search for: dlpn_dialplan1

...allerid=asreceived group=0 cidstart=polarity context=default channel => 1 group=1 priindication=outofband signalling=fxs_ks channel => 1 Here is my CLI output for a call (not from extension 170) *CLI> -- Starting simple switch on 'DAHDI/1-1' -- Executing [s at DLPN_DialPlan1:1] NoOp("DAHDI/1-1", "all "Mickey Mouse" <170>") in new stack -- Executing [s at DLPN_DialPlan1:2] NoOp("DAHDI/1-1", "ANI 170") in new stack -- Executing [s at DLPN_DialPlan1:3] NoOp("DAHDI/1-1", "NUM 170") in new...

...},1) > I have tried both contexts, [open-bts] and [sip_external] and both don't work If I want to call the mobile phone (6201) with a Twinkle soft phone (6000) I get following message in the CLI-window from Asterisk: > == Using SIP RTP CoS mark 5 > -- Executing [6201 at DLPN_DialPlan1:1] Macro("SIP/6000-00000013", > "stdexten,6201,SIP/6201") in new stack > -- Executing [s at macro-stdexten:1] Set("SIP/6000-00000013", > "__DYNAMIC_FEATURES=") in new stack > * [Jul 12 12:14:29] WARNING[7092]: ast_expr2.fl:468 ast_yyerr...

...<100>" or "abc company <205-555-1212>" I just get "asterisk". Here is what the cdr looks like 2055551212 s default abc company <2055551212> DAHDI-1 SIP/100 Dial SIP/100 104 DLPN_DialPlan1 DAHDI-1 SIP/104 Dial SIP/104 Any clues? TIA Danny Nicholas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090715/98f9f9b6/attachment.htm

Hi all, I would like to know if there is a function/setting for extracting the call setup time in asterisk (1.4 or 1.6). I need this value for every call processed by asterisk as specified in (ETSI TR 101 329-1 v3.1.2): Call set-up time is the time elapsed from the end of the user interface command by the caller (keypad dialling, E-mail alias typing, etc.) to the receipt by the caller of

I'd like to find out how to block everyone outside of the our LAN. The following phone call got through: 1. accountcode: Blank 2. src: Caller*ID number Blank 3. dst: Destination extension 901185294464086 4. dcontext: Destination context DLPN_DialPlan1 5. clid: Caller*ID with text Blank 6. channel: Channel used SIP/xxx-088c48d8 7. dstchannel: Destination channel DAHDI/1-1 8. lastapp: Last application if appropriate Dial 9. lastdata: Last application data (arguments) Dahdi/g1/01185294464086 10. start: Start of call 2010-12-...

...t; 0 ?1-BUSY|1:1-out|1") in new stack -- Goto (macro-trunkdial-failover-0.3,1-BUSY,1) [Jun 6 13:24:29] DEBUG[4667]: app_macro.c:337 _macro_exec: Executed application: Gotoif == Auto fallthrough, channel 'SIP/6006-015d0004' status is 'BUSY' -- Executing [9075763441 at DLPN_DialPlan1:1] Macro("SIP/6006-015d0004", "trunkdial-failover-0.3|Zap/g1/075763441|Zap/g2/075763441|trunk_1|trunk_2") in new stack -- Executing [s at macro-trunkdial-failover-0.3:1] Set("SIP/6006-015d0004", "CALLERID(num)=6498287700") in new stack [Jun 6 13:31:41] D...

Hi, Say I wanted to know what context a SIP registration is using to dial out in my dialplan, what would I do? For example, I have phones on a "local-calls-only" context (as defined in sip.conf), others in "unrestricted-calls". In my dialplan, I`d like to act on that knowledge. Mike -------------- next part -------------- An HTML attachment was

...Mailing List - Non-Commercial Discussion >>> Subject: Re: [asterisk-users] Unable to make outbound calls >>> >>> Sorry. I don't get many opportunities to test this system as its live. Here >>> are the results: >>> >>> -- Executing [9 at DLPN_DialPlan1:1] Dial("SIP/19722-b650fb80", "DAHDI/1") >>> in new stack >>> -- Requested transfer capability: 0x00 - SPEECH >>> -- Called 1 >>> -- Channel 0/1, span 1 got hangup, cause 90 >>> WARNING[6454]: app_dial.c:765 wait_for_answ...

Thank you for the reply. On 8 Dec 2010 at 13:38, Danny (Danny Nicholas <danny at debsinc.com>) commented about RE: [asterisk-users] Configuring Softphone: > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gary Kuznitz > Sent: Wednesday, December 08, 2010 1:27 PM > To: Asterisk