search for: incoming_call

I am posting here since you guys are my last hope. I am trying to configure a Polycom Soundpoint IP 335 with MWI. Is there any way to eliminate the scrolling messages and Msgs softkey? I am trying to get it where it's just the light that indicates the new messages. I don't know if Asterisk has to send a different notification or what have you. Thanks, --Eric -------------- next

...I) Card 0 Span 2 UNCONFI 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) *CLI> dahdi show channels Chan Extension Context Language MOH Interpret Blocked State pseudo default default In Service 1 incoming_calls en default In Service 2 incoming_calls en default In Service 3 incoming_calls en default In Service 4 incoming_calls en default In Service 5 incoming_calls en default In Service 6...

Hi, When a VOIP user dials an external number, the calls are routed through our SIP provider. Is there a simple way to check whether the DDI exists locally before dialling out to the sip provider? Something like GotoIfExists(5551234 at incoming_calls) Currently, I'm paying for all calls, regardless of whether they exist locally. All DDIs exist in the incoming_calls context. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101025/44412...

...rrently trying to configure a PBX make use of a multiple of outgoing lines, currently my extensions.conf looks something like below >> ; extensions.conf ; 20th October 2008 [globals] sip1=201 sip2=202 sip3=203 sip4=204 [general] autofallthrough=yes [default] [incoming_calls] exten => _89859715,1,Dial(SIP/201) exten => _89859716,1,Dial(SIP/202) [macro-sipmail] exten => s,1,Verbose(1,Extension ${ARG1}) ;line req to pick up ext if it's not reg. exten => s,n,Dial(SIP/${ARG1},30) exten => s,n,GotoIf($["${DIALSTATUS}" = "BUSY&...

...nf [general] canreinvite=yes [1001] username=1001 password=1001 type=friend context=phones host=dynamic [1002] callerid=1002 username=1002 password=1002 type=friend context=phones host=dynamic [root at localhost asterisk]# cat extensions.conf [globals] [general] autofallthrough=yes [default] [incoming_calls] [phones] exten => _1XXX,1,NoOp() exten => _1XXX,n,Dial(SIP/${EXTEN},30) exten => _1XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail) exten => _1XXX,n,Hangup() PS: My sip server and softphones are in the same network subnet. There are not any firewall or iptables rule...

...qualify=no port=5060 nat=never dtmfmode=rfc2833 context=internal canreinvite=no insecure=very callerid=XXXXXXXXXX <xxxxxx> [Digital_out] type=peer secret=xxxxxx username=XXXXXXXXXX host=plasma.digitalvoice.ca fromuser=XXXXXXXXXX fromdomain=plasma.digitalvoice.ca insecure=very context = incoming_calls qualify=yes nat=yes EXTENSIONS.CONF [default] exten => s,1,Answer( ) exten => s,2,Playback(demo-echotest) exten => s,3,Hangup( ) [internal] exten => _NXXNXXXXXX,1,dial(SIP/${EXTEN}@Digital_out,30) exten => _1NXXNXXXXXX,1,dial(SIP/${EXTEN}@plasma.digitalvoice.ca,30) exten =>...

...940) to any external number using my PSTN line, but when I call my PSTN line from my cell phone, the Cisco doesn't ring (and no message appears in the Asterisk CLI). Here are my config files: zaptel.conf fxsks=1 loadzone = be defaultzone = be zapata.conf [channels] context=incoming_calls usecallerid=yes hidecallerid=no immediate=no signalling=fxs_ks callerid = test <123> echocancel=yes group=1 channel=>1 extensions.conf [globals] [general] autofallthrough=yes [default] exten => s,1,Verbose(1|Unrouted call handler) exten => s,n,Answer() ext...

...14413] rtp.c: Channel '<unspecified>' has no RTP, not doing anything [Mar 16 15:11:13] DEBUG[14413] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Mar 16 15:11:13] DEBUG[13304] chan_sip.c: Checking device state for peer 241-081d7a50 [Mar 16 15:11:13] DEBUG[14413] pbx.c: Spawn extension (incoming_calls,7977529,2) exited non-zero on 'SIP/7977529-081d60d0' [Mar 16 15:11:13] DEBUG[14413] channel.c: Soft-Hanging up channel 'SIP/7977529-081d60d0' [Mar 16 15:11:13] DEBUG[14413] channel.c: Hanging up channel 'SIP/7977529-081d60d0' [Mar 16 15:11:13] DEBUG[14413] chan_sip.c: Hangu...

Hi, I have a server with 2 TE110P cards. 1 card is plugged to telco line, another card is plugged with a Hicom PBX. I want to send some call to VoIP phones and all other to my PBX. I don't known how to make my dialplan : ===========Extensions.conf========== [incoming_call] exten => 090200000,1,Goto(callcenter,100,1) exten => 022956353,1,Goto(callcenter,100,1) exten => 022956388,1,Goto(callcenter,100,1) exten => 022956355,1,Goto(callcenter,101,1) exten => s,1,Goto(go_to_pbx,200,1) [callcenter] exten => 100,1,Answer exten => 100,2,SetMusicOnHold(...

