i always use the Goto application. seems to work quite well for testing those "s" extensions.
exten = 2500,1,Goto(context,s,1) will take you to step 1 in the s extension in whatever context. Jason Kawakami ----- Original Message ----- From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, May 24, 2004 5:20 PM Subject: Asterisk-Users digest, Vol 1 #3886 - 9 msgs > Send Asterisk-Users mailing list submissions to > [EMAIL PROTECTED] > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > [EMAIL PROTECTED] > > You can reach the person managing the list at > [EMAIL PROTECTED] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > > 1. Re: Re: Making a SIP call (Eric Wieling) > 2. RE: testing asterisk on FXS lines (Jay Milk) > 3. SIP Authentication Problem (Chuck Ramirez) > 4. RE: 2 Sip phones behind un-natted Asterisk (Chad Brown) > 5. Re: extensions/sip from database? (Fran Boon) > 6. Using Blacklist (Steven E. Frazier) > 7. Asterisk connected to DataBase (pesb) > 8. mpg123 (Simon Brown) > 9. Re: Using Blacklist (Dorian Gray) > > --__--__-- > > Message: 1 > Date: Mon, 24 May 2004 16:20:36 -0500 > From: Eric Wieling <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Re: Making a SIP call > Reply-To: [EMAIL PROTECTED] > > [EMAIL PROTECTED] wrote: > > I am still having this problem of only capturing part of the IP address, I > > am currently checking into a possible hardware/software issue on the > > client side but was wondering if there are any setting I need to set on > > the asterisk server to allow an peer to peer call. I have set > > dtmfmode=inband. Is there anything else I need to set? > > dtmfmode=inband only works with the ulaw and alaw codecs. If you use > any other codec you MUST use rfc2833 or info DTMF modes (set on the > phone AND on Asterisk) > > --__--__-- > > Message: 2 > From: "Jay Milk" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] testing asterisk on FXS lines > Date: Mon, 24 May 2004 16:29:39 -0500 > Reply-To: [EMAIL PROTECTED] > > For $49.99+S&H I can sell you an FXO/FXS test-cable... just kidding. > Use a regular RJ11 cable to connect one of your FXS ports to the FXO > port you want to test, pick up another FXS and dial the extension... and > you're promptly delivered to the [incoming] context. I test all my FXO > configs using a Sipura FXS port to make it ring. I'd still like that > $50 though :) > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Michael > George > Sent: Monday, May 24, 2004 3:57 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] testing asterisk on FXS lines > > > On May 24, 2004, at 4:00 PM, Michael George wrote: > > I am configuring an asterisk server and I want to test the incoming > > configuration with my FXS handsets. > > > > I have the FXS lines able to call eachother and they can connect out > > the FXO lines. > > > > I changed the context for the FXS lines to "incoming" so that they > > would be able to test the setup for incoming calls. > > > > For the incoming context I have: > > [incoming] > > exten => s,1,Wait(1) > > exten => s,2,Answer() > > exten => s,3,Background(hello2) ; this is the file I need to test the > > playback of first > > > > And I do a restart. When I pickup one of the FXS handsets, though, I > > get this from asterisk (running with the -vvvc arg): > > Starting simple switch on 'Zap/1-1' > > and that is it. > > > > I know that the context is right because I put a hard-dial of "202" in > > there and when I dialed it, it would connect to that extension (Zap/2) > > > and if I dialed anything else I would get fast busy. > > > > I have checked and the line right after the last exten above is > > another context marker. > > > > The asterisk output also shows the s extensions being loaded under the > > correct context when I do a reload after the restart (to see just the > > messages from the contexts being loaded). > > > > What am I missing to get