search for: 2000,102

I'm trying to see if this is even possible. When you dial ext 2000 I want it to ring my sip phone for 20 sec then call my cell and let it ring for 10 sec if I do not pick up the call on my cell I would like it to go back to * and leave a voice message for me. Here is what I have so far in my extensions.conf Everything works except the call will not go back to * a...

Hi all, I have the Asterisk set up and 2 softphone (Xlite) set up on two other PC. With the following configuration, I can use one softphone (2000) to call the other one (2001) and/or the voicemail at 2999. Here is my problem: 1. When I dial 9+nxxx-xxxx with one of the softphone to the PSTN via X100P card, I got busy tone. (i.e. I want to use the phone line which is connected to the X100P to call out) 2. When I use my cell p...

...ning Linux ==>SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all ; Allow all codecs allow=ulaw context = bogon-calls ; Send SIP callers that we don't know about here [2000] type=friend ; This device takes and makes calls username=2000 ; Username on device secret=1234 ; Password for device host=dynamic ; This host is not on the same IP addr every time context=from-sip ; Inbound calls from this host go here mailbox=100...

...; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls disallow=all allow=ulaw allow=alaw tos=lowdelay ;tos=184 maxexpirey=18000 ; Max length of incoming registration we allow defaultexpirey=12000 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY [opoffice] type=friend secret=opoffice host=dynamic dtmfmode=rfc2833 mailbox=1000 context=local callerid="Operator Office" <1000> [opfield] type...

...configuration of asterisk SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) allow=all ; Allow all codecs context = bogon-calls ; Send SIP callers that we don't know about here [2000] type=soft1 ; This device takes and makes calls username=2000 ; Username on device secret=friend ; Password for device host=dynamic ; This host is not on the same IP addr every time context=from-sip ; Inbound calls from this host go here mailbox=100 ; Acti...

...ng exten => _72X.,1,DBput(SIP/${CALLERIDNUM}=${EXTEN:2}) exten => _72X.,2,SayDigits,${CALLERIDNUM} exten => _72X.,3,SayDigits,${EXTEN:2} exten => _72X.,4,Hangup exten => 71,1,DBdel(SIP/${CALLERIDNUM}) exten => 71,2,Hangup ; i'm sure we can do this with macro's exten => 2000,1,DBget(temp=SIP/2000) exten => 2000,2,Goto(${temp}|1) exten => 2000,102,Goto(2000|3) exten => 2000,3,DBget(dnd=DND/2000) exten => 2000,4,Goto(2000|6) exten => 2000,104,Goto(2000|5) exten => 2000,5,Dial(SIP/2000,20) exten => 2000,6,Voicemail2(u2000) exten => 2000,7,Hangup ex...

...shows that I am logged in. But it doesn't show my system(client) IP when I issue command at astrisk CLI. The O/P is as below. *CLI> sip show peers Name/username Host Mask Port Status 2001/2001 (Unspecified) (D) 255.255.255.255 0 UNKNOWN 2000/2000 (Unspecified) (D) 255.255.255.255 0 UNKNOWN I am pasting sip.conf & extension.conf sip.conf [general] port = 5060 bindaddr = 0.0.0.0 context = INVALID ;Autocreatepeer= yes [2000] type=friend username=2000 secret=2000 host=dynamic context=from-sip mailbox=100 canrein...

...c reinvite=no canreinvite=no dtmfmode=inband callerid="Fred F"<200> dissallow=all Extensions.conf [default] exten => 1000,1,Dial,Zap/1|20 exten => 1000,2,Voicemail,u1000 exten => 1000,3,Hangup exten => 1000,102,Voicemail,b1000 exten => 1000,103,Hangup exten => 2000,1,Dial,Zap/2|20 exten => 2000,2,Voicemail,u2000 exten => 2000,3,Hangup exten => 2000,102,Voicemail,b2000 exten => 2000,103,Hangup exten => _NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30) ; Dial Broadvoice for 30 seconds exten => _NXXNXXXXXX, 2, congestion() ; No answer,...

...; IP address to bind to (0.0.0.0 binds to all) allow=all [3000] type=friend allow=all username=3000 secret=my_passwd host=dynamic context=sip dtmfmode=rfc2833 my extension.conf [globals] davidHand=>Zap/1 davidVoicemail=>1000@microfortune johnHand=>Zap/2 johnVoicemail=>2000@microfortune davidout=>Zap/3 johnout=>Zap/4 [internal] exten => 1000,1,Dial(${davidHand},10,r) exten => 1000,n,Voicemail(u${davidVoicemail}) exten => 1000,n,Playback(/var/lib/asterisk/sounds/vm-goodbye) exten => 1000,n,Wait(1) exten => 1000,n,Hangup() exten => 1000,102,Voic...

