search for: ast_waitstream

Displaying 18 results from an estimated 18 matches for "ast_waitstream".

2004 Apr 08
3
Re: : External access to voicemail
...;o", 1, chan->callerid)) - ecodes = "#0"; + ecodes = "*#0"; /* Play the beginning intro if desired */ if (strlen(prefile)) { if (ast_fileexists(prefile, NULL, NULL) > 0) { if (ast_streamfile(chan, prefile, chan->language) > -1) - res = ast_waitstream(chan, "#0"); + res = ast_waitstream(chan, "*#0"); } else { ast_log(LOG_DEBUG, "%s doesn't exist, doing what we can\n", prefile); res = invent_message(chan, vmu->context, ext, busy, ecodes); @@ -1138,6 +1138,10 @@ silent = 1; res = 0;...
2005 Aug 02
3
priority "a" in macro to access voicemail
I have added the following to a macro that is used for all extensions so a user can access voicemailmain by pressing * during the voicemail prompt ; check voicemail exten => a,1,voicemailmain(${macro_exten}) exten => a,2,hangup The behavior is a little weird, the * key is not recognized during the portion of the greeting where the extension number is being played back, after it is
2003 Oct 07
1
[PATCH] allow announcements in app_dial
...); @@ -670,6 +683,11 @@ ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0); ast_channel_setoption(peer,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0); } + if (announce && announcemsg) + { + res = ast_streamfile(peer,announcemsg,peer->language); + res = ast_waitstream(peer,""); + } res = ast_bridge_call(chan, peer, allowredir_in, allowredir_out, allowdisconnect | clearchannel); if (clearchannel) {
2004 Sep 15
1
Extension based call forwarding using capiECT
...xxx") in new stack Sep 15 19:09:05 NOTICE[245776]: app_capiECT.c:65 capiECT_exec: ECT to 279xxxx:017520xxxxx Sep 15 19:09:17 NOTICE[245776]: app_capiECT.c:74 capiECT_exec: call was answered -- Playing 'digits/0' (language 'de') Sep 15 19:09:17 WARNING[245776]: file.c:902 ast_waitstream: Unexpected control subclass '14' -- Playing 'digits/1' (language 'de') -- Playing 'digits/6' (language 'de') -- Playing 'digits/2' (language 'de') -- Playing 'digits/4' (language 'de') -- Playing '...
2006 Mar 31
3
Echo cancellation problem
...") in new stack -- Executing Playback("CAPI/ISDN3/'xxxxxxxxx'-1", "wsa_benvenuto_lib_uni") in new stack == ISDN3: Answering for 'xxxxxxxxx' -- Playing 'wsa_benvenuto_lib_uni' (language 'it') Mar 31 16:40:21 WARNING[30181]: file.c:1029 ast_waitstream: Unexpected control subclass '14' == ISDN3: Setting up echo canceller (PLCI=0x103, function=1, options=4, tail=64) == ISDN3: Setting up DTMF detector (PLCI=0x103, flag=1) -- ISDN3: Error setting up echo canceller (PLCI=0x103) Mar 31 16:40:21 WARNING[29878]: chan_capi.c:3334 show_capi...
2007 Dec 04
0
Queue App - crash (1.4.15)
...2 #2 0x080827df in ast_waitfor (c=0x820b1a0, ms=1000) at channel.c:2051 #3 0x0809c9a4 in waitstream_core (c=0x820b1a0, breakon=0xb742e930 "", forward=0x8120cad "", rewind=0x8120cad "", skip_ms=0, audiofd=-1, cmdfd=-1, context=0x0) at file.c:1093 #4 0x0809b972 in ast_waitstream (c=0xfffffffc, breakon=0xfffffffc <Address 0xfffffffc out of bounds>) at file.c:1191 #5 0xb742e71f in playback_exec (chan=0x820b1a0, data=0xb7197ad4) at app_playback.c:434 #6 0x080c638d in pbx_extension_helper (c=0x820b1a0, con=0xfffffffc, context=0x820b320 "my-queue", exten=0x820...
2010 Jan 11
1
MeetMe Conferencing - Announce your own join/leave to yourself and other conference members
...ference and announces the caller's name to the rest of the conference with the announce_thread function. Without the chan data available, it makes quick and dirty hacks even impossible without more insight into the structure of the app ( i was thinking of just adding a seperate ast_streamfile / ast_waitstream with the chan variable using an if current->announcetype == CONF_HASJOIN or something like that). Unless I'm missing a way to pass the Asterisk API function ast_pthread_create_background more than one argument and then modify the announce_thread to accommodate it, I'm at a bit of a loss...
2003 Sep 14
6
chan_capi
...lecting to my other ISDN line; as laid out in the Readme) I get REASON=0x3490 as a disconnect indicator for the onHOLD party, trying ECT. capiECT says the call has been answered (I picked up the ringing phone where the call was being transferred to), but then I get a warning "file.c Line 823 (ast_waitstream): Unexpected control subclass '14'". This nice lady is reading the digits of the calling party, then nothing happens until a timeout occurrs. I am not a C programmer but I know how to read some source code ... I'd gladly help where I can! Any suggestions are greatly appreciated....
