search for: ast_channel

Displaying 20 results from an estimated 61 matches for "ast_channel".

2018 May 28
2
Dial to FastAGI application appears as 1-second CDR - how do I fix?
In my application, I am using AMI to run an Originate command between a channel and a dialplan application (NOT a context). In my case, the application I want to invoke is FastAGI. The Originate AMI command works correctly, but Asterisk generates a very short (0-1s) duration for the CDR that results from this call, regardless of the time spent running the FastAGI application. I want the CDR
2006 Dec 20
0
asterisk run on vxworks for hardware pbx
Hi My hardware PBX run asterisk on vxworks,Because the vxworks not support perl. Now I want to add a callback function to my pbx. now it can store Caller and Called party numbers in queue when Called party is busy Then I malloc a new ast_channel to call.It is should use ast_get_channel_by_exten_locked() or ast_channel_alloc() , my program as follow,But it isn't work, anyone know how to do this. { struct ast_channel *callbk; char *callbk_real_context; char xferto[256],dialstr[265]; c...
2008 Dec 29
3
Manager API
Hi I have a problem with Asterisk-1.6.0.3-rc1 and manager API. I want to dial out from manager's console and with Asterisk 1.4.X this settings were OK. Action: Originate Channel: SIP/384 Context: main Exten: 102 Priority: 1 Callerid: 384 I could dial out, but with asterisk 1.6 I get this error. Response: Error Message: Channel not specified I have originate and system privilege in
2004 Jan 22
3
MGCP w/8x8 DTA-310 and as5300 pstn gateway
...5-1 tansaction 22 Jan 22 22:08:31 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/1@172.16.2.25-1 tansaction 23 -- Resetting interface aaln/1@172.16.2.25 -- Endpoint 'aaln/1@172.16.2.25-1' observed 'hu' -- MGCP handle_request(aaln/1@172.16.2.25-1) ast_channel already destroyed -- MGCP handle_request(aaln/1@172.16.2.25) set vmwi(-) Jan 22 22:08:31 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/1@172.16.2.25-1 tansaction 24 Jan 22 22:08:31 NOTICE[213006]: chan_mgcp.c:396 dump_queue: Removing message fro m aaln/1@172.16.2.25-1...
2004 May 17
4
*8 problem still there?
I upgraded to the latest stable version of 1.0 today and am still seeing the *8 problem where the phone that was originally dialed keeps on ringing even after another phone picks up. Are other people also seeing this? Has somebody figured out how to make this go away? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:
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
2003 Nov 11
4
Registering an application
...gt; #include <stdlib.h> #include <pthread.h> static char *tdesc = "Alex's app"; static char *app = "Alex"; static char *synopsis = "Alex test"; static char *descrip ="Test"; STANDARD_LOCAL_USER; LOCAL_USER_DECL; static int alex_exec(struct ast_channel *chan, void *data) { return 0; } int unload_module(void) { STANDARD_HANGUP_LOCALUSERS; return ast_unregister_application(app); } int load_module(void) { return ast_register_application(app, alex_exec, synopsis, descrip); } char *description(void) { return tdes...
2003 Jul 08
0
Patch to fix some segfaults in Asterisk
...==================== RCS file: /usr/cvsroot/asterisk/channel.c,v retrieving revision 1.25 diff -u -r1.25 channel.c --- channel.c 4 Jul 2003 16:49:11 -0000 1.25 +++ channel.c 8 Jul 2003 10:48:13 -0000 @@ -279,8 +279,8 @@ return NULL; PTHREAD_MUTEX_LOCK(&chlock); tmp = malloc(sizeof(struct ast_channel)); - memset(tmp, 0, sizeof(struct ast_channel)); if (tmp) { + memset(tmp, 0, sizeof(struct ast_channel)); pvt = malloc(sizeof(struct ast_channel_pvt)); if (pvt) { memset(pvt, 0, sizeof(struct ast_channel_pvt)); Index: frame.c ============================================================...
2003 Nov 17
7
Updated iaxComm binaries available for WinXP, Red Hat 9.0
iaxComm is a cross-platform IAX2 softphone available for Win32 and Linux. Win32 and Linux binaries as well as the LGPL source are available at: http://iaxclient.sourceforge.net Recent improvements are a less cluttered user interface, audible ringback and audible outgoing ring, and of course IAX2 protocol support. iaxComm is based upon the wxWindow GUI framework and compiles on Microsoft
2014 Mar 13
1
Any help Address 0xfffffffe out of bounds in app_confbridge.casterisk-11.5.1 using confbridge.conf
...ress 0xfffffffe out of bounds why and how to solve.MyConfbridgeCount(conferencenumber,variablename )return total number of user in conference given by conferencenumber otherwise zero.At runtime using MyConfbridgeCount(4000,count ).now app2: MyConfbridgeCount will call function count_exec(struct ast_channel *chan, const char *data).But at compile time char * data cause core dumped. Asterisk-11.5.1 Centos6 app_confbrige.c confbridge.conf ===================================================== Task: Using Dailplan user want to retrive no of user in conference '6050' => 1. Verbose(3,"tes...
