search for: peerflag

...n ast_channel_bridge (c0=0xb659fcd0, c1=0x9455ca0, config=0xb6c4feb0, fo=0xb6c4f988, rc=0xb6c4f98c) at channel.c:3524 #10 0xb78fad29 in ast_bridge_call (chan=0xb659fcd0, peer=0x9455ca0, config=0xb6c4feb0) at res_features.c:1319 #11 0xb7099301 in dial_exec_full (chan=0xb659fcd0, data=0xb6c4feb0, peerflags=0xb6c50568) at app_dial.c:1577 #12 0xb7097dc5 in dial_exec (chan=0xb7ed1900, data=0xb7ed1900) at app_dial.c:1619 #13 0x0808e445 in pbx_extension_helper (c=0xb659fcd0, con=0xb7ed1900, context=0xb659fe20 "op05_x", exten=0xb659ff14 "00116", priority=1, label=0x0, callerid=0x0...

...1) at chanvars.c:71 71 if (var->name[0] == '_') { (gdb) bt #0 ast_var_name (var=0x1) at chanvars.c:71 #1 0x0808934e in pbx_builtin_getvar_helper (chan=0x0, name=0xf5bc2d46 "OUTBOUND_GROUP") at pbx.c:5904 #2 0xf5bbe1e4 in dial_exec_full (chan=0xa281820, data=0x0, peerflags=0xf469fee8) at app_dial.c:964 #3 0xf5bc23ed in dial_exec (chan=0x0, data=0x1) at app_dial.c:1601 #4 0x08090aed in pbx_extension_helper (c=0xa281820, con=0x0, context=0xa281970 "default", exten=0xa281a64 "2348053004990", priority=2, label=0x0, callerid=0xf46a40b0 "ss7...

...(first use in this function) chan_sip.c:10094: warning: type defaults to `int' in declaration of `__s' chan_sip.c:10094: warning: comparison of distinct pointer types lacks a cast chan_sip.c: In function `build_peer': chan_sip.c:10176: parse error before `struct' chan_sip.c:10221: `peerflags' undeclared (first use in this function) chan_sip.c:10221: `mask' undeclared (first use in this function) chan_sip.c:10391: warning: type defaults to `int' in declaration of `__s' chan_sip.c:10391: warning: comparison of distinct pointer types lacks a cast make[1]: *** [chan_sip.o]...

...ere [AgentQ] is called by the queue command to a member added by addqueuemember(Local/99@AgentQ) why don't I get to the NoOp if the agent hangs up during the announcement message (to the agent) ? I see in the app_dial.c program that the "g" flag is tested thus: if ((ast_test_flag(peerflags, OPT_GO_ON)) && (!chan->_softhangup) && (res != AST_PBX_KEEPALIVE)) res = 0; So this would indicate that if all three of these conditions are met then res would be set to 0, and things would behave how I want. In chan_agent.c, the following line is where the agent has hung...

Hi List, I am currently using the H parameter in the dial command. The issue that I am having is that if the user is calling an ivr that requires him to press the * key then the call gets hung up on. How would I go about changing it so that the user will have to press say ** for the H parameter to come in to effect ? Thanks a lot. Dovid -------------- next part -------------- An HTML attachment

...ae9f35cd20, fo=0x7fae9f35bd08, rc=0x7fae9f35bd00) at channel.c:5072 #3 0x00007faec5bbfdb4 in ast_bridge_call (chan=0x7faeac06a320, peer=0x7faea4711570, config=0x7fae9f35cd20) at res_features.c:2281 #4 0x00007faec532d47e in dial_exec_full (chan=0x7faeac06a320, data=<value optimized out>, peerflags=0x7fae9f35d2f0, continue_exec=0x0) at app_dial.c:1930 #5 0x00007faec532f375 in dial_exec (chan=<value optimized out>, data=<value optimized out>) at app_dial.c:1978 #6 0x000000000048b66f in pbx_exec (c=0x7faeac06a320, con=<value optimized out>, context=<value optimized o...

Hi what version of openh323 and pwlib are suggested for asterisk 1.4.21.1.? Thanks to all -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser

...;BUSY") at pbx.c:5825 newvariable = (struct ast_var_t *) 0x10f1d58a0 headp = (struct varshead *) 0xf460880 nametail = 0x2aaabf53cbf7 "DIALSTATUS" __PRETTY_FUNCTION__ = "pbx_builtin_setvar_helper" #2 0x00002aaabf53938e in dial_exec_full (chan=0xf460320, data=0x417a2e50, peerflags=0x417a0c50, continue_exec=0x0) from /usr/lib/asterisk/modules/app_dial.so res = 0 u = (struct ast_module_user *) 0xf0e48d0 rest = 0x0 cur = 0x0 outgoing = (struct dial_localuser *) 0xf13eba0 peer = (struct ast_channel *) 0x0 to = 0 numbusy = 0 numcongestion = 0 numnochan = 0 cause =...

I am using usb-audio for Console/Dsp with asterisk. it is crashing 1.4.21 and also svn. During the brief times its working the audio is choppy but understandable. I have used aplay and arecord at the same time on the same wave file and they work fine every time and I have done it MANY times. Asterisk failes after 1 or 2 times. Any ideas on something I can try? Jerry

Hi, I am using asterisk 1.8.22 and have a problem when calling in parallel several SIP endpoints and I am not sure how to resolve it. In this case Asterisk will not bridge any audio to the caller before the 200 OK. Which means any progress announcements, including remotely generated ringback, are not passed back to the caller. This behavior is completely correct, because there is no way to know

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I just noticed that the Skype API for linux seems to be available. I've read before a number of posts where people were talking about implementing a chan_skype with the skype API. I wonder if there is any progress in that direction, and if anyone is working on it. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to

Hello list, I'm observing what I believe to be inconsistent behaviour regarding "Newstate" AMI events for the "Ringing" state. As such I come to you asking for experience or advice: am I wrong or should I file a bug ? I present you a short introduction which I feel is relevant; however, if you want to go straight to my technical question, please scroll