2004 Jun 01
0
Record Application Problem
Hi everybody, I am having a problem with * Record Application. The thing is I don't want the "beep" before recording, so I removed the instructions: ast_streamfile(chan, "null", chan->language); ast_waitstream(chan, ""); ast_stopstream(chan); Now I am having a strange problem. After I record the sound, the recorded file gets a 3 second of silence before the actual recorded sound. Can anyone solve this?? I can workaround this by playing a silent sound file of about 0.25s before start recordi...
2004 Sep 20
0
Installation problem; collect2: ld returned 1 exit status
..._app_getdata_full': /usr/src/asterisk/app.c:63: undefined reference to `ast_streamfile' app.o(.text+0x2311): In function `ast_play_and_wait': /usr/src/asterisk/app.c:494: undefined reference to `ast_streamfile' app.o(.text+0x2334):/usr/src/asterisk/app.c:497: undefined reference to `ast_waitstream' app.o(.text+0x233f):/usr/src/asterisk/app.c:498: undefined reference to `ast_stopstream' asterisk.o(.text+0xcee): In function `main': /usr/src/asterisk/asterisk.c:1836: undefined reference to `ast_file_init' collect2: ld returned 1 exit status make: *** [asterisk] Error 1 / Stig...
2005 Jul 12
0
meetme an customized menu
...ase '3': /* Invite another Conferee */ menu_active = 0; ast_log(LOG_WARNING,"Taste 3 gedrueckt -> now one more for testing!"); if (!ast_streamfile(chan, "whatNumberToInvite", chan->language)) dtmf = ast_waitstream(chan, AST_DIGIT_ANY); else dtmf = -1; ast_log(LOG_WARNING,"Something pressed ?? :%d \n",dtmf); break; My thougt was, that ast_waitstrem waits a certain time period and then return the dtmf-code. but somehow the code executes without pause between t...
2005 Sep 14
0
oh323 and Asterisk: Calls always hang up
...3/-----@213.30.225.5-0148 answered. -- Playing 'tt-monkeysintro' (language 'en') Call 'ip$213.30.225.5:42873/1893' cleared. -- H.323 call 'ip$213.30.225.5:42873/1893' cleared, reason 24 (Call ended with Q.931 cause) Sep 14 10:30:42 WARNING[14895]: file.c:970 ast_waitstream: Unexpected control subclass '5' Call 'ip$213.30.225.5:42873/1893' with owner has already been cleared (2). Call 'ip$213.30.225.5:42873/1893' has been hungup. -- Hungup 'OH323/-----@213.30.225.5-0148' Call 'ip$213.30.225.5:42873/1893' without owner has a...
2014 Jul 31
0
AGI Record File / what does randomerror mean? res_agi.c / line 2377
...ast_streamfile(chan, "beep", ast_channel_language(chan)); 2370 2371 if ((argc > 7) && (!strchr(argv[7], '='))) 2372 res = ast_streamfile(chan, "beep", ast_channel_language(chan)); 2373 2374 if (!res) 2375 res = ast_waitstream(chan, argv[4]); 2376 if (res) { 2377 ast_agi_send(agi->fd, chan, "200 result=%d (randomerror) endpos=%ld\n", res, sample_offset); 2378 } else { 2379 fs = ast_writefile(argv[2], argv[3], NULL, O_CREAT | O_WRONLY | (sample_offset ? O_APPE...
2005 Jun 20
2
app_valetparking.c
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
2005 Jun 20
1
Re: app_valetparking.c for * STABLE (1.0.X)
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
2008 Mar 12
3
DTMF problems while greeting is playing (Background())
Hi, I have a Digium TE410p T1 card and I've noticed that under asterisk 1.4.17/18 I have problems detecting DTMF in IVRs. I think I've narrowed the problem down to some sort of interference between the greeting that is playing and the DTMF tones. DTMF detection seems to work very reliably when I am in Read() or WaitExten(), but is absolutely unusable while in Background(). I hope someone
2005 Aug 25
2
Custom Application For Asterisk
...text(context); context = NULL; } if (login) { tds_free_login(login); login = NULL; } connected = 0; return 0; } static int play_file(struct ast_channel *chan, char *filename) { int res; ast_stopstream(chan); res = ast_streamfile(chan, filename, chan->language); if (!res) res = ast_waitstream(chan, ""); else res = 0; if (res) { ast_log(LOG_WARNING, "ast_streamfile failed on %s \n", chan->name); res = 0; } ast_stopstream(chan); return res; } int load_module(void) { struct ast_config *cfg; char *s; int res = 0; cfg = ast_load(config); if (!cfg) {...
2011 May 14
10
Asterisk-cpu utilization > 60 %
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