similar to: Crash in Asterisk

Hello my friends, I'm very worry about a problem i'm having...my asterisk got freez some times, every 5 or 6 days with NO trace in /var/log/asterisk/messages What i want to know is if safe_asterisk has something to be with this? This is what i have on my server: [root at mypbx ~]# ps -A | grep asterisk 9118 ? 00:01:30 asterisk [root at dreampbx ~]# ps aux | grep asterisk root

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

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

Hello my friends, My asterisk is going down randomly, following you will find some errors that i could see in the /var/log/asterisk/message at the moment of the crash: [Feb 5 10:32:45] WARNING[6519] chan_sip.c: Maximum retries exceeded on transmission 1850202354 at 10.4.1.152 for seqno 21 (Critical Response) [Feb 5 10:32:45] WARNING[6519] chan_sip.c: Hanging up call 1850202354 at 10.4.1.152 -

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

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

i was looking but i dont find how do this: configure the password for the extensions read the messages and some other things related with this can some bady help me with some material or a explicit example. thanks in advance Rodney Acosta Coya.

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

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

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

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

El 07/04/15 a las 17:38, Alex Villac??s Lasso escribi?: > I am trying to collect enough information about an problem a client is having with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution. > > Background: this client is a telemarketing call-center that generates outgoing calls with close to a hundred

I'm running an E400P. Every now and then Asterisk stops receiving incoming calls. This turns up in the messages log: Nov 25 10:49:12 WARNING[65541]: File chan_zap.c, Line 5793 (pri_dchannel): Ring requested on channel 1 already in use on span 1. Hanging up owner. Nov 25 10:49:15 WARNING[81926]: File chan_zap.c, Line 5793 (pri_dchannel): Ring requested on channel 1 already in use on

Have you tried Asterisk 13? The bridging mechanism has been completely rewritten on Asterisk 12, so there's no longer channel masquerading and zombie channels. Might be worth a try. 2015-04-07 20:33 GMT-03:00 Alex Villac??s Lasso <a_villacis at palosanto.com>: > El 07/04/15 a las 17:38, Alex Villac??s Lasso escribi?: > > I am trying to collect enough information about an

On 2/15/16 12:50 PM, Joshua Colp wrote: > Carlos Chavez wrote: >> Is it possible to use serveral protocols for a single transport section >> in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you >> cound use webrtc along with your phones but if I try: >> >> [transport-udp] >> type=transport >> protocol=udp,ws,wss >> bind=0.0.0.0 >

Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine. However, using outgoing call files the CS1000 is hanging up after I answer the call. I dont know why? Thanks, for any assistance. Jerry my sip.conf entry is: [Nortel] type=friend dtmfmode=rfc2833 username=XXXXXXXXX disallow=all allow=ulaw allow=alaw

I have been approached with a question that I am not sure about... A Shorewall system has only one interface, with a public IP-adress. The same system is the endpoint for a few OpenVPN-tunnels. Is it possible to add an aliased IP to the interface, and NAT traffic to a OpenVPN-endpoint? The endpoint is on 10.4.2.3 and the Shorewall-box has an interface of 10.4.2.1.