search for: ast_copy_string

...c -> ast_expr2.o [CC] strcompat.c -> strcompat.o [LD] aelparse.o aelbison.o pbx_ael.o ael_main.o ast_expr2f.o ast_expr2.o strcompat.o -> aelparse aelparse.o(.text+0x3029): In function `ael_yylex': /usr/src/asterisk1/asterisk-1.4.0-beta3/utils/ael.flex:417: undefined reference to `ast_copy_string' ast_expr2f.o(.text+0x1198): In function `ast_expr': /usr/src/asterisk1/asterisk-1.4.0-beta3/utils/ast_expr2.fl:248: undefined reference to `ast_copy_string' collect2: ld returned 1 exit status make[1]: *** [aelparse] Error... plz could answer this issue. -nsthi -------------- next p...

...n, sizeof(connection), "c=IN IP4 %s\r\n", ast_inet_ntoa(dest.sin_addr)); - } - if (add_audio) { capability = p->jointcapability; @@ -24594,9 +24574,6 @@ snprintf(global_sdpowner, sizeof(global_sdpowner), "%s", DEFAULT_SDPOWNER); global_prematuremediafilter = TRUE; ast_copy_string(default_notifymime, DEFAULT_NOTIFYMIME, sizeof(default_notifymime)); - - ast_copy_string(global_rtpip, DEFAULT_RTPIP, sizeof(global_rtpip)); - ast_copy_string(sip_cfg.realm, S_OR(ast_config_AST_SYSTEM_NAME, DEFAULT_REALM), sizeof(sip_cfg.realm)); ast_copy_string(default_callerid, DEFAULT_CALLER...

...ast_md5_hash(resp_hash, resp); + + + /* To a Magicjack domain */ + if (strstr(uri,"talk4free.com")) + { + char callid[256]; + char newnonce[256]; + char *c; + int i; + ast_copy_string(callid, p->callid, sizeof(callid)); + ast_copy_string(newnonce, p->nonce, sizeof(newnonce)); + + strcat(newnonce, "_"); + c = newnonce + strlen(newnonce); + char hex[2]; + hex[1] = 0; + for (i...

...+0xf): undefined reference to `ast_register_file_version' ast_expr2f.o: In function `__unregister_file_version': ast_expr2f.c:(.text+0x1f): undefined reference to `ast_unregister_file_version' ast_expr2f.o: In function `ast_expr': ast_expr2f.c:(.text+0x3e19): undefined reference to `ast_copy_string' Has this been deprecated?

Minor update - use the following: > if (strcasecmp(data, > "x-Asterisk-Request-URI-pseudo-header")==0) > { > ast_copy_string(buf, p->initreq.rlPart2, len); > -----Original Message----- > From: Steve Langstaff > Sent: 23 October 2006 09:58 > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [asterisk-users] SIP_HEADER function; what names > are available? >...

Hi list, My need is to append a site specific parameter to the Contact: header on all INVITEs exiting * via a SIP trunk. I'd like it to look something like this: Contact: <bob:3125551212 at 10.10.10.10;SITE-ID=us.here> where SITE-ID=us.here is set in a config file that * parses on startup. Or in a Dial() command option? Or I don't care exactly how. :-) It is possible to

Hello Armin, hello List I'm trying to get chan_capi working with asterisk from debian stable (asterisk 1.0.7, the debian version number is 1:1.0.7.dfsg.1-2). I managed to get it compiled by providing my own version of ast_copy_string. This is an Austrian PTP line. I can do outgoing calls fine (no comprehensive tests yet). For incoming calls, I'm getting "No answer" on the remote end (GSM-phone) and the last output on the asterisk console (capi debug + set verbose 15) is == ISDN1: Incoming call '0650621XXXX...

In Asterisk 1.2.13 in app/app_voicemail.c, line 4700 ext_pass_cmd is checked to decide whether to use vm_change_password or vm_change_password_shell to change a user's password for his voicemail account. I wonder, what is the difference between vm_change_password and vm_change_password_shell - what is that shell? The only reference I found on the Internet was the following bug report:

...line 832 in app_voicemail.c) is used to retrieve a voice message. What I don't understand however is why ".txt" is appended to the end of the filename. Could someone shed some light on this for me? Thanks, Jez if (msgnum > -1) make_file(fn, sizeof(fn), dir, msgnum); else ast_copy_string(fn, dir, sizeof(fn)); snprintf(full_fn, sizeof(full_fn), "%s.txt", fn); f = fopen(full_fn, "w+"); ____________________________________________________________________________________ Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited

...t); if (!ast_strlen_zero(callbk->macrocontext)) callbk_real_context = callbk->macrocontext; else callbk_real_context = callbk->context; ast_copy_string(xferto,pu->dst,sizeof(xferto)); cid_num = callbk->cid.cid_num; cid_name = callbk->cid.cid_name; if (ast_exists_extension(callbk, callbk_real_context,xferto, 1, cid_num)) { snpr...

