search for: ast_set_flag

...^ 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: expanded from macro 'ast_set_flag' typeof ((p)->flags) __p = (p)->flags; \...

Hi All I'm using the PickUp application from Bristuff to allow me to pick up channel groups across Zap and Sip. The only snag is that having picked up a call with it, you can't then transfer it on. Taking a dive into app_dial, it looks like when you specify the T option, it does: ast_set_flag(&(config.features_caller), AST_FEATURE_REDIRECT); Since there's nothing like that in app_pickup, I guess that's why you can't do the transfer (nothing has enabled the flag) Is there any easy way to allow transfers on calls picked up using PickUp? Failing that, is there a way to...

I released app_changrab.c lastnight really late... It includes a way to hijack a channel and originate calls from the CLI. /b --- Keep Your Friends Close, But Your Enemies Even Closer...

...ely won't use Asterisk below major version 13 any more. Just use 'directmedia'. They are the same setting (snippet from chan_sip's configuration parsing): } else if (!strcasecmp(v->name, "directmedia") || !strcasecmp(v->name, "canreinvite")) { ast_set_flag(&mask[0], SIP_REINVITE); ... Note that these settings and their behaviour is the same from 1.8 through 13. While I'm glad to see anyone using the latest and greatest - yay Asterisk 13! - this isn't a reason to go to Asterisk 13. Matt -- Matthew Jordan Digium, Inc. | Engineer...

Hi, I am using asterisk 1.4. I have confgured the musiconhold.conf file. However, when i make a call and then hold the call it does nothing. in the CLI i do not see the starting/stopping musiconhold messages. i am making calls from sip to h323 using asterisk assip/h323 gateway (with gnugk and ooh323). i get the following messages when putting the call on hold: -- Executing [204 at default:1]

Hi, with canreinvite=no and directmedia=no I and getting the message in the logs for all calls "switching from simple_bridge technology to native_rtp" -- Executing [102 at mkg:1] Dial("SIP/101-00000017", "SIP/102") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/102 -- SIP/102-00000018 is ringing -- SIP/102-00000018 answered SIP/101-00000017

Hello, at first, thanks for helping! In the meantime, I have done a lot of research and trial and error, and I could solve that specific problem. Obviously, the dialplan application "Answer" was playing a key role here. My original dialplan snippet (which produced that problem) was: exten => _00., 1, NoOp() same => n, Set(FAXOPT(gateway)=yes) same => n,