the FXS lines, in the context "incoming", to > > do the wait/answer/background? > > > > Thanks! > > For some reason, the s extension is not being executed for the FXS > lines. I changed their default context back to "internal" and added > "exten => s,1,Background(hello2)" to the internal context, thinking > that when I pick up the handset I will get the hello2 audio file played > as it waits for me to enter digits. > > But the audio file is not played... I must be missing an essential > concept here... > > -Michael > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > --__--__-- > > Message: 3 > Date: Mon, 24 May 2004 14:25:00 -0700 (PDT) > From: Chuck Ramirez <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] SIP Authentication Problem > Reply-To: [EMAIL PROTECTED] > > --0-909188567-1085433900=:35567 > Content-Type: text/plain; charset=us-ascii > > > I have a group of users configured as extensions in *.These users are registered with a SIP Proxy Server and can receive calls very well. The problem happens when any user tries to make an outbound call. The proxy replies with a "401 Unauthorized" and * don't try another INVITE including credentials. > > Here is part of the content of sip.conf. > > [general] > port = 5061 > bindaddr = *.IP > context = invalidcalls > > ;This account is used for inbound and outbound calls > register => myuser:[EMAIL PROTECTED]/999 > > [mydomain] > type=peer > host=myproxy > context=sip > username=myuser > secret=mypass > fromuser=myuser > fromdomain=mydomain > > [user1] > type=friend > host=dynamic > defaultip=default.IP > username=user1 > secret=secret1 > dtmfmode=rfc2833 > context=users > callerid="User 1" > nat=yes > > > > Here is part of the content of extensions.conf. > > ;This part is working fine > [sip] > exten => 999,1,Dial(SIP/user1,,tr) > > [users] > exten => _8.,1,Dial,SIP/[EMAIL PROTECTED],tr > > > > When I dial the number 812345 from my SIP Phone, this is the message sequence > Phone -> Asterisk: INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Asterisk -> Phone: SIP/2.0 407 Proxy Authentication Required > Phone -> Asterisk: ACK sip:[EMAIL PROTECTED] SIP/2.0 > Phone -> Asterisk: INVITE sip:[EMAIL PROTECTED] SIP/2.0 (with authentication header) > Asterisk -> Phone: SIP/2.0 100 Trying > Asterisk -> Proxy: INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Proxy -> Asterisk: SIP/2.0 401 Unauthorized > Asterisk -> Proxy: ACK sip:[EMAIL PROTECTED] SIP/2.0 > > The next message I would expect is another INVITE from * to the proxy with the authentication header. > Why * hasn't send it? Can someone give me a help? > > Thanks in advance > Chuck Ramirez > > > > > --------------------------------- > Do you Yahoo!? > Friends. Fun. Try the all-new Yahoo! Messenger > --0-909188567-1085433900=:35567 > Content-Type: text/html; charset=us-ascii > > <P align=left>I have a group of users configured as extensions in *.These users are registered with a SIP Proxy Server and can receive calls very well. The problem happens when any user tries to make an outbound call. The proxy replies with a "401 Unauthorized" and * don't try another INVITE including credentials.</P> > <P align=left>Here is part of the content of sip.conf.</P> > <P align=left>[general]<BR>port = 5061<BR>bindaddr = *.IP<BR>context = invalidcalls</P> > <P align=left>;This account is used for inbound and outbound calls<BR>register => myuser:[EMAIL PROTECTED]/999</P> > <P align=left>[mydomain]<BR>type=peer<BR>host=myproxy<BR>context=sip<BR>usernam e=myuser<BR>secret=mypass<BR>fromuser=myuser<BR>fromdomain=mydomain</P> > <P align=left>[user1]<BR>type=friend<BR>host=dynamic<BR>defaultip=default.IP<BR >username=user1<BR>secret=secret1<BR>dtmfmode=rfc2833<BR>context=users<BR>ca llerid="User 1"<BR>nat=yes</P> > <P align=left> </P> > <P align=left>Here is part of the content of extensions.conf.