similar to: Asterisk SIP DNS srv records

If anyone is interested ... just in case! :-)... I have tried to write , based on the cdr_mysql.so module, an Sybase module. To compile you can use something like that: export SYBPLATFORM=linux export SYBASE=/opt/sybase cc -I$SYBASE/include -c -o cdr_sybase.o cdr_sybase.c cc -shared -Xlinker -x -o cdr_sybase.so cdr_sybase.o -lsybdb -lm -L$SYBASE/lib (anyone could write the corect Makefile

Hello *, is the rtpip patch still valid for asterisk 1.6 (with some code changes, obviously)? https://issues.asterisk.org/view.php?id=8161 Or, in asterisk 1.6 there is an alternative to using it? This is the difffile I produced for chan_sip.c in asterisk 1.6.2.11 --- chan_sip.c 2010-10-12 13:47:49.000000000 +0200 +++ chan_sip.c.orig 2010-10-12 13:47:27.000000000 +0200 @@ -987,9 +987,6 @@

I'm trying to clean up some notices/warnings that are repeatedly logged in *.Any Help would be appreciated as I'm not sure of the cause /solution. Here are the errors: Nov 17 15:53:38 WARNING[1217602880]: File chan_zap.c, Line 1321 (zt_call): cidspill already exists?? +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ /* Don't send audio while on hook, until the call

Hi, Since Carl has kindly provided us with fax support for CAPI based cards, we have been using it with much success. Today I have modified app_capiFax so that it now supports a dynamic CSID. The following example uses the DNID created by chan_capi on an AVM Fritz! card. * Receive a fax with CAPI API. * Usage : capiAnswerFax2(path_output_file.SFF|stationID) * * This function can be

I forgot to take out the portion that would read in the voicemail boxes from the text file. If you want to leave it in then you could have some voicemail boxes defined in the text voicemail.conf. I do not, so I have removed it. Below is the new patch: *** app_voicemail.c 2004-06-23 07:55:54.000000000 -0600 --- app_voicemail.c.new 2004-06-23 07:55:47.000000000 -0600 *************** *** 49,61 ****

Hello all, I am just getting going on building my system, but I thought I'd send you all a patch that I wrote so the command: show voicemail users issued from the CLI works properly when there is a postgres backend for the voicemail. The current version of the app does not display the voicemail boxes found in a database. It is called in the load_config function. I haven't done

Hi, As you asked, I have included my diff to what I did for the DIAL command. I probably didn't stick to some * pre-agreed standard of coding or something, so if these things offend you then I suggest that you close your eyes. :) The biggest thing to consider when you are doing a prepaid system is, what if the person with the same account in/out calls twice? I chose, for now, just to keep

Your log file almost looks like a bug in Asterisk doesn't it? Why call poll() with a zero timeout while passing only one FD? and then why do the read when there is no data? Read the man pages for all the system calls Take a look at the source chan_sip.c /* Wait for sched or io */ res = ast_sched_wait(sched); if ((res < 0) || (res > 1000))

Hi, This patch fixes a couple of segfaults in music-on-hold, frame smoother routines and channel allocation in Asterisk. Mark, feel free to apply it in CVS (if approved). Regards, Michael. -------------- next part -------------- Index: channel.c =================================================================== RCS file: /usr/cvsroot/asterisk/channel.c,v retrieving revision 1.25 diff -u

Hi All I just completed a custom application for Asterisk (i m not a C guru so i just copy codes from other application and alter according to my needs) attached files is the source file this application is working fine but still i need you people to give suggestion to improve it Primary task of this application is to get a parameter from extensions.conf, query sql server and play a files

I am trying to write a customized app using C that would fetch voice file from SQL Server 2000 using ODBC and FREETDS. Currently I am only able to fetch first 63 KB chunk from the DB, and not able to fetch the rest of the file, below is the code that i am using to do so, fd = open(fullpath, O_RDWR | O_CREAT | O_TRUNC, 0770); if (fd < 0) { ast_log(LOG_WARNING, "Failed to write

Here is the diff from chan_sip.c 15 days ago and 16 days ago. 15 days ago is the point outgoing calls made via grandstream budgetone stopped working. Any help on why it breaks? Any possible fix? /tmp# diff asterisk/channels/chan_sip.c asterisk.works/channels/chan_sip.c 289d288 < int capability; 3921,3922d3919 < p->capability = user->capability;

Hi, today i have taken a strong look at meetme.c what i am trying to accomplish is the following: it should be possible to access an menu from within the conference in order to perform special tasks, eg. to dial another number so that the called person is joined with the conderence. my first try was to use an agi-script for this, but as with agi enabled sip-channels (for which

Hi, I am a beginner in asterisk. Implementing it in my dept in India using TDM400b card with asterisk, zaptel, libpri version latest of CVS HEAD Callerid on my system is coming tough. Asterisk doesnot finishes the callerid spill and Cancells it. After going through code in Callerid.c and chan_zap.c I found that my line is providing caller id of length 8867. Flow enters in zt_call and

I have a Toshiba PBX connected via a QSIG PRI to Asterisk. I can make calls from the Toshiba to Asterisk and internal calls from Asterisk to the Toshiba. What I can't do is make an call with an outside destination from Asterisk to the Toshiba. The Toshiba is looking for 9 to grab an outside line then it expects to see the 10 digits. In the FreePBX dial plan I use 9|. which sends 9 plus the 10

Thanks to those on IRC confirming quickly that this was not something supported (yet) in Asterisk. Below is a quick fix/patch to tcptls.c for Asterisk 18 against this particular provider. Dw static int check_tcptls_cert_name(ASN1_STRING *cert_str, const char *hostname, const char *desc) { unsigned char *str; int ret; ret = ASN1_STRING_to_UTF8(&str, cert_str);

Hi. So, I applied the patch, works, but I could not figure out a fail2ban regex which will hit that line, have you got one I can use? Thanks. On Thu, 30 Aug 2018 11:03:08 -0400, sean darcy wrote: > > On 08/29/2018 09:33 PM, John Covici wrote: > > OK, Thanks. I have a couple of questions -- the line numbers do not > > match exactly, so can you tell me a couple of lines before

Hi, I've looked around the archives and have spent a while on voip-info.org but not found an answer so forgive me if this is in a FAQ somewhere. We've got several Asterisk servers with E1 cards (some Digium, some Sangoma). We provide non geographic numbers for customers and route calls to their existing phone numbers. Calls come in over the PSTN and into Asterisk. This works perfectly

Has anyone had success patching chan_sip.c so that Asterisk will recognize the tel: header for RDNIS information? exten = get_in_brackets(tmp); if (!strncasecmp(exten, "sip:", 4)) { exten += 4; } else if (!strncasecmp(exten, "sips:", 5)) { exten += 5; } else { ast_log(LOG_WARNING, "Huh? Not an

hi, I installed icecast-2.2.0.tar.gz and ices-2.0.1.tar.gz on Fedora3 linux-2.6.12-1.1372_FC3). It works fine for playlist.ogg from the other CPU, such as 'xmms http://192.168.0.3:8000/listplay.ogg'. But when I use 'stdinpcm' like 'asterisk-ices.xml' which send voip's voice udp packets to 'asterisk-ices.xml' such as; .......(snip)...... <stream>