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Hi Richard, Thank you for your response. But after I remove the parameters of dial command (tTkK). The call was still not native bridge. Let me know if you have any suggestion. Best regards, Charles 2015-01-30 0:34 GMT+08:00 Richard Mudgett <rmudgett at digium.com>: > > > On Wed, Jan 28, 2015 at 8:27 PM, Charles Wang <lazy.charles at gmail.com> > wrote: > >>...

...11.14.2 and DAHDI 2.8.0. I try to set callwaiting = no AND callwaitingcallerid = no in chan_dahdi.conf. But I can't find native bridging information from CLI(opened debug mode in logger.conf). How can I test the dahdi_bridge in native bridge mode? I use normal dial command ex: Dial(DAHDI/2,30,tTkK) to dial from FXS1 to FXS2. Does anyone kind to help me solve it? -- Best Regards Charles -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150129/79a619e0/attachment.html>

...489", "__DYNAMIC_FEATURES=") in new stack -- Executing [s at macro-stdexten:2] GotoIf("IAX2/6223-10489", "0?5:3") in new stack -- Goto (macro-stdexten,s,3) -- Executing [s at macro-stdexten:3] Dial("IAX2/6223-10489", "SIP/ 6123&IAX2/6123|20|tTkK") in new stack etc... My question is: Who called macro-stdexten ? I grep'ed for it in /etc/ asterisk/* and could only find the definition of the macro in extensions.conf. I'm curious to know if this macro is something hard coded in asterisk for dialing extensions with voicemail e...

...; I try to set callwaiting = no AND callwaitingcallerid = no in > chan_dahdi.conf. > But I can't find native bridging information from CLI(opened debug mode in > logger.conf). How can I test the dahdi_bridge in native bridge mode? > > I use normal dial command ex: Dial(DAHDI/2,30,tTkK) to dial from FXS1 to > FXS2. > > Does anyone kind to help me solve it? > Native bridging cannot happen if Asterisk has an interest in the audio stream. Remove the tTkK flags in the Dial command. Richard -------------- next part -------------- An HTML attachment was scrubbed... URL: &...

Hi all. We have a problem with correct dialstatus and cdr(disposition) when using call-limit. When call-limit reached dialstatus is CHANUNAVAIL and CDR(disposition)='NO ANSWER' -- Executing [0014 at sub_pbxdialco:49] Dial("SIP/1295-000001f8", "SIP/0014,12,tTkK") in new stack == Using SIP RTP CoS mark 5 [2013-07-03 15:22:27] NOTICE[29728]: chan_sip.c:6003 update_call_counter: Call to peer '0014' rejected due to usage limit of 1 -- Couldn't call 0014 == Everyone is busy/congested at this time (0:0/0/0) -- Executing [0014 a...

Hi, I am facing the following issues with call parking: 1) Asterisk does not announce the parking extension:- I press # and hear "transfer" followed by a beep. I dial 700 but do not hear the parking extension number. 2) After parking time, the original extension rings and i accept the call. When I again try to park, the call park sequence fails. I have attached features.conf and

Hello Doug, Friday, December 13, 2019, 11:03:37 AM, you wrote: >> This is exactly what I do - “press 1 for a human” >> Works great > I do this as well, but I also do a database lookup to see if the number > is on our speeddial list and if so, pass the call directly on without > the IVR prompts. I do something similar for calls without caller ID, but I was still getting