</P> > <P align=left>;This part is working fine<BR>[sip]<BR>exten => 999,1,Dial(SIP/user1,,tr)</P> > <P align=left>[users]<BR>exten => _8.,1,Dial,SIP/[EMAIL PROTECTED],tr</P> > <P align=left> </P> > <P align=left>When I dial the number 812345 from my SIP Phone, this is the message sequence<BR>Phone -> Asterisk: INVITE sip:[EMAIL PROTECTED] SIP/2.0<BR>Asterisk -> Phone: SIP/2.0 407 Proxy Authentication Required<BR>Phone -> Asterisk: ACK sip:[EMAIL PROTECTED] SIP/2.0<BR>Phone -> Asterisk: INVITE sip:[EMAIL PROTECTED] SIP/2.0 (with authentication header)<BR>Asterisk -> Phone: SIP/2.0 100 Trying<BR>Asterisk -> Proxy: INVITE sip:[EMAIL PROTECTED] SIP/2.0<BR>Proxy -> Asterisk: SIP/2.0 401 Unauthorized<BR>Asterisk -> Proxy: ACK sip:[EMAIL PROTECTED] SIP/2.0</P> > <P align=left>The next message I would expect is another INVITE from * to the proxy with the authentication header.<BR>Why * hasn't send it? Can someone give me a help?</P> > <P align=left>Thanks in advance<BR> Chuck Ramirez</P><BR><BR><p> > <hr size=1><font face=arial size=-1>Do you Yahoo!?<br>Friends. Fun. <a href="http://messenger.yahoo.com/";>Try the all-new Yahoo! Messenger</a> > --0-909188567-1085433900=:35567-- > > --__--__-- > > Message: 4 > Subject: RE: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk > Date: Mon, 24 May 2004 14:36:00 -0700 > From: "Chad Brown" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Reply-To: [EMAIL PROTECTED] > > After further investigation it looks like it was as simple as both > phones trying to listen on the same port. I will continue testing to > verify. > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Shaun Dawson > Sent: Monday, May 24, 2004 10:03 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk > > What does the Xten diagnostic log say about a single > session? > > Also, what does the * SIP debug output say? I'd be > very interested to see what IPs and ports SIP is > trying to set the RTP connection on. (Since SIP > appears to be working fine, it's the RTP part that is > breaking). > > Are both the Xten and the 7960 trying to listen on the > same RTP port (my Xten is configured to listen on > 8000)? > > Pardon me if I sound like an idiot, but I'm somewhat > new to VoIP, SIP _and_ Asterisk. :) > > Shaun > > > --- Bruce Komito <[EMAIL PROTECTED]> wrote: > > John, In my case, the two ports are not using the > > same IP port (one is on > > 5060, the other on 5061), but of course, they are on > > the same IP address. > > I think that is what is confusing the NAT server, > > but I don't know what to > > do about it. > >=20 > > Bruce Komito > > High Sierra Networks, Inc. > > www.servers-r-us.com > > (775) 284-5800 ext 115 > >=20 > >=20 > > On Mon, 24 May 2004, John Fraizer wrote: > >=20 > > > Chad Brown wrote: > > > > > > > I have 2 SIP phones (Cisco 7960 & XTen) behind a > > NAT provided by a > > > > Linksys firewall that supports UPnP. The > > Asterisk server has a public > > > > IP. Here are the problems that I am having with > > this configuration... > > > > > > > > > > > > > > > > 1. The 2 SIP phones can call MeetMe and have > > a conference but cannot > > > > call each other. (Yes, they connect but no > > audio either direction) > > > > 2. I have verify=3Dyes in the sip.conf for both > > phones. Both phones > > > > constantly go Unreachable. (However, the > > connection is very fast > > > > between * and sip phones) > > > > 3. Sometimes but not always when I try to > > call phone1 phone2 rings. > > > > > > > > > > > > > > > > Is this Nat messing with me or something else? > > In any case...Any advice > > > > out there? > > > > > > > > > > > > > > > > Thanks, > > > > > > > > Chad > > > > > > > > > > > > > The problem is probably that both of your SIP > > phones are using the same > > > port. I played with two 7960's behind a Linksys > > on Saturday and finally > > > got them playing right when I changed the > > following: > > > > > > In Phone 1's