I have a few SIP phones, Cisco 7960s, and was looking into implementing some compression, ala G.729. I'm looking into purchasing a g729 licenses just to get an idea of performance and voice quality, over lans, wireless and single channel isdn. Does anyone have positive/negative experience w/ getting licenses/support from Digium? Hows the sound quality compared w/ g.711? Is 729 better

...stered for '{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}' = = Everyone is busy at this time My config files are below: sip.conf [general] port = 5060 bindaddr = 0.0.0.0 disallow=all allow=gsm allow=ulaw allow=alaw allow=G723.1 context=from-sip [2000] type=friend username=2000 secret=xxxxxxxx host=dynamic mailbox=2000 extensions.conf [general] static=yes writeprotect=yes [globals] PSTN-1=Zap/1 [from-sip] exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hang...

...gt; Proxy Server <----------------> User Agent (Ubiquity) Asterisk SIP (Ubiquity) My sip.conf and extensions.conf is as follows: sip.conf [general] port = 5060 bindaddr = 0.0.0.0 allow=all context = bogon-calls [2000] type=friend username=2000 secret=9overthruster7 host=dynamic context=from-sip [2001] type=friend username=2001 secret=11bbanzai9 host=dynamic context=from-sip extensions.conf [general] static=yes writeprotect=yes [b...

...terisk sip.conf [general] port = 5060 bindaddr = 192.168.255.33 disallow=all allow=ulaw allow=alaw context=bogon-calls [iconnect] context=from-sip type=peer secret=******* username=35205*** host=natrelay.deltathree.com dtmf=rfc2833 callerid="Me" <35205***> canreinvite=no nat=yes [2000] type=friend username=2000 secret=***** host=dynamic defaultip=192.168.255.54 context=from-sip mailbox=2000@local dtmfmode=info callerid="Me" <2000> My extensions.conf [general] static=yes writeprotect=yes [bogon-calls] exten => _.,1,Congestion [from-sip] exten =>...

...; IP address to bind to (0.0.0.0 binds to all) allow=all [3000] type=friend allow=all username=3000 secret=my_passwd host=dynamic context=sip dtmfmode=rfc2833 my extension.conf [globals] davidHand=>Zap/1 davidVoicemail=>1000@microfortune johnHand=>Zap/2 johnVoicemail=>2000@microfortune davidout=>Zap/3 johnout=>Zap/4 [internal] exten => 1000,1,Dial(${davidHand},10,r) exten => 1000,n,Voicemail(u${davidVoicemail}) exten => 1000,n,Playback(/var/lib/asterisk/sounds/vm-goodbye) exten => 1000,n,Wait(1) exten => 1000,n,Hangup() exten => 1000,102,Voic...

Hi All We are unable to hear any voice where as in tcpdum it shows that RTP is flowing both ways ERROR CONDITION --------------- -- Executing Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack -- Called 2000 -- SIP/2000-0ead is ringing -- SIP/2000-0ead answered SIP/2001-f6c4 -- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead Have searched web and archive w/o good results. Thks in advance for any help, sip.conf -------- [general] port...

...1) [inbound] ; This is the list if inbound lines exten => 2181,1,Answer exten => 2181,2,Playback(silence/1) exten => 2181,3,Goto(default,main,1) exten => 2181,3,Hangup exten => h,1,Hangup exten => t,1,Hangup exten => i,1,Hangup exten => T,1,Hangup [extentions] exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup exten => 2001,1,Dial(SIP/2001,20) exten => 2001,2,Voicemail(u2001) exten => 2001,102,Voicemail(b2001) exten => 2001,103,Hangup ;exten => 2002,1,Dial(IAX2/iaxpho...

ERROR CONDITION --------------- -- Executing Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack -- Called 2000 -- SIP/2000-0ead is ringing -- SIP/2000-0ead answered SIP/2001-f6c4 -- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead Have searched web and archive w/o good results. Thks in advance for any help, Dave sip.conf -------- [general]...

I am just starting out with * so bear with me please. I have tdm400p with 4 fxo modules on it. When I call into the asterisk box from my mobile, I can see the asterisk console picks the call up and routes it to my computer with x-lite. There was no sound coming from either - just silence. I then decided to route it directly to voice mail to see if that would narrow the problem down, but it

...----------------------------------------------------------------------------------- /etc/asterisk/zapata.conf ------------------------- ; ; Zapata telephony interface ; ; Configuration file [channels] ; ; Default language ; language=en ; ; Default context ; callerid="PBX Operator" <2000> signalling=fxo_ks relaxdtmf=yes channel=>1 context=local --------------------------------------------------------- /etc/asterisk/sip.conf ----------------------- ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0...

Problem I have is this. outside firewall (extension 2003) can call me inside firewall (extension 2000) and all is fine. If I call from inside firewall (extension 2000) to outside firewall (extension 2003) I hear no ringing and person at other end can pick up and I hear for maybe a half second then I go to voicemail. If I add another extension on the outside then communication between outside and...