cannot seem to find any reference which would indicate
2005 Aug 05
1
TE405P Dropping Calls
Hi, Urgently response would be wonderful, system is a Fedora Core 2. I have a Ericsson BP250 connected to 1 port on the TE405P and another connected to a local telco ISDN30. I have been running CVS-HEAD from about a 2 months ago and upgraded it again just in cause it was a version issue (didn't fix it) but this is what I am getting. When a person calls out from an extension on the BP250 to
2009 May 26
2
Discovery on packages
If I download a package -- how can I explore what it does? For instance, I downloaded the TTR package of off CRAN -- but how can I find out what it does? Is there a type of API doc for each package? JB
2011 Sep 13
1
Getting Rcpp SEXP data in C++
Friends I am looking at Rcpp and I am a bit stuck on a simple matter. (I am calling R from c++, if there is a better way...) Given this simple example using the TTR package and the SMA function which returns a simple moving average.... Rcpp::NumericVector rv; for(int i = 0; i < 100; i++){ rv.push_back(rand()); } Rcpp::Environment TTR("package:TTR");
2010 Aug 15
1
Moving average in R
Hi, I want to fit moving average trend in R. In google, I see that it is in the package 'TTR'. But, I can't install this package. I have used the following code: >install.packages("TTR") But, it says there is no package called 'TTR'. Can you help me? Regards, Suman Dhara [[alternative HTML version deleted]]
2004 Apr 12
0
Re: Asterisk-Users digest, Vol 1 #3402 - 17 msgs
Hi all, Can any one please help me in intergrating PHP/Mysql with my running asterisk server to configure IAX or SIP users? I will highly appreciate any help in this regard. Regards Nawaz. --- asterisk-users-request@lists.digium.com wrote: > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, >
2011 Aug 05
2
Which is more efficient?
Greetings all, I am curious to know if either of these two sets of code is more efficient? Example1: ## t-test ## colA <- temp [ , j ] colB <- temp [ , k ] ttr <- t.test ( colA, colB, var.equal=TRUE) tt_pvalue [ i ] <- ttr$p.value or Example2: tt_pvalue [ i ] <- t.test ( temp[ , j ], temp[ , k ], var.equal=TRUE) ------------- I have three loops, i, j, k. One to test the all of
2004 May 22
4
sip call using name in sip.conf
i try to place a call exten => _X.,1,Dial(SIP/${EXTEN}@foo:5061,60,Ttr) where sip.conf has an entry [foo] secret=torture callerid="local ext 103" <19146665555> type=friend fromuser=asterisk auth=both host=dynamic canreinvite=yes context=in-914 mailbox=001 i get May 22 23:11:31 WARNING[140400128]: chan_sip.c:902 create_addr: \
2010 Oct 10
1
Help needed for getYahooData in TTR package & writing the Yahoo data to excel
Dear all, I'm totally new to R. Recently I've been trying to use getYahooData in TTR package in order to download stock index daily open/high/low/close. The downloaded data is in the format of Open High Low Close Volume 2000-01-04 18937.45 19187.61 18937.45 19002.86 0 2000-01-05 19003.51 19003.51 18221.82 18542.55 0 2000-01-06