SIP[macaddr].cnf: > > > > > > voip_control_port: 5061 > > > > > > In Phone 2's SIP[macaddr].cnf: > > > > > > voip_control_port: 5062 > > > > > > The default control port is 5060. Note: This is > > the port that the > > > PHONE uses to initiate the connection to * and not > > the port it is > > > connecting to. > > > > > > John > > > _______________________________________________ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > > =20 > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > >=20 > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > =20 > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > =09 > =09 > __________________________________ > Do you Yahoo!? > Yahoo! Domains - Claim yours for only $14.70/year > http://smallbusiness.promotions.yahoo.com/offer=20 > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > --__--__-- > > Message: 5 > Date: Mon, 24 May 2004 22:50:44 +0100 > From: Fran Boon <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] extensions/sip from database? > Reply-To: [EMAIL PROTECTED] > > Manuel Wenger wrote: > > We are planning to deploy a pretty large asterisk server with many SIP extensions (might be up to 10000 in the future), and I have a few questions: > > 1) is this possible, or are we running into some kind of limitation in the software that I wasn't aware of and that I didn't find by browsing through the archives and through Wiki? No, we don't need any G729-G711 transformations, it would only be acting as a SIP proxy (even if asterisk isn't a proxy). > > /Should/ be psosible with canreinvite=yes & no use of T,t in the dial > commands, so that Asterisk can stay out of the media path except when > absolutely necessary. > > > 2) is there a way to store extensions.conf and/or sip.conf in some kind of database, maybe MySQL? This would make life easier if someone wanted to change his SIP password. Or how would you otherwise solve this problem? > > http://voip-info.org/wiki-Asterisk+configuration+from+database > Option 1 is being enhanced through the development of ast_data. > I currently use Option 2 > > > 3) is there a quick way of reloading only a part of sip.conf/extensions.conf, for example if only a user password changed, or an extension's behaviour (eg. routing to voicemail instead of a SIP user)? > > sip reload > extensions reload > > That's as granular as it gets. > Should be harmless to keep doing this, though. > > > Maybe I'm looking at the wrong software here and SER would be better for what I want to do... I know asterisk is supposed to be a PBX replacement, but the functions and flexibility it has really tells me "stick with asterisk". Or am I way off with these assumptions? > > Possibly - depends whether you're after a SIP proxy or a PBX ;) > > F > > --__--__-- > > Message: 6 > From: "Steven E. Frazier" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Date: Mon, 24 May 2004 17:55:17 -0400 > Subject: [Asterisk-Users] Using Blacklist > Reply-To: [EMAIL PROTECTED] > > I am attempting to write in incoming context for calls. > > 1. If the caller id is given and it is not black listed it will Playback = > a > greeting and then right the phone or go to voicemail under busy or > unavailable conditions > 2. If no caller id is given, then Privacy Manager will ask for the = > number. > I am testing 6145551212 to see if the black list will work > 3. If a caller id is given, and it is blacklisted (in the blacklist db) = > I > would like for it to go to Playback/black-list-blocked message > > > > > The db shows: > > asterisk*CLI> database show blacklist > /blacklist/<1010987/18887975686number> : 1 > > /blacklist/<name/number> : 1 > > /blacklist/unlisted/6145551212 : 1 > > asterisk*CLI> > > > exten =3D> 2129,1,Wait(1) > exten =3D> 2129,2,Zapateller(answer|nocallerid) > exten =3D> 2129,3,NoOp > exten =3D> 2129,4,PrivacyManager > exten =3D> 2129,5,LookupBlacklist > exten =3D> 