similar to: Is SIPPEER curcalls working for you ? [SOLVED]

Hi, In this thread http://lists.digium.com/pipermail/asterisk-users/2008-October/219592.html , I wondered whether SIPPEER curcalls was working. I could test this anew today. Here are my findings : Alice, Bob and Carol ar all using SIP Phones. Whenever Alice is calling Bob, - if Carol is calling Alice, SIPPEER(Alice:curcalls) equals 0 - if Carol is calling Bob, SIPPEER(Bob:curcalls) equals 1

Greetings, I am writing an AGI script that needs to check on the idle/busy status of a number of SIP peers (mostly SPA9xx phones, with a few Polycoms and Snoms thrown in for fun). Is it possible to call Asterisk functions (e.g. SIPPEER) from AGI scripts? Based on my Googling, I would guess in the negative. I have tried various permutations of Set() and Eval() without success. I have also

On Wed, Jan 20, 2010 at 2:42 PM, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I'm using Asterisk 1.4 branch and checking the status of some SIP > Peers with the functions ${SIPPEER(101:status)} and the result is "OK > (48 ms)". ?Seems to work fine. > > Now I would like to use the function CUT to set a variable with the > 'OK'

Hi All, I'm using Asterisk 1.4 branch and checking the status of some SIP Peers with the functions ${SIPPEER(101:status)} and the result is "OK (48 ms)". Seems to work fine. Now I would like to use the function CUT to set a variable with the 'OK' portion of the status "OK (48 ms)" and then do some follow on stuff if the status is OK. I'm running into syntax

Hi, When I need to route calls depending on the number of (incoming and outgoing) calls a SIP device is currently handling, I mostly use function SIPPEER and its curcalls option. I can read and there references to function GROUP for the same usage, but I intuitively thought that though this method also applies to non-SIP devices and a large range of asterisk versions, it would require more work

Hi, In sip.conf, it's possible to add a line such as setvar=MYFIELD=foo and access this value from diaplan with SIPPEER function. 1. Which function is available to access values in users.conf such as vmsecret ? 2. Is it possible to extend users.conf with custom keys/values ? Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, With setvar statements in chan_dahdi.conf, we have a convenient way to store DAHDI channels specific values. Unfortunately, we don't have a function to access this data from the dialplan as easily as SIPPEER ou IAXPEER would for SIP or IAX trunks. Using AST_CONFIG, you can access DAHDI setvar value but: 1. only one setvar value (see bellow) 2. AST_CONFIG reads values from current

Hi, I have a question. How can I configure my sip.conf to make a SIP phone busy on incoming and outcoming calls? I explain my problem. When SIP phone receive a call and then I try to call that phone, I find it busy. When SIP phone make a call and I try to call that phone, I find it avaible and it rings but I want to find it busy. I configure sip.conf like following: [10] type=friend qualify=yes

Hello all, I've been testing a Kamailio Asterisk Realtime integration, and found a strange situation. My problem is that when using the integration, everything seems ok but Asterisk does not see the clients as registered. Kamailio and the clients report registered clients. Also calls fail. In Asterisk cli sip show peers shows nothing but for example realtime load sipusers name 660 shows the

Hello, I have a problem with a call between 2 webrtc clients. Asterisk removes the ice-related lines from the sdp when it sends the INVITE out, and the called webrtc client rejects the INVITE due to the missing ice lines. Both webrtc clients are defined exactly the same way, same values in all fields except the number of the peer. There's probably something I've changed that causes this

Hi All, PLEASE READ if you depend on Asterisk CDR's and support transfers. Apologies for the shout but I'm desperate to get others to agree Asterisk has a big problem with the CDR's that are generated for transfers. I can understand why not too many people are interested as transfers are complicated and messy. However for those of us having to support transfers and depending on

Hi Every one, I have a dataframe "class" with "name", "sex", "age", "height", "Weight". if i caluclate summary statistics with the below code numSummary(class[,c("Height", "Weight")], groups=class$Name, statistics=c("mean", "sd", "quantiles"), quantiles=c(0, .25,.5,.75,1)) iam getting

Hello. I'm using Asterisk 1.4.22 with Postgres 8.3 in a Ubuntu 8.04 Server. I configured Asterisk to get sip from Postgres, and set qualify for all sips as yes, but the sip show peers command show the status of the peers as UNKNOWN srvcentral*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status Realtime 4900/4900 (Unspecified) D

Here is what I have in my extensions.conf file now. Trustrcid and sendrcid are turned to "yes" in the conf file. I'm not fully sure how the SIPCalledRPID works though. The example I found seems to try and provide the stuff automatically (id and name), but does the SIPPEER stuff even exist? I think this is probably the right track though. Any insight would be much appreciated.

I'm trying to use samba to serve a number of home directories to other samba Linux clients on my network. ie. I have a set of client machines that want to use the /home directory that my samba server will provide I thought I could set up a share that goes something like this [homedirs] path = /home writable = true On a client machine, I smbmount //myserver/homedirs

It seems that there is a problem in displaying subtitle in general, independently from multi-plot display. when I do plot (c(1,2,3), c(9,8,7), type = "l") title(main = "Main title", sub ="Sub title",cex.main=2, cex.sub = 2) subtitle doesn't get displayed > --- On Sat, 11/21/09, David Winsemius <dwinsemius at comcast.net> > wrote: >

hi list, environment: Windows 2008 Domain Centos 7 server running samba 4.4.4 Problem: I am creating a number of samba shares on the Centos server to be used on the Domain. Right now I have two major directories setup as shares with minor directories being created. How do I specify read/write permissions on the minor directories without having to create a share for each directory?  When I

I am using R-2.10.1 binary from CRAN on a WinXP Pro system. I also use SAS v9.2 on the same box. I just started using the SAS ODBC driver that comes with version 9 of SAS. I have been able to set up an ODBC source for SAS datasets using the driver, and then with RODBC I am able to read a sample SAS dataset. > library(RODBC) > ch <- odbcConnect('sasodbc', believeNRows=FALSE)

Hello We have recently upgraded our internal PBX from 1.4 to 1.8. This made the BLF lamps on our Polycom phones stop working. After a lot of googling and a lot of testing, I have been unable to find a solution. I did try to change the call-limit value from 4 to 1, and this actually made BLF work (noone suggested this, and what documantation I can find states that this option is deprecated). This

Hi, I''ve started learning about EventMachine right now and have some basic doubts. Hope you could clarify them. I''m developing a SIP server. For now I''ve started it from scratch but I expect to migrate it to EventMachine. SIP is a very complex protocol. For example: if I use a SIP proxy in front of my server then all the data will arrive to my server using the same TCP