search for: pbx_builtin_setvar_helper

...h a "404 Not Found" SIP message. That doesn't seem to be the case, however. Here is a bit of the verbose console output: (Please note that I added some extra ast_log calls to the source code to generate some extra debugging information.) Apr 21 12:35:18 WARNING[16430]: pbx.c:5983 pbx_builtin_setvar_helper: chan=SIP/nyct-901-539f, name=SIPURI, value=sip:nyct-901@192.168.74.33:5060 Apr 21 12:35:18 WARNING[16430]: pbx.c:5983 pbx_builtin_setvar_helper: chan=SIP/nyct-901-539f, name=SIPDOMAIN, value=192.168.74.254 Apr 21 12:35:18 WARNING[16430]: pbx.c:5983 pbx_builtin_setvar_helper: chan=SIP/nyct-901-5...

Hi, I have an Asterisk CVS-HEAD-08/29/05-13:21:43 built on a Redhat 9. I have problems to receive faxes with spandsp-0.0.2pre11 and libtiff-3.5.7-11. I'm trying with a fax machine Panasonic KX-FT25, 14.4Kbps modem speed. I have tested sending a Fax document from Asterisk to the fax machine, it is working fine, but when I try to receive with asterisk, I receive transmission error on the fax

...entation fault. #0 0x000000000044da80 in ast_var_name (var=0x10f1d58a0) at chanvars.c:69 69 if (name[0] == '_') { (gdb) bt full #0 0x000000000044da80 in ast_var_name (var=0x10f1d58a0) at chanvars.c:69 name = 0x10f1d58b0 <Address 0x10f1d58b0 out of bounds> #1 0x000000000049948f in pbx_builtin_setvar_helper (chan=0xf460320, name=0x2aaabf53cbf7 "DIALSTATUS", value=0x417a0690 "BUSY") at pbx.c:5825 newvariable = (struct ast_var_t *) 0x10f1d58a0 headp = (struct varshead *) 0xf460880 nametail = 0x2aaabf53cbf7 "DIALSTATUS" __PRETTY_FUNCTION__ = "pbx_builtin_setvar_...

...???????????????????????????ast_verbose(VERBOSE_ PREFIX_3 "Redirecting %s to fax extension\n", ast->name); ????????????????????????????????????????????????/* Save the DID/DNIS when we transfer the fax call to a "fax" extension */ ????????????????????????????????????????????????pbx_builtin_setvar_helper(as t,"FAXEXTEN",ast->exten); ????????????????????????????????????????????????if (ast_async_goto(ast, ast->context, "fax", 1, 0)) ????????????????????????????????????????????????????????ast_log(LOG_WARNING, "Failed to async goto '%s' into fax of '%s'\n&...

...do this. What am I missing? S ----------------- Changes to res_parking.c: ... /* Use the non-macro context to transfer the call */ if(strlen(transferer->macrocontext)) transferer_real_context=transferer->macrocontext; else transferer_real_context=transferer->context; + pbx_builtin_setvar_helper(transferee,"TRANSFERER",transferer->name); "new" dialplan: exten => 7,1,Wait,1 exten => 7,2,Answer exten => 7,3,GotoIf($["${TRANSFERER}" = ""]?10) exten => 7,4,agi(synth|You are being transferred to the other parties personal conference room...

...l me. Asterisk 1.4.13 Zaptel 1.4.5.1 Libpri 1.4.1 Addons 1.4.4 #0 0x000000000044da80 in ast_var_name (var=0x2aabcc04bf20) at chanvars.c:69 69 if (name[0] == '_') { (gdb) bt #0 0x000000000044da80 in ast_var_name (var=0x2aabcc04bf20) at chanvars.c:69 #1 0x000000000049948f in pbx_builtin_setvar_helper (chan=0x2aaac801a890, name=0x2aaab69395a8 "RTPAUDIOQOS", value=0x2aaac80ecf20 "ssrc=1967815032;themssrc=917073588;lp=61288;rxjitter=0.000165;rxcount=3668;txjitter=0.005142;txcount=1515;rlp=0;rtt=3.924000") at pbx.c:5825 #2 0x00002aaab6925a94 in handle_request_bye (p...