2129,6,Dial(Zap/4,5,Ttr) > exten =3D> 2129,7,Answer > exten =3D> 2129,8,Wait(1) > exten =3D> 2129,9,Playback(personal/hello) > exten =3D> 2129,10,Playback(personal/i-am-not-in-at-the-moment) > exten =3D> 2129,11,VoiceMail2(u${EXTEN}) > exten =3D> 2129,12,Hangup > exten =3D> 2129,102,VoiceMail2(b${EXTEN}) ; Busy Voicemail if extension = > is > busy > exten =3D> 2129,106,Playback,personal/black-list-blocked > exten =3D> 2129,108,Wait(2) > exten =3D> 2129,110,Hangup > > When I dial my test extension of 2129, I get: > > > asterisk*CLI>=20 > -- Starting simple switch on 'Zap/7-1' > -- Disabling Caller*ID on Zap/7-1 > -- Executing Wait("Zap/7-1", "1") in new stack > -- Executing Zapateller("Zap/7-1", "answer|nocallerid") in new stack > -- Executing NoOp("Zap/7-1", "") in new stack > -- Executing PrivacyManager("Zap/7-1", "") in new stack > =3D=3D Parsing '/etc/asterisk/privacy.conf': =3D=3D Parsing > '/etc/asterisk/privacy.conf': Found > -- Playing 'privacy-unident' (language 'en') > -- Playing 'privacy-prompt' (language 'en') > -- Playing 'privacy-thankyou' (language 'en') > -- Changed Caller*ID to "Privacy Manager" <6145551212> > -- Executing LookupBlacklist("Zap/7-1", "") in new stack > -- Executing Dial("Zap/7-1", "Zap/4|5|Ttr") in new stack > -- Called 4 > -- Zap/4-1 is ringing > -- Zap/4-1 is ringing > -- Nobody picked up in 5000 ms > -- Hungup 'Zap/4-1' > -- Executing Answer("Zap/7-1", "") in new stack > -- Executing Wait("Zap/7-1", "1") in new stack > -- Executing Playback("Zap/7-1", "personal/hello") in new stack > -- Playing 'personal/hello' (language 'en') > -- Executing Playback("Zap/7-1", = > "personal/i-am-not-in-at-the-moment") > in new stack > -- Playing 'personal/i-am-not-in-at-the-moment' (language 'en') > -- Executing VoiceMail2("Zap/7-1", "u2129") in new stack > -- Playing 'vm-theperson' (language 'en') > -- Playing 'digits/2' (language 'en') > -- Playing 'digits/1' (language 'en') > -- Playing 'digits/2' (language 'en') > -- Playing 'digits/9' (language 'en') > -- Playing 'vm-isunavail' (language 'en') > -- Playing 'vm-intro' (language 'en') > -- Playing 'beep' (language 'en') > -- Recording the message > > It goes to the unavailable voice mail box. > > According to the documentation and my understanding: > > > LookupBlacklist: Looks up the Caller*ID number on the active channel in = > the > Asterisk database (family 'blacklist'). If the number is found, and if = > there > exists a priority n + 101, where 'n' is the priority of the current > instance, then the channel will be setup to continue at that priority = > level. > Otherwise, it returns 0. Does nothing if no Caller*ID was received on = > the > channel.=20 > Example: database put blacklist <name/number> 1 > > > Could someone tell me what I am doing wrong that it won't go to Priority = > 106 > and Playback black-list-blocked. > > Would someone share their context that is using blacklist to show me how > they are doing it? > > Thanks. > > --__--__-- > > Message: 7 > From: pesb <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Date: Mon, 24 May 2004 17:58:30 -0400 > Subject: [Asterisk-Users] Asterisk connected to DataBase > Reply-To: [EMAIL PROTECTED] > > Hi there, > I want to have all my sip.conf data inside a DataBase, so that my asterisk= > =20 > admintration system would be through a Web Interface connected to the DB. > Is there any way to put the sip.conf file in a Data Base and then to read= > =20 > from it, in such a way that the sip.conf file would have some line that=20 > points to the DataBase? > I have seen wiki's page=20 > http://www.voip-info.org/wiki-Asterisk+configuration+from+database > Possibility n=BA 2 and 3 do not convince myself. > I have tried possibility n=BA1(Dynamic), but did not find much info about t= > he=20 > command DBget. Could somebody give some info on how to use it? > Or, could someone recommend me another scheme that could