similar to: Get data from the SDPof SIP INVITE message

B.H. Hello, i'm using AMI Originate action (with async=true) to send outgoing calls to a SIP trunk (using asterisk-java library to connect to AMI). The problem is that in case of failed originate, OriginateResponse event is returning only the reason code which is sometimes not sufficient to determine the real cause of failure. Also, there's no way to link between the channel that was

B.H. Hello, all. We have noticed many calls on our PBX get "stuck" - the other end sends BYE, and our side sends ACK but the call remains active (no hangup event on AMI, the call is listed in 'core show channels') and it's impossible to hang up until asterisk is restarted. Asterisk's log shows lots of messages like this: chan_sip.c: Autodestruct on dialog .... with

B.H. Hello! We have several Asterik boxes that are connected to PSTN using PRI cards and they are interconnected using IAX2 trunks so that incoming calls are delivered from PSTN to the servers they belong to. In past we were using asterisk 1.4 on the server that is receiving IAX connections and everything worked as expected. Recently, we have switched to a newer box with asterisk 1.8.22 and

B.H. Hello, all :-) We have a cluster of Asterisk (v. 11.9) servers that host IVR applications. The servers work behind SIP proxy (kamailio) for load balancing. All servers are in 2 processor configuration, 8-10 cores per CPU. When a particular server gets about 500 concurrent calls, the sound quality begins to degrade, the sound plays slowly and with clicks. As far as i understand, it's

B.H. Hello, all. We have a dialer software that runs outgoing telephony campaigns. We have been using it successfully with PRI cards, now we're evaluating it's use also with a SIP trunk. Most of the things run perfectly good without a need to change anything except for dial string, but there's some strange problem with asterisk interpreting SIP result codes. Our software is written

B.H. Hello, all :-) We have some OPENVOX D410P PRI cards and we are successfully using them with Asterisk boxes which are based on stock ubuntu 12.04 DAHDI and Asterisk packages. The card is recognized by DAHDI as 'Wildcard TE410P (2nd Gen)' and it uses wct4xxp driver. Now, i'm trying to run this hardware with DAHDI 2.6.2 package which is available from asterisk.org site and looks

Hi All, I;m using asterisk 1.4 with FreePBX and a Grandstream 4108. I am noticing a delay calling in and out via the FXO, but calls to local extension are ok. What i noticed when i used ngrep is that, it sends invite but got no response from the server, send another invite but got no response again, then again until it finally gets it. but if you will notice on the 2nd ngrep, the asterisk

Hi, I have two asterisk machines side by side both running debian sarge, one running sarge's version of asterisk (1.0.7.dfsg.1-2) and the other running the version of asterisk from www.backports.org (1:1.2.1.dfsg-2bpo1). I also have a SIP provider who is routing blocks of DID's to both machines. The sip.conf is nearly identical on both machines (the general section, and the section

I've being testing a couple of GrandStream ATA 286 which with no reason start responding 486 Busy to all new incoming INVITES. They are connected to an Asterisk installation as SIP client. Running ethereal between them, I could notice that, for some reason unknown for me at this time, Asterisk sends some stranges INVITE's AFTER the communication has been established and acknowledged

I have the following in sip.conf. It was my understanding that this configuration (ie with deny/permit) would only allow connections from hosts 192.168.10.4 and 192.168.10.5. That doesn't seem to be the case. Asterisk is accepting INVITE's from other addresses. [a00090101] type=friend context=Company1 username=a00090101 ;secret=180 ;insecure=very host=dynamic mailbox=company1@vmusers

Hi, Got problems with incoming SIP calls. Scenario: Server1: 3cx or any other server Server2: Asterisk 16.2.1 . PJPROJECT 2.8 Server2 registers on Server1 with SIP ID 1121. Registration is OK. Server2 outgoing calls are OK. INVITE, unauthorized, INVITE with password, OK, RINGING,... Troubles with incoming calls / incoming INVITE's . I can not identify endpoint by IP, I have multiple

I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration was based on the reference at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but has been heavily modified. Currently asterisk runs on localhost and only listens on SIP/RTP at 127.0.0.1 . Therefore, all of the SIP traffic appears to

dear R users, I would like to fill a circle with yellow stripes instead of a uniform yellow color. To draw the circle I used the following command after having loaded the (very nice !) plotrix library : library(plotrix) pdf("MWE.pdf",width=8, height=8)

On Mon, 2 Mar 2015, Mordechay Kaganer wrote: > When a particular server gets about 500 concurrent calls, the sound > quality begins to degrade, the sound plays slowly and with clicks. As > far as i understand, it's because asterisk is unable to send the voice > stream in time i.e. the server is overloaded. > > What i don't understand is, at the time that the server

Sun and Informatica Present "Metadata Solutions: Enabling the Intelligent Enterprise" Date: Wednesday, August 27, 2003 Time: 11:00amPDT/1:00pmCDT/2:00pmEDT Duration: One hour In today's economy, having in-depth and accurate data is key to getting visibility across your enterprise. Metadata management is required to access the impact of change and improve operational performance.

I don't see a question. If your question is whether R supports pattern fills, AFAIK it does not. If that is not your question, ask one. -- Sent from my phone. Please excuse my brevity. On June 14, 2017 7:57:41 AM PDT, jean-philippe <jeanphilippe.fontaine at gssi.infn.it> wrote: >dear R users, > >I would like to fill a circle with yellow stripes instead of a uniform

I have a 45 x 16 data frame consisting of dissimilarities among 10 colors, giving in each column the 45 = 10*9/2 pairwise judgments for one of 16 subjects. The rownames identify each pair of colors, e.g, "AC" = ("A","C"), and the pairs are ordered by columns in the lower triangle of each distance matrix. > helm.raw <-

In Asterisk 1.8, I used the following line in extensions.conf to allow me to pass "*82" in front of a dialed number, to disable the callerid block that's normally on that POTS line: ; disable callerid block exten => _*82.,1,Dial(${POTS}/${EXTEN}) But this seems to have stopped working when I upgraded to Asterisk 11.7. I get the following debug output, with a "no

Hi, I am new to R so forgive me if the following query is somewhat simple.  I have a small tab-separated file with n records of glucose values, one record per day, seven measurements per day. It looks like this: date    sober    no    vm    nm    va    na    vs 20091229    NA    6.8    NA    2.7    11.7    2.7    6.2 I'd like to make a graph on which the glucose day curves are plotted

hi, i need a function that can replace na's in univariate ts objects with a value interpolated linearly using the adjacent non-na values. for example, the function i need (myfun) would do the following > x <- ts(10,11.4,NA,12,9.7) > y <- myfun(x) > y 10 11.4 11.7 12 9.7 i can code an na.action method myself to accomplish this, but has someone else already coded routines of