2009 Dec 17
2
Integrate a CPE with Asterisk in MGCP
...2-1 -- Attempting native bridge of MGCP/aaln/1 at 020202020202-1 and SIP/mgcp-out-09d998c0 -- Resetting interface aaln/1 at 020202020202 == Spawn extension (mgcp, 0141909872, 2) exited non-zero on 'MGCP/aaln/1 at 020202020202-1' -- MGCP handle_request(aaln/1 at 020202020202-1) ast_channel already destroyed, resending DLCX. -- MGCP handle_request(aaln/1 at 020202020202) set vmwi(-) - Receive incoming calls through SIP PROXY : OK too, but as with the outgoing calls, the interface seems to self restart : -- Executing Dial("SIP/XXXSIPCALLID", "MGCP/aaln/1 at 02020...
2005 Jan 10
1
"make clean" DO IT!
Just an FYI to all out there that are upgrading after this weekend's run of CVS updates that are in now... MAKE SURE YOU DO "make clean". If you don't and asterisk acts funny this is why. Anytime any struct like ast_channel (which was changed over the weekend) and you don't make clean you'll end up with an asterisk box that acts retarded. So please before reporting a bug do a fresh checkout or make clean and try again. Thanks, Brian
2005 Jun 08
1
Latest CVS and app_rxfax
With the current CVS-HEAD line 88 of app_rxfax.c causes an error. #if (ASTERISK_VERSION_NUM <= 010300) chan->callerid, app_rxfax.c:88: error: 'struct ast_channel' has no member named 'callerid' Commenting out the if else combination of course gives a clean compile. -- Dave Cotton <dcotton@linuxautrement.com>
2006 May 22
1
exten => *0. not possible
...extensions.conf. I changed disconnect => *0 in features.conf to something else. From what I can tell with the little C knowledge I have is that it's caused by a hardcoded *0 value chan_zap.c. Line 5730 of chan_zap.c (svn rev 1077) shows: } else if (!strcmp(exten, "*0")) { struct ast_channel *nbridge = p->subs[SUB_THREEWAY].owner; struct zt_pvt *pbridge = NULL; Can I just change that value to something else like *999999999 or even totally remove the code so I can use exten => _*0. in my dialplan? I'd also appreciate some guidance how to go about removing all the hardcod...
2005 Aug 25
2
Custom Application For Asterisk
...static char abcdcharset[30] = ""; static char abcdlanguage[30] = ""; #define DEFAULTCHARSET "iso_1" #define DEFAULTLANGUAGE "us_english" static int connected = 0; static int mssql_connect(void); static int mssql_disconnect(void); static int play_file(struct ast_channel *chan, char *filename); AST_MUTEX_DEFINE_STATIC(tdslock); static TDSSOCKET *tds; static TDSLOGIN *login; static TDSCONTEXT *context; STANDARD_LOCAL_USER; LOCAL_USER_DECL; struct abcd_user { char moh[80]; char announce[80]; char context[80]; int handled; time_t start; int queuet...
2009 Oct 05
3
Questions about app_jack.c
Hello, My configuration is : Card 0 - kernel dummy sound card Card 1 - my soundcard I have a jackd running in background. My jackd launch command is : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & 1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to
2005 Jan 22
0
chan_capi patch: app_capiFax modifications
...f->Infos[1], stationID); len2 = strlen(headLine); B3conf->Infos[len1 + 1] = (unsigned char)len2; strcpy((char *)&B3conf->Infos[len1 + 2], headLine); B3conf->len = (unsigned char)(2 * sizeof(unsigned short) + len1 + len2 + 2); } static int capi_change_bchan_fax(struct ast_channel *c, char* stationID) { struct ast_capi_pvt *i = c->pvt->pvt; MESSAGE_EXCHANGE_ERROR error; _cmsg CMSG; B3_PROTO_FAXG3 B3conf; SetupB3Config(&B3conf, FAX_SFF_FORMAT, stationID); // Format ignored by eicon cards DISCONNECT_B3_REQ_HEADER(&CMSG, ast_cap...
2007 Jul 12
0
No subject
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"...
2004 Apr 12
2
SwissVoice IP10S not able to dial calls
.../aaln/1@10.1.24.112-1) created in state: Down -- Endpoint 'aaln/1@10.1.24.112-1' observed '7' -- Endpoint 'aaln/1@10.1.24.112-1' observed '9' -- Endpoint 'aaln/1@10.1.24.112-1' observed 'hu' -- MGCP handle_request(aaln/1@10.1.24.112-1) ast_channel already destroyed -- MGCP handle_request(aaln/1@10.1.24.112) set vmwi(-) Here are my configuration files: MGCP.conf =========== [10.1.24.112] context=local host=10.1.24.112 callerid = "Brad Chilton <7726>" callgroup=0,2-5 canreinvite=no pickupgroup=0,1 nat=no threewaycalling=y...