...directory.\n"; /* For simplicity, I'm keeping the format compatible with the voicemail config, but i'm open to suggestions for isolating it */ @@ -345,6 +347,20 @@ pos = strsep(&stringp, ","); if (pos) { ast_copy_string(name, pos, sizeof(name)); + if (last == 2) /* Check the first name as well */ + { + conv = convert(pos); + if (conv) { + if (!strcmp(conv, ext)) { +...

...rs: linux:/home/share/chan_sccp-20050705 # make install sh ./create_config.sh "/usr/include" Checking Asterisk version... * no 'struct ast_channel_tech', using old pvt * no 'struct ast_callerid' * no 'AST_CONTROL_HOLD' * no 'ast_config_load' * no 'ast_copy_string' config.h complete. Now compiling .... sccp_actions.c 853 lines sccp_actions.c: In function `sccp_handle_unregister': sccp_actions.c:124: parse error before `*' sccp_actions.c:125: `r1' undeclared (first use in this function) sccp_actions.c:125: (Each undeclared identifier...

...ce number\n "); return -1; } if (!ao2_container_count(conference_bridges)) { ast_verb(3, "No active conferences."); ast_log(LOG_NOTICE, "No active conferences."); return -1; } if (!(localdata = ast_strdupa(data))){ return -1; } AST_STANDARD_APP_ARGS(args, localdata); ast_copy_string(tmp.name, args.confno, sizeof(tmp.name)); conf = ao2_find(conference_bridges, &tmp, OBJ_POINTER); if (conf) { ao2_lock(conf); count = conf->markedusers; ao2_unlock(conf); }else{ count = 0; } if (!ast_strlen_zero(args.varname)) { snprintf(val, sizeof(val), "%d", count); pbx_b...

...^ chan_bridge_media.c:125:39: error: use of undeclared identifier 'pvt' if (!(pvt = ast_unreal_alloc(sizeof(*pvt), ast_unreal_destructor... ^ chan_bridge_media.c:129:18: error: use of undeclared identifier 'pvt' ast_copy_string(pvt->name, data, sizeof(pvt->name)); ^ chan_bridge_media.c:131:15: error: use of undeclared identifier 'pvt' ast_set_flag(pvt, AST_UNREAL_NO_OPTIMIZATION); ^ /home/jeffrey/asterisk-12.4.0/include/asterisk/utils.h:72:15: note:...

...ast_log' res_config_mysql.o(.text+0x1d4d):/usr/local/src/asterisk-addons-1.4.1/res_config_mysql.c:590: undefined reference to `ast_log' res_config_mysql.o(.text+0x1db9): In function `mysql_reconnect': /usr/local/src/asterisk-addons-1.4.1/res_config_mysql.c:603: undefined reference to `ast_copy_string' res_config_mysql.o(.text+0x1dcf):/usr/local/src/asterisk-addons-1.4.1/res_config_mysql.c:605: undefined reference to `ast_copy_string' res_config_mysql.o(.text+0x1dee):/usr/local/src/asterisk-addons-1.4.1/res_config_mysql.c:611: undefined reference to `mysql_init' res_config_mysql.o(...

...ce number\n "); return -1; } if (!ao2_container_count(conference_bridges)) { ast_verb(3, "No active conferences."); ast_log(LOG_NOTICE, "No active conferences."); return -1; } if (!(localdata = ast_strdupa(data))){ return -1; } AST_STANDARD_APP_ARGS(args, localdata); ast_copy_string(tmp.name, args.confno, sizeof(tmp.name)); conf = ao2_find(conference_bridges, &tmp, OBJ_POINTER); if (conf) { ao2_lock(conf); count = conf->markedusers; ao2_unlock(conf); }else{ count = 0; } if (!ast_strlen_zero(args.varname)) { snprintf(val, sizeof(val), "%d", count); pbx_b...

List, Does anyone know how to limit the amount of registrations that a sip user can have? For example, I have 2 softphones that I use on my laptop & desktop, both use the same username & password. If I have both softphones up at the same time, I can make simultaneous calls with each of them. I know you can have call-limit=1 but in this case, I want to allow them to have 3 way calling