...tring(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_builtin_setvar_helper(chan, args.varname, val); } else { if ( ast_channel_state(chan)!= AST_STATE_UP) { ast_answer(chan); } res = ast_say_number(chan, count, "",ast_channel_language(chan),(char *) NULL); } return res; } static int load_module(void) { ast_verb(3 ,"==Inside load_module=="); res |=...

I was wondering if it is possible (in 1.2.x) to get the SIP response code back after doing Dial(). Dial() seems to treat most call-setup problems as dialstatus CONGESTION, and some are NOANSWER, but I want to know the SIP response code, so I could return the right tones to the user, not just a congestion tone for every fault. Anyone know a way to find out that information, so I want the

...es. For the inital test I'm simply trying to connect to the * demo menu. The drivers compile (with a few warning that I believe aren't important - see attachments). chan_zap comiles with the warning: chan_zap.c: In function `pri_dchannel': chan_zap.c:6344: warning: passing arg 1 of `pbx_builtin_setvar_helper' from incompatible pointer type The qozap driver appears to load correctly and I get this in the log : Apr 1 17:51:07 debian kernel: Zapata Telephony Interface Registered on major 196 Apr 1 17:51:07 debian kernel: qozap: start Apr 1 17:51:07 debian kernel: PCI: Enabling device 00:0b.0 (0...

...so.6 Thread 2 (process 21752): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002f6a9e in __lll_mutex_lock_wait () from /lib/tls/libc.so.6 #2 0x0028800b in _L_mutex_lock_3800 () from /lib/tls/libc.so.6 #3 0x0028bb61 in strcasecmp () from /lib/tls/libc.so.6 #4 0x08090667 in pbx_builtin_setvar_helper (chan=0xb01657f0, name=0xb6a41858 "connid", value=0xb6a41760 "5") at pbx.c:6030 #5 0x00460212 in ?? () from /usr/lib/asterisk/modules/app_addon_sql_mysql.so #6 0xb01657f0 in ?? () #7 0xb6a41858 in ?? () #8 0xb6a41760 in ?? () ---Type <return> to continue, or q <retu...

...tring(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_builtin_setvar_helper(chan, args.varname, val); } else { if ( ast_channel_state(chan)!= AST_STATE_UP) { ast_answer(chan); } res = ast_say_number(chan, count, "",ast_channel_language(chan),(char *) NULL); } return res; } static int load_module(void) { ast_verb(3 ,"==Inside load_module=="); res |=...

...y of APP_realtime somehow, so I started digging around in the func_realtime source - here's what I came up with: For 1.6.0, look at line 86 of func_realtime.c ast_str_append(&out, 0, "%s%s%s%s", var->name, args.delim2, var->value, args .delim1); I simply changed this to: pbx_builtin_setvar_helper(chan, var->name, var->value); Now when I call the realtime function, I get the channel variables populated instead of having to parse that godawful string to get at my data. Again, yell at me if you will, but even Mark Spencer commented on how func_realtime wasn't all that great, but no...

Is it possible for you to explain in more detail the situation involved. I'm still thinking that what you're trying to achieve can be done at least with the help of DUNDi weights, but I still don't think I have a full grasp of the solution you're crafting. Regards, - Brad -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Hi all, Running 1.2.14 When I call a FastAGI script such as this script for an incoming call: [calldirect] exten=>s,1,Answer() exten=>s,2,AGI(agi://192.168.1.175/calldirect?check&${CALLERID(num)}) exten=>s,3,Goto(check_time,s,1) and the FastAGI server is not running (Asterisk gets "connection refused" TCP error), Asterisk just terminates the call like so: May 17

Hi, I'm using the following setup : Alice ---- IPPhone ------<LAN>----- Media gateway ----<PSTN> ------- Phone ---- Bob For certain calls, users complains about echo : they can ear their own voice in their handset, though media gateway echo cancel is turned on. I'm wondering how this echo cancelation engine is supposed to work. My understanding of echo is that most probably,