similar to: Channels stuck on CONFBRIDGE_INFO

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

Hi all, I'm using ConfBridge within Asterisk 1.6.20 and want to record the conference, so I'd like to start the recording when the second user joins, so in the example below, for example, how can I get the current user count in ConfBridge 3000? [conferences] ;authenticated conference (ext C-O-N-F = 2663) exten => 2663,1,Answer same => n,Wait(1) same => n,Authenticate(143382)

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! 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

Hello, I'd like to use the AMI interface to originate a call to a context in a dialplan, and handoff the dial control to the context. Whenever I execute the below action, the recipient does ring, but when I answer it dials the recipient again. I believe this is because once answered the system is going to execute the Context/Exten/Prio in the Originate action? Action: Originate Channel:

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

B.H. Hello, all I'm using Asterisk 11.7, connected to PSTN using SIP trunk. I'm looking for a way to get data from INVITE's SDP. Specifically, i would like to get a value of o= for incoming call from PSTN because it contains data about the operator that the call originates from. I have googled for a solution and found this patch:

Any tips on solving the following performance conundrum: Asterisk 1.8.12.2 running on HP DL360 G5 and G7s tcpdump running to capture UDP 5060/SIP signaling to .pcap files All calls are ultimately B2BUA client -> asterisk -> PSTN Media stays on Asterisk at all times AGI script has exit handler that connects and updates an external database upon BYE from either side. I know that if exit

All; What I want to do is create a way to easily send callers into a conference room to have an N-way conference call. I created an extension '100' that calls the MeetMe() command. Then all I need to do is transfer a caller using a blind transfer (*2 in my case) to extension 100. Then I can dial a feature code that sends me into that conference (*15 in my case). So far, a piece of

In some situations I got the following message on asterisk console: * Autodestruct on dialog '857128833 at 192.168.2.129 <857128833 at 192.168.2.129>' with owner SIP/1015-00000002 in place (Method: BYE). Rescheduling destruction for 10000 ms* I would like to raise a manager event, to take some action when it is happening. To do so, I believed that was just a matter of adding an

I think that there are no qdiscs that permit to drop the oldest frame of a queue when this queue is full, but I would like to be wrong: bfifo drops arriving frames when the max queue length is reached. red also drops arriving frames in a more elaborate fashion, with a drop probability that increases above a limit and becomes a drop certitude when the max queue length is reached. sfq drops

My friends, I'm having some problems in my Asterisk, the thing is that Asterisk seem to be crashed (or dead) sometimes (2 times in 3 weeks) I noticed this today, when i could not make any internall call, tha calls to the voicemail (*1) did not work it just don't say nothing, nothing appears in console; i tried to make a CLI>stop now but nothing happens, i could not stop the asterisk

Hi All, This is the first time I'm submitting a patch, and I hope it fixes more than it breaks. I'm putting it here, since John Todd mentioned a while ago about the heavy load Mark and crew have at Digium (doing such good work), so I thought all of us could test this first, and if ok submit for inclusion in CVS later if appropriate. This is an extension to work done earlier (sorry I

Hi All I have dozens of these messages on CLI complaining about database connection and error writing CDR to disk. The curious thing is I can find them all inside the database. I "selected" them using uniqueid and manually compared each column with the cdr_adaptive_odbc.c error line. "mysqlcheck -a -e -v DBase" and "mysqlcheck -c -e -v DBase" both returned OK for

On Sat, 25 Apr 2015 17:11:34 +0200 jg <webaccounts173 at jgoettgens.de> wrote: > > > Hi All > > > > I have dozens of these messages on CLI complaining about database > > connection and error writing CDR to disk. > > > > The curious thing is I can find them all inside the database. > > I "selected" them using uniqueid and manually

Hello everybody! I get this message when making outbound calls: [Feb 21 14:24:46] WARNING[25204]: chan_sip.c:3621 __sip_autodestruct: Autodestruct on dialog 'ee162385cac5cc9c at 10.1.1.13' with owner in place (Method: BYE) All inbound calls are fine. In other SIP users everything seems fine and I can make outbound calls. Asterisk: 1.8.2/ Any idea??? Best regards, Fellipe

I have built a dialplan which dial to someone with option M. Dial (SIP/1000,,M(MYMACRO)) Both parties are SIP phones. MYMACRO expects person on SIP/1000 dial 5 (using read) then exits - and doing so it bridges my phone (SIP/2000) with SIP/1000. If SIP/1000 hangs up before dial 5 - ok the call ends. if SIP/2000 hangs up before SIP/1000 dial 5 - the macro is unaware and keeps waiting SIP/1000

Asterisk 11.11.0 on CentOS 6.5, installed from RPMs. OpenSIPS fronting Asterisk from a Tekelec T9000. I'm accumulating stuck channels. I'm googling now and I recognize that Friday afternoons are the worst time to ask questions, but I'm getting desperate because this is keeping me from rolling a system out to production. (Yup, I know. Who rolls out a system on a Friday

> Hi All > > I have dozens of these messages on CLI complaining about database connection and error writing CDR to disk. > > The curious thing is I can find them all inside the database. > I "selected" them using uniqueid and manually compared each column with the cdr_adaptive_odbc.c error line. > > "mysqlcheck -a -e -v DBase" and "mysqlcheck -c -e