search for: 101,2

Morning, Have some questions regarding receiving and sending faxes... 1:st example: exten => 101,1,Answer() exten => 101,2,Wait(3) exten => 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) exten => 101,4,System(tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff > /var/spool/asterisk/tmp/fax.pdf) exten => 101,5,System(mutt -s 'New FAX for you sir' -a /var/spool/asterisk/tm...

...loadzone = us defaultzone=us loaded modules zaptel and wcfxs /etc/askterisk/zapata.conf contains [channels] language = en signalling = fxo_ks context = phones channel => 1-4 /etc/askterisk/extensions.conf contains [general] static=yes writeprotect=yes [phones] exten => 101,1,Ringing() exten => 101,2,Dial(Zap/1,10) exten => 101,3,Congestion I also uncommented the "noload => chan_oss.so" in /etc/asterisk/modules.conf because I don't have a sound card. Other than that, all conf files are the originals from "make samples". But...

Hi all, I have Asterisk running on FreeBSD 4.x and I have made configurations to sip.conf, extensions.conf and voicemail.conf. I have also setup SIP phones on two different PCs. My problem is that when one of the SIP phones logins in, the other won't. My sip.conf has: [101] type=friend host=dynamic username=101 secret=test dtmfmode=rfc2833 context=from-sip mailbox=201 callerid="101" <2125> nat=yes My extensions.conf has: exten => 101,1,Dial(SIP/101,20,tr) exten => 101,2,VoiceMail,u101 exten => 101,102,VoiceMail,b101 My voicemail.conf has: 1...

Hi all, I have Asterisk running on FreeBSD 4.x and I have made configurations to sip.conf, extensions.conf and voicemail.conf. I have also setup SIP phones on two different PCs. My problem is that when one of the SIP phones logins in, the other won't. My sip.conf has: [101] type=friend host=dynamic username=101 secret=test dtmfmode=rfc2833 context=from-sip mailbox=201 callerid="101" <2125> nat=yes My extensions.conf has: exten => 101,1,Dial(SIP/101,20,tr) exten => 101,2,VoiceMail,u101 exten => 101,102,VoiceMail,b101 My voicemail.conf has: 1...

Anyone can share me with Music Onhold Configuration sample? Thanks in advance for your help, Kang

...no luck. Any help is apprecaited. Sean [800-in] exten => s,1,Answer exten => s,2,Background(billing-welcome) exten => s,3,ResponseTimeout(5) exten => s,4,Background(billing-menu) exten => t,1,Goto(s,3) exten => i,1,Playback(pbx-invalid) exten => i,2,Goto(s,2) exten => 101,1,Ringing exten => 101,2,Wait(1) exten => 101,3,Macro(ext,101) exten => 113,1,Ringing exten => 113,2,Wait(1) exten => 113,3,Macro(ext,113)

I searched the 22490 messages I have in my own personal asterisk-users archive and have not found the answer, and it also does not appear on the wiki. I have a SIP phone and a regular phone on a TDM400P FXS interface. Extensions are 100 and 101, respectively. On the SIP phone I can hit #, get the "Transfer" prompt and enter an extension I want to transfer to. No problem. I can do the same thing on the FXS port. My question is does anyone have a dialplan that will bring the call I transfered back to me if the transfer fails...

Hi everybody In advance sorry for my bad english and if my problem was already exposed (I didn't find any tips in the mailing list archive. Bad luck) I have some questions about asterisk 1.6 release : 1) how can I do a n+101 priority jumping if a SIP canal is busy ? I read that the general parameter "priorityjumping" is depreciated in the 1.6 release and I already try the "j" option in dial() application but no way. Here a sample of my simple dialplan : exten => 101,1,Ringing exten => 101,2,A...

...d it from softphones and pstn line phones Can anyone tell me what I'm doing wrong? Required contexts Exentions.conf below [from-broadvoice] exten => s,1,Answer exten => s,2,Playback(pbx-candles-welcome) exten => s,3,Background(pbx-candles-mainmenu) exten => 1,1,dial,SIP/101|45|r exten => 1,2,voicemail,1111 exten => 1,3,hangup() exten => 2,1,dial,SIP/101|45|r exten => 2,2,voicemail,1111 exten => 2,3,hangup() exten => 3,1,dial,SIP/102|45|r exten => 3,2,voicemail,1111 exten => 3,3,hangup() exten => 4,1,Goto(from-broadvoice,s,2) [fr...

I know I can dial multiple channels in sequence exten => 101,1,Dial(SIP/101,10) exten => 101,2,Dial(SIP/102,10) extne => 101,3,Dial(Zap/1/5551212) What the boss would really like is to be able to ring 2 lines simultaneously. exten => 101,Dial(Sip/101,10) && Dial(Sip/102,10) so that both extensions ring at the same time... mostly so that...

