similar to: SIP METHOD BYE

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

> 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

Hello folks, since I started with asterisk 1.8.2 I got this messages in my console when finish a call. -- Executing [1610 at from-e1:1] Dial("SIP/xxx-00000027", "SIP/1610,60") in new stack == Using SIP RTP CoS mark 5 -- Called 1610 -- SIP/1610-00000028 is ringing -- SIP/1610-00000028 answered SIP/xxx-00000027 -- Locally bridging SIP/xxx-00000027 and

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

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

Asterisk 1.8.3.2 I have been getting this warning constantly on CLI in a call scenario where I use local channels to connect SIP with PSTN. I use callfile and local channel to first call a PSTN number and if answered, use local channel to call SIP phone with music on hold enabled in Dial string. If I call PSTN from SIP directly or vice versa I don't see this warning coming. On SIP I have

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

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

Fellipe, CXXR development has moved to github, and we haven't fixed up the build for using git yet. Could you send a pull request with your change to the repo at https://github.com/cxxr-devel/cxxr/? Also, this patch may be useful for pqR too. https://github.com/radfordneal/pqR Thanks On Mon, Jan 19, 2015 at 2:35 PM, Dirk Eddelbuettel <edd at debian.org> wrote: > > On 19

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

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

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

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

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

Has anybody encountered this problem? I have a box that has FC2 installed. I wanted to trash the FC2 installation and install CentOS 4. I have a pxeboot/dhcp/kickstart environment and so I tried automatic disk partitioning (clear all partitions and then create new ones), manual disk partitioning but everything else is automated and finally, zero automation, just a manual installation through

Hi all, I've read several discussions about asterisk adding 'received' parameter to the top Via header. In our case asterisk (release 11.4) gets BYE from sip proxy (with BYE top via header containing proxy ip address and port) but added 'received' parameter contains ip address from a 2nd Via (or from "From') and OK gets lost. I'm just trying to adjust sip

Hello, I seem to be having an issue with the configuration of my PRI on a new asterisk server I've created to replace an old install that I have. The card is Digium Wildcard TE133. I continually get messages like "Primary D-Channel on span 1 down", rather irregularly: [2013-07-29 17:31:39] VERBOSE[3621] sig_pri.c: == Primary D-Channel on span 1 up [2013-07-29 17:31:39]

Hi, im using a Asterisk Server which is not behind NAT. First i had problems with the fax detection. But this is now solved after adding a wait(2) at the correct place. But i'm still unable to receive a fax due to res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short after the Fax session has started. My sip.conf includes [general] allowguest=no alwaysauthreject=yes sendrpid=rpid