similar to: Adding an Event on chan_sip.c (asterisk 1.8.22)

Hi List, I did follow the procedure to install Free Fax for Asterisk successfully till i came accross this isssue: i can't load the fax module: pbx3*CLI> module load res_fax_digium.so Unable to load module res_fax_digium.so Command 'module load res_fax_digium.so' failed. [Sep 30 10:50:12] WARNING[5427]: loader.c:429 load_dynamic_module: Error loading module

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

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

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

static void senddialevent(struct ast_channel *src, struct ast_channel *dst) { manager_event(EVENT_FLAG_CALL, "Dial", "Source: %s\r\n" "Destination: %s\r\n" "CallerID: %s\r\n" "CallerIDName: %s\r\n" "SrcUniqueID: %s\r\n" "DestUniqueID: %s\r\n" "CDRUserfield: %s\r\n", src->name,

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

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

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

List, Does anyone know how to limit the amount of registrations that a sip user can have? For example, I have 2 softphones that I use on my laptop & desktop, both use the same username & password. If I have both softphones up at the same time, I can make simultaneous calls with each of them. I know you can have call-limit=1 but in this case, I want to allow them to have 3 way calling

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

Hello All, I upgraded Asterisk 1.8.10 to Asterisk 1.8.22 since upgrading I can't get meetme feature to work when dial meetme extension, can you please help? It always worked before, also I do not have dahdi installed on this machine, never did. -- Executing [104 at sipphones:1] MeetMe("SIP/101-00000813", "104") in new stack == Parsing

Hello, I want to share mailbox between two extensions Ext. 101 Ext. 102 I want the messages to go to mailbox 101, when when checked mailbox from extension 102 to be able to clear the bliking red light. here is extensions.conf exten => 102,hint,SIP/${EXTEN} exten => 102,1,Dial(SIP/101&SIP/102,20,t) exten => 102,2,Voicemail(101,u) exten => 102,102,Voicemail(101,b) exten =>

Hello, I have a fully functional Asterisk Server, I want to configure this server to be able to process call from Skype, can someone point me to a howto? or if there are suggestions on best way to approach this problem. Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, recently I have seen spike in attacks on my asterisk server, this is what I get on the LCD of my phone: 201 at 76.220.5.205 or calls from 1000 sip1000 at 76.2230.5.205, have any idea on how to stop this calls? Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good Day everyone, Yesterday I upgraded the firmware on my 7960 to Sip 8.12 as provided by Cisco, however now the phone does not and will not read the RINGLIST.dat file. I've tried rebooting the phone, tried resetting the phone back to factory, have deleted the RINGLIST.dat file and reloaded the phone then reinstalled the RINGLIST.dat, and still the bloody phone will not read the file.