...he Billion ISDN on a Debian machine. I proved to call with a ISDN telephone conected to ISDN Box and it is OK. So I connect the Billion ISDN to the ISDN Box and I call from a extension to outside. But it doesn't work, that is what I have in the CLI: *CLI> -- Executing Dial("SIP/101-f9eb", "ZAP/g1/943833473|45|tTwW") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/943833473 -- Channel 0/1, span 1 got hangup, cause 42 -- Zap/1-1 is circuit-busy -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (...

...y threads, and most if not all, talk abot the dtmf setiings, but both Ekiga and Asterisk are configured for rfc2833. Here is what I get in the console: [Mar 30 10:30:23] WARNING[1790]: app_voicemail.c:7236 vm_authenticate: Couldn't read username Thanks beforehand! Alejandro Imass sip.conf [101] username=101 type=friend secret=xxxxxx qualify=yes nat=no host=dynamic canreinvite=no context=home mailbox=101 at home dtmfmode=rfc2833 extensions.conf [home] ...snip... ;internal sip extensions exten => 101,1,Dial(SIP/101,15) exten => 101,2,Voicemail(101 at home) ...snip... ;voice mai...

Hi everyone, My manager interface seems to be producing wrong CallerIDs when internal extensions call each other. Can anyone see anything wrong in the configuration snippets pasted below? The following instance has extension 101 call 103. The phone does show the right caller ID, but notice that the manager interface has the CallerID as the target number (103). Thanks a lot for your time. Manager interface output: CallerIDName: <unknown> State: Ringing Event: Newstate Privilege: call,all Uniqueid: 1201748091.843 Ch...

...ll 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(default) exten => 100,3,DigitTimeout,5 exten => 100,4,SetVar(QUEUE_PRIO=5) exten => 100,5,Background(welcome) exten => 100,6,Queue(hotline) ;VoIP Phones exten => 101,1,A...

...th callerid on outgoing calls. The recipient of the call only sees "unknown" rather than the number I'm specifying. If I set callerid info when calling an internal extension then I see the callerid name and number when I call that extension. I did that thusly: exten => 101,1,Set(CALLERID(number)=1112223333) exten => 101,2,Set(CALLERID(name)=fiznucked) exten => 101,3,Dial(SIP/officeata1,20,tr) that works. But the callerid doesn't work when I try to call out through teliax. exten => _1XXXXXXXXXX,1,Set(CALLERID(number)=1112223333...

Hello, I have some problems with mISDN. I can't send or receive call from the Billion ISDN card Mi configuration files are thoose: extensions.conf: [general] static=yes writeprotect=yes [SOME] exten => 101,1,Dial(SIP/101,30,Ttm) exten => 101,2,Hangup exten => 102,1,Dial(SIP/102,30,Ttm) exten => 102,2,Hangup include => outgoing [outgoing] exten =>_9XXXXXXXX,1,Dial(mISDN/1/${EXTEN},45,tTwW) exten =>_9XXXXXXXX,2,Hangup() exten =>_9XXXXXXXX,102,Hangup() [default] exten => s,1,...

hello, I have instaled a analog line, and when I call on the console apears that: I want to redirect the call to 101 extension. *CLI> -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at default,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at default,s,1 still failed so falling back to context 'default' Apr 27 08:15:53 WARNING[3494]: pbx.c:2377 __ast_pbx...

...nt parts from my conf files: iax.conf [general] tos=lowdelay jitterbuffer=no register => username:secret@zoot.netlogic.net [netlogic] type=friend host=dynamic context=sourcekit-main auth=plaintext username= secret= disallow=all allow=ulaw allow=all extensions.conf [sourcekit-sip] exten => 101,1,Dial(SIP/SK-101,20) exten => 101,2,Voicemail(u101) exten => 101,102,Voicemail(b101) exten => 101,103,Hangup exten => 2999,1,VoicemailMain(${CALLERIDNUM}) [sourcekit-main] include=>sourcekit-sip exten => +19193233010,1,GoTo(sourcekit-sip,101,1) exten => _1NXXNXXXXXX,1,SetCal...

...m discussed in this group but i try my best and all in vein so finally i am here hoping you ppl helping me out.I discussed this problem in asterisk's-users group and adding feedback from asterisk-users group my configs are sip.conf [general] port=5060 bindaddr=192.168.10.193 allow=all [101] username=101 type=friend secret=12345678 host=192.168.10.193 context=from-sip callerid="101"<101> defaultip=192.168.10.176 extensions.conf [globals] 101=SIP/101 [incoming] exten => s,1,Dial(Zap/1,20) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u...

Hi, I want to know how I can get a value from a table. Say, I have a table sip_buddies for storing sip user account information. There is a field called 'accountcode' that I want to get its value in the dial plan. As I find that there is no direct way to get the value from the table. Does anyone can tell me how can I get its value in the dial plan? Thanks!