similar to: How to use an Application from inside an Application?

I'm trying to use a simple page function. It starts a MeetMe conference with the devices I've listed, but the devices hang up after 3-5 seconds. After doing some research I found this was a problem, and I needed to remove a (5) from app_page.c Well, my app_page.c didn't have the (5). I did make clean; make install again just in case I had some weird compiled version installed that

Dan et al; This looks like a perfect solution. However, I have one issue. If I initiate the macro manually (put it in the proper context/dialplan) it works. I see the *.wav file being created and growing in the /var/spool/asterisk/monitor directory. If I try to implement it adding the MixMonApp => *1,self/both,Macro,mixmon to the [applicationmap] in features.conf, I cannot get it to

Since www.bkw.org seems not to exist anymore (getting response from some hosting provider), does anyone happend to have a copy of app_valetparking.c from www.bkw.org - the one that should work with * stable 1.0.X ? If so please contact me. One that can be downloaded from www.loligo.com dosn't compile with 1.0.X, and SuperValletParking (www.asterlink.com/svp/) seems to be for * HEAD

Nope ! This is the one that tries to include PRE 1.0.X header file <parking.h>. It cannot compile on * 1.0.X (I have tried also to include <features.h> instead of <parking.h> (as far as I know features.h is successor to parking.h), but still without results). Thanks anyway. Nenad > > Try this > >> Since www.bkw.org seems not to exist anymore (getting

Hi asterisk-users, We are using Asterisk 1.2.12.1, and are trying to use the Page application. It seems to work but after approx 4-5 seconds the call is hung up. The dialplan code look like this: exten => _*2XX,1,AGI(get-paging-devices.agi,${EXTEN:2}) exten => _*2XX,n,GotoIf($[ "${PAGING_DEVICES}" = "invalid" ]?i,1) exten => _*2XX,n,SIPAddHeader(Call-Info:

2023-11-08 18:14:13] ERROR[571246][C-000017e2] : Got 19 backtrace records # 0: [0x5bd18a] asterisk utils.c:2800 __ast_assert_failed() # 1: [0x4618e3] asterisk astobj2.c:589 __ao2_ref() # 2: [0x58e660] asterisk stasis_cache.c:824 update_create() # 3: [0x58efed] asterisk stasis_cache.c:903 caching_topic_exec() # 4: [0x586b90] asterisk stasis.c:1380 dispatch_message() # 5: [inlined] asterisk

Hello, my configuration is the following: a TDM400P board with an fxs and fxo daughter boards on it. I thus connect a fax to my FXS port, after having verified that this port was correctly functioning. For this, I had tried before with a simple phone, and with some basic voicemail exten scripts. Here is my simple dialplan for my fax reception: exten => 300,1,Ringing() exten =>

Is anyone out there using app_dbodbc (http://www.bkw.org/~brian/app_dbodbc.c)? Any problems with it? I was able to get it all working, but it causes * to segfault every now and then. It does not appear to be related to any specific function (ODBCget,ODBCput,ODBCdel,ODBCdelltree). It is 100% repeatable. If I noload the module, everything works fine, but when its running, after calls to any of the

Like some others on the list spandsp is segfaulting asterisk when recieving a fax. I'm on debian testing/unstable with freshly checked out asterisk CVS and sandsp. My libtiff version is 3.6.1. Here is the GDB output --- snip ----- Changed from phase 5 to 4 Start rx document - compression 2 Start rx page >>> CFR: 84 HDLC underflow in state 5 Post trainability Changed from phase

Hello, I've recently setup a small load test against an instance of Asterisks. I've tested on asterisk 13.5 and 14.6 with the same results. I am using PJSIP. My dial plan is, [test] exten => 1001,1,Answer exten => 1001,n,MusicOnHold(15) exten => 1001,n,Hangup I am using SIPP to test. I can share XML if desired but it simply waits on the line while music plays for 8

Hi, On 64 bit centos 5.6 I have virtualbox 4 and 64 bit elastix latest. Since yesterday cpu utilization has been constantly peaking 65-75%. Hardly 1-2 concurrent calls. No other activity on server. Top shows asterisk on top. Its quad xeon server with 4 gb ram. Any suggestion where should I start and how should I go about with my investigation. Thank you and have a great weekend. Sans

I applied the florz patch but my problems remain. Now I get all sorts of weird errors on the console and I cannot make outgoing calls. Incoming calls work as expected. I am using a single HFC-S card with BRI. Any clue what these errors below are? Ri = 44651 TEI msg = 3 TEI = 7f Ri = 3800 TEI msg = 3 TEI = 7f Ri = 42399 TEI msg = 3 TEI = 7f Ri = 42409 TEI msg = 3 TEI = 7f Ri = 22078 TEI msg = 3

Hi Richard, Thank you for the reply Correct, I did mean 13.15. I set no optimize and better backtrace through "make menuselect" and the output is now [Aug 28 21:41:16] ERROR[17171][C-0000392d]: frame.c:343 ast_frdup: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x21962b0 (0) Got 26 backtrace records #0: [0x61923f] main/utils.c:2475

Hi, I'm running the latest CVS HEAD version of asterisk, and I'm experiencing hangups during voice conversation. This happens quite regularely and often. The problem is in dsp.c, line 1235, where it says accum /= len; But `len', at this point, is 0, resulting in a SIGFPE. The routine ast_frame *i4l_read() in channels/chan_modem_i4l.c:411 is setting p->fr.datalen to

In the past, there have been reports of problems with Asterisk with multiple processors and/or HyperThreading. I'm having a !@#$ of a problem with an HPDL380 with 2 3.4gHz Xeon processors, 2 gb RAM -- if I got 24 hours I'd think I had died and gone to heaven :) Am I missing something obvious like "Asterisk is single CPU, single core?" I can't access the ILO so I

Hi, I was messing around with FireFly last night and got asterisk to crash hard. It looks like the bug is a division by zero in chan_iax2.c. I reproduced it and here are some infos I got from gdb: [Switching to Thread 245775 (LWP 23251)] 0x41154918 in calc_timestamp (p=0x816b710, ts=0, f=0x424eef24) at chan_iax2.c:2896 2896 int diff = ms % (f->samples / 8);

Hello Everyone; I am looking for a solution to record calls that come into our Asterisk server. I am hoping for something that is easy to use - however, if I have to modify it to make it easier to use, I do not mind. Does anyone know of any opensource or otherwise solutions out there that I can try out? Thanks much. Glen

Hello, all. With the assistance of very helpful folks, our brand new multi-tenant setup seems to be working smoothly from start to finish with just a bump or two. The biggest is parking. Now that we got most kinks worked out, I'm a little more comfortable in trying to resolve this. There seem to be two problems: 1. Parking assigns parking spaces from the default group no matter

Hello all. I'm new to asterisk and have been using and testing it for about a week now. My initial hope has been to use it as a sip<->h323 gateway to tie SIP & H323 based ip phones together with my Cisco AS5300 and Lucent MaxTNT/MVAM networks. I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM. I have tried a couple CVS version from the past

Hi everyone: I'm trying to compile Asterisk on FreeBSD 6.0-RELEASE and I'm getting the following error: cc -O2 -fno-strict-aliasing -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -DMAKE_VALGRIND_HAPPY -I/usr/local/include -L/usr/local/lib -I/usr/local/include/spandsp -DZAPTEL_OPTIMIZATIONS