search for: ast_queue_fram

...r/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC -c -o chan_oss.o chan_oss.c chan_oss.c: In function `oss_call': chan_oss.c:461: error: too many arguments to function `ast_queue_frame' chan_oss.c:467: error: too many arguments to function `ast_queue_frame' chan_oss.c: In function `oss_new': chan_oss.c:712: warning: assignment from incompatible pointer type chan_oss.c: In function `console_answer': chan_oss.c:809: error: too many arguments to function `ast_queue_...

...r/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC -c -o chan_oss.o chan_oss.c chan_oss.c: In function `oss_call': chan_oss.c:461: error: too many arguments to function `ast_queue_frame' chan_oss.c:467: error: too many arguments to function `ast_queue_frame' chan_oss.c: In function `oss_new': chan_oss.c:712: warning: assignment from incompatible pointer type chan_oss.c: In function `console_answer': chan_oss.c:809: error: too many arguments to function `ast_queue_...

...} memset(&f, 0, sizeof(f)); f.frametype = AST_FRAME_DTMF; f.subclass = resp; f.offset = 0; f.data = NULL; f.datalen = 0; ast_queue_frame(p->owner, &f, 0); } On line 3986, any # or * digit I entered was translated to 0(zero). So any apps depends on # for terminating (voicemail for example) won't work. My question is , why not take just buf[0]? why translate? my UA always send something like d= (one digit...

I am running asterisk 1.4.26.1 and using ALSA not oss dahdi 2.2.0 and libpri-1.4.10 I am calling into console/dsp I hear the audio just fine then after the hangup I hear ringing on the console/dsp. Why would that be? I found this bug for OSS https://issues.asterisk.org/view.php?id=13686 Does the same thing exist in ALSA??? some traces below Jerry == Parsing

....... chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 4980 (do_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 4980 (do_monitor): Got it eventually... chan_sip.c line 948 (sip_hangup): Error obtaining mutex: Device or resource busy channel.c line 370 (ast_queue_frame): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or resource busy chan_sip.c line 5044 (restart_monitor): Error obtaining mutex: Device or re...

...pdate zaptel" channel.c: In function `ast_channel_alloc': channel.c:298: error: `ZT_TIMERPONG' undeclared (first use in this function) channel.c:298: error: (Each undeclared identifier is reported only once channel.c:298: error: for each function it appears in.) channel.c: In function `ast_queue_frame': channel.c:407: error: `ZT_TIMERPING' undeclared (first use in this function) channel.c: In function `ast_read': channel.c:1153: error: `ZT_EVENT_TIMER_EXPIRED' undeclared (first use in this function) channel.c:1155: error: `ZT_EVENT_TIMER_PING' undeclared (first use in this f...

...chan_agent.c: In function `agent_hangup': chan_agent.c:566: warning: implicit declaration of function `ast_say_digit_str' chan_agent.c: In function `agent_new': chan_agent.c:736: warning: assignment from incompatible pointer type chan_agent.c:754: error: too many arguments to function `ast_queue_frame' chan_agent.c: At top level: chan_agent.c:787: warning: function declaration isn't a prototype make[1]: *** [chan_agent.o] Error 1 make[1]: Leaving directory `/usr/src/cvs-src/asterisk/channels' make: *** [subdirs] Error 1 ________________________________________________________...

Hi, if I dial remote user (over CAPI) and transfer it to conference, if it presses * or #, asterisk crashes with following core trace... This works OK with internal extensions .... bt full 0 0x0805dae6 in ast_queue_frame (chan=0x819b738, fin=0x40d99660) at channel.c:384 f = (struct ast_frame *) 0x810d7b8 prev = (struct ast_frame *) 0x0 cur = Variable "cur" is not available. I have : exten => 80,1,MeetMe(|aMsE) exten => _8[1-8],1,MeetMe(${EXTEN}|pMs) I've submitte...

...es to `dlsym' follow loader.o: In function `ast_load_resource': /usr/src/asterisk/loader.c:282: undefined reference to `dlclose' loader.o: In function `ast_update_module_list': /usr/src/asterisk/loader.c:434: undefined reference to `pthread_mutex_trylock' channel.o: In function `ast_queue_frame': /usr/src/asterisk/channel.c:396: undefined reference to `pthread_kill' channel.o: In function `ast_do_masquerade': /usr/src/asterisk/channel.c:2112: undefined reference to `pthread_kill' channel.o: In function `tonepair_generator': /usr/src/asterisk/channel.c:2418: undefined...

Configuration: Linux wpbx 2.4.9-13 #1 Tue Oct 30 20:11:04 EST 2001 i686 unknown P4 2.5 GHz, 1 GB RAM T400P with 3 T1s plugged in. A flow of 46 calls (spread out over 3 T1s). Each call gets transferred (Dial) to the SIP platform and stays for 5 min. Case 1. Asterisk built out of CVS Mar. 19. Test was running for 3 days. Segmentation fault. Case 2. Asterisk built out of CVS Apr. 1. Test was running

What models? Jeff Noxon (jeff-asterisk at planetfall.com) wrote*: > >I just purchased a bunch of Nortel Meridian POTS phones that support >intercom on the 3rd pair. I intend to get it working with Asterisk. >The phones support MWI, have a 3-line display, callerID, call waiting >callerID, 2 lines...very nice. > >On Thu, Feb 27, 2003 at 01:07:19AM -1000, James H. Thompson