search for: 2001,102

If I try to execute this dialplan, and nobody picks up at any of the three extensions (7780 7781 and 7782), it's supposed to go to voice mail; instead, it hangs up and gives me a busy signal: exten => 2001,1,Dial(sip/7780,20) exten => 2001,2,Goto(2001,102) exten => 2001,102,Dial(sip/7781,20) exten => 2001,103,Goto(2001,203) exten => 2001,203,Dial(sip/7782,20) exten => 2001,204,Goto(2001,304) exten => 2001,304,VoiceMail2(u7782) exten => 2001,305,Hangup However,...

...ret=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 ; Activate the message waiting light if this ; voicemailbox has messages in it [2001] ; Duplicate of 2000, except with different auth data type=friend username=2001 secret=1234 host=dynamic context=from-sip mailbox=101 ==>Extensions.conf [general] static=yes writeprotect=yes [bogon-calls] exten => _.,1,Congestion [from-sip] exten => 2000,1,Di...

...treams". If I enable SIP debugging on asterisk, then I find the following output "-- Got SIP response 500 "Internal server error (cannot align media streams)" back from 197.7.75.129" followed by the following debug message (no NAT) to 197.7.75.129:5060 -- SIP/2001-a513 is circuit-busy == Everyone is busy at this time We're at 197.7.75.85 port 16816 Answering with preferred capability 2147483647 Answering with non-codec capability 1 Reliably Transmitting (no NAT): SIP/2.0 200 OK Below is the configuration of asterisk SIP.CONF [general] p...

...to configure the server. When I connect to our server, SIP user-agent 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 use...

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 phone to call the phone line wh...

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

...mware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24 srvlookup = yes allow=all [2001] ;grandstream 2 type=friend username=2001 secret=1945 canreinvite=yes reinvite=yes host=dynamic dtmfmode=rfc2833 qualify=yes ;mailbox=2001 nat=1 allow=all [2002] ; soft phone type=friend username=2002 secret=1945 canreinvite=yes reinvite=yes host=dynamic dtmfmode=rfc2833 qualif...

...(IAX2/ my FWD #:my fwd password@iax2.fwdnet.net/${EXTEN:2},120,r) exten => _98*.X.X,4,Congestion exten => _7x.,1,SetCallerID(my FWD #) exten => _7x.,2,SetCIDName,"FWD.Fahad" exten => _7X.,3,Dial,Zap/1/${EXTEN:1} exten => _7x.,4,Congestion [from-sip] exten => 2001,1,Dial(SIP/2001,20) exten => 2001,2,Voicemail(u2001) exten => 2001,102,Voicemail(b2001) exten => 2001,103,Hangup exten => _7X.,1,Dial,Zap/1/${EXTEN:1} [pstn_fwd_forwarding] ;exten => s,1,Dial(IAX2/my_uname:my_secret@iax2.fwdnet.net/my_fwd_number|60|tT) exten => s...

...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 [bogon-calls] exten => _.,1,Congestion [from-sip] exten => 2000,1,Dial(SIP/2000,20) exten => 2000,102,Hangup exte...

...e cluebat and open my eyes a little. A little about the * box OS => debian (woody) Asterisk CVS-04/08/04-09:04:44 no firewalls, just on a LAN test seg.. And now for the configs.... /etc/asterisk/h323.conf [general] port = 1720 bindaddr = 0.0.0.0 ; allow=gsm dtmfmode=rfc2833 context=default ; [2001] type=friend host=192.168.1.51 context=from-h323 ;incominglimit=4 /etc/asterisk/sip.conf [general] port = 5060 bindaddr = 0.0.0.0 context = default ; disallow=all allow=gsm ; [2000] type=friend username=2000 secret=blah host=192.168.1.50 nat=1 context=from-sip /etc/asterisk/extensions.conf...

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.co...

...efault,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/iaxphone,20) ;exten => 2002,2,Voicemail(u2002) ;exten => 2002,102,Voicemail(b2002) ;exten => 2002,103,Hangup exten => 2999,1,Voicemai...

...cking up the configs all ok. Everything _seems_ to be working, but I cant make any calls - either internally or externally. (apologies in advance for the copious code below) Having a look, I have placed the following lines into the extensions.cfg file to allow for the extensions to work. [2001] exten => 2001,1,Dial(SIP/2001,15,t) exten => 2001,2,Voicemail(u2001) exten => 2001,102,Voicemail(b2001) exten => 2001,103,Hangup [2002] exten => 2002,1,Dial(SIP/2002,15,t) exten => 2002,2,Voicemail(u2002) exten => 2002,102,Voicemail(b2002) exten => 2002,103,Hang...

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...

...vironmental variable ; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid ; [globals] CONSOLE=/dev/dsp [local] ; Operator/ Console exten => 2000,1,Dial,Zap/1|30 exten => 2000,2,Voicemail,u2000 exten => 2000,102,Voicemail,b2000 ; SIP Phones exten => 2101,1,Dial,SIP/snom1|30 exten => 2001,2,Voicemail,u2001 exten => 2001,102,Voicemail,b2001 exten => 2102,1,Dial,SIP/snom2|30 exten => 2002,2,Voicemail,u2002 exten => 2002,102,Voicemail,b2002 exten => 2103,1,Dial,SIP/snom3|30 exten => 2003,2,Voicemail,u2003 exten => 2003,102,Voicemail,b2003 exten => 2104,1,Dial...

Hi, I am very new to Asterisk. I wanted to know how to retrive the Voicemails. I could see some voicemails assosiated with some extensions. Any ideas?? --------------------------------- How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo! Photos -------------- next part -------------- An HTML attachment was scrubbed... URL:

...ts in ; a "busy" code, then Dial will jump to 101 + (current priority) ; which in our case will be 101+1=102. This +101 jump is built ; into Asterisk and does not need to be defined. ; exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup ; Now, what if the number dialed was 2001? 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(SIP/2002,20) exten => 2002,2,Voicemail(u2002) exten => 2002,102,Voicemail(b2002) exten => 2002,103,Hangup exten => 2003,...

hi do u have the sip phones extensions in the extension.conf and are they in the right context (sip-incoming)??? are the sip phone registering to asterisk?? try stop asterisk and reconnect as asterisk -vvvvvvvc to check see them registering... ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Sistemista WebSolvingJaa Sent: Fri 7/1/2005 6:43 PM

...evice host=dynamic ; This host is not on the same IP addr every time ;context=from-sip ; Inbound calls from this host go here mailbox=2000 ; Activate the message waiting light if this ; voicemailbox has messages in it dtmfmode=rfc2833 [2001] ; Duplicate of 2000, except with different auth data type=friend username=2001 secret=homemonkey host=dynamic ;context=from-sip mailbox=2001 dtmfmode=rfc2833 [2002] ; Duplicate of 2000, except with different auth data type=friend username=2002 secret=homemonkey ho...

Hi This is hard work :) I have read the Asterisk Handbook, BudgeTone User Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource Pages and more. I am not a linux newbie but am new to Asterisk. I have failed to find any docs that explain how to get a very very simple, minimal, system up and I am trying to get the following to work: 2 BudgePhone 102D connected on a LAN to a