...02 81 82] > Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) > Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] -- Executing AGI("Zap/14-1", "incoming_call.pl") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/incoming_call.pl < Protocol Discriminator: Q.931 (8) len=9 < Call Ref: len= 2 (reference 15996/0x3E7C) (Originator) < Message type: RELEASE (77) < [08 02 83 86] < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU)...

...muser=703XXXYYYY secret=securepassword username=703XXXYYYY insecure=very ;insecure=port,invite context=incoming authname=703XXXYYYY dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no --------extensions.conf-------- [globals] [general] autofallthrough=yes [incoming_calls] exten => 1703XXXYYYY,1,Dial(SIP/5000) [internal-phones] include => outgoing exten => 5000,1,Dial(SIP/5000,20) exten => 5002,1,Dial(SIP/5002,20) [outgoing] exten => _X.,1,NoOp() exten => _X.,n,Dial(SIP/enter_broadvoice/${EXTEN}) --------SIP Registry------------------ -*CLI...

...----------------- My extentions.conf: -----cut here----------------------------------- [globals] [general] [default] exten => s,1,Verbose(1|Unrouted call handler) exten => s,n,Answer() exten => s,n,Wait(1) exten => s,n,Playback(tt-weasels) exten => s,n,Hangup() [outgoing_calls] [incoming_calls] [internal] exten => 500,1,Verbose(1|Echo test application) exten => 500,n,Echo() exten => 500,n,Hangup() exten => 501,1,Verbose(1|Playback test application) exten => 501,n,Playback(vm-review) exten => 501,n,Wait(1) exten => 501,n,Hangup() [phones] include => internal --...

...s, I have a playback function there. But CLI reports: CLI [Jun 18 14:20:22] WARNING[2287]: pbx.c:9542 ast_context_verify_includes: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*' Extensions.conf [general] autofallthrough=yes [default] [incoming_calls] [phones] include => internal include => hovedmeny [internal] include => to_SIPtrunk include => nighttime exten => _10X,1,NoOp() exten => _10X,n,Dial(SIP/${EXTEN},10) exten => _10X,n,Playback(kuntiltestt_) ;exten => _10X,n,Playback(vm-nobodyavail&tt-monkeysintro&amp...

Hi, I have the problem that WaitExten is not responding to key presses. Here are the sections from my extensions.conf: [globals] incoming_call=0 menu=0 announce=0 [internal] exten => 777,1,Goto(hotline,${EXTEN},1) [hotline] exten => _X.,1,Set(CALLERID(name)=Hotline) exten => _X.,n,Set(original_extension=${EXTEN}) exten => _X.,n,GotoIf($[${announce}=1]?4:10) exten => _X.,n,Answer exten => _X.,n,NoOp(Ansage: Das Problem...

...-- Executing [92952210 at internal:2] Congestion("SIP/222-09ab3588", "") in new stack == Spawn extension (internal, 92952210, 2) exited non-zero on 'SIP/222-09ab3588' localhost*CLI> --------------sip.conf --------- [general] bindaddr=0.0.0.0 [Cisco1760] context=incoming_calls type=friend host=172.17.2.1 dtmfmode=rfc2833 disallow=all allow=ulaw insecure=very ----------extensions.conf------------ [globals] OUTBOUNDTRUNK=SIP/Cisco1760 [outbound-local] exten => _9NXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) exten => _9NXXXXXX,n,Congestion() exten => _9NXXXXXX,n...

I don't use digitalvoice, but based on a similar provider you may need to have your username inserted in your extensions.conf context.... [incoming_calls] exten => username,1,Answer( ) exten => username,2,Playback(demo-echotest) exten => username,3,Hangup( ) Just an idea....

Hi, I've been reading all I can on Google (and Asterisk TFOT book) looking for ideas on how to implement an automated blacklist feature. I would like to automatically blacklist a incoming number based on timestamp and count information. For example, if I get a prank call from the same number 5 times within 15 minutes, I want my dialplan to automatically blacklist this number. Should I be

...012 664 2300 Cellphone : 079 522 6519 Fax : 012 644 2902 E-mail : ian at vddi.co.za Skype : vddb_igcoetzee */etc/asterisk/zapata.conf* ; Span 1: WCTDM/0 "Wildcard TDM800P Board 1" (MASTER) ;;; line="1 WCTDM/0/0" ;Cellphone signalling=fxs_ks callerid=asreceived context=incoming_calls callerid= group=2 busydetect=yes usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes pulsedial=no callprogress=yes busycount=5 toneduration=500 subscribecontext=GXP_BLF overlapdial=no channel => 1 ;;; line="2 WCTDM/0/1" ;La...