work? > > thanks in advance, > Pablo Salinas > > > > > --__--__-- > > Message: 8 > Date: Tue, 25 May 2004 08:11:30 +1000 > From: "Simon Brown" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] mpg123 > Reply-To: [EMAIL PROTECTED] > > When I start * I get 6 mpg123 processes start as well. Is this normal? > Often after a couple of days these mpg123 processes start to consume cpu = > and > I have to kill them off. > I do not have a sound card in the server and I have noload =3D> = > chan_oss.so > > Simon > > --__--__-- > > Message: 9 > Date: Mon, 24 May 2004 18:17:09 -0400 > From: Dorian Gray <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Using Blacklist > Reply-To: [EMAIL PROTECTED] > > the following has been working well for me, and I think it does similar > to what you want... > > > [macro-blackdrop] > exten => s,1,Playback(giggle1) > ; something is terribly wrong...! > exten => s,2,Playback(tt-somethingwrong) > ; oh, it's those damnable weasels again...! > exten => s,3,Playback(tt-weasels) > exten => s,4,Playback(goodbye) > exten => s,5,Hangup > > [inbound-analog] > exten => s,1,SetMusicOnHold,random > exten => s,2,Zapateller(answer|nocallerid) > exten => s,3,NoOp > exten => s,4,PrivacyManager > exten => s,5,LookupCIDName > exten => s,6,LookupBlacklist > exten => s,7,Background(pls-wait-connect-call) > exten => s,8,Dial(${PHONE1}&${PHONES1},20,Ttm) > exten => s,9,Answer > exten => s,10,Wait(1) > exten => s,11,Macro(vmessage,${PHONE1VM}) > exten => s,105,Macro(blackdrop) > exten => s,107,Macro(blackdrop) > > hm maybe I should move lookupcidname after lookupblacklist and save a > few cycles ^_^ > > > > Steven E. Frazier wrote: > > I am attempting to write in incoming context for calls. > > > > 1. If the caller id is given and it is not black listed it will Playback a > > greeting and then right the phone or go to voicemail under busy or > > unavailable conditions > > 2. If no caller id is given, then Privacy Manager will ask for the number. > > I am testing 6145551212 to see if the black list will work > > 3. If a caller id is given, and it is blacklisted (in the blacklist db) I > > would like for it to go to Playback/black-list-blocked message > > > > > > > > > > The db shows: > > > > asterisk*CLI> database show blacklist > > /blacklist/<1010987/18887975686number> : 1 > > > > /blacklist/<name/number> : 1 > > > > /blacklist/unlisted/6145551212 : 1 > > > > asterisk*CLI> > > > > > > exten => 2129,1,Wait(1) > > exten => 2129,2,Zapateller(answer|nocallerid) > > exten => 2129,3,NoOp > > exten => 2129,4,PrivacyManager > > exten => 2129,5,LookupBlacklist > > exten => 2129,6,Dial(Zap/4,5,Ttr) > > exten => 2129,7,Answer > > exten => 2129,8,Wait(1) > > exten => 2129,9,Playback(personal/hello) > > exten => 2129,10,Playback(personal/i-am-not-in-at-the-moment) > > exten => 2129,11,VoiceMail2(u${EXTEN}) > > exten => 2129,12,Hangup > > exten => 2129,102,VoiceMail2(b${EXTEN}) ; Busy Voicemail if extension is > > busy > > exten => 2129,106,Playback,personal/black-list-blocked > > exten => 2129,108,Wait(2) > > exten => 2129,110,Hangup > > > > When I dial my test extension of 2129, I get: > > > > > > asterisk*CLI> > > -- Starting simple switch on 'Zap/7-1' > > -- Disabling Caller*ID on Zap/7-1 > > -- Executing Wait("Zap/7-1", "1") in new stack > > -- Executing Zapateller("Zap/7-1", "answer|nocallerid") in new stack > > -- Executing NoOp("Zap/7-1", "") in new stack > > -- Executing PrivacyManager("Zap/7-1", "") in new stack > > == Parsing '/etc/asterisk/privacy.conf': == Parsing > > '/etc/asterisk/privacy.conf': Found > > -- Playing 'privacy-unident' (language 'en') > > -- Playing 'privacy-prompt' (language 'en') > > -- Playing 'privacy-thankyou' (language 'en') > > -- Changed Caller*ID to "Privacy Manager" <6145551212> > > -- Executing LookupBlacklist("Zap/7-1", "") in new stack > > -- Executing Dial("Zap/7-1", "Zap/4|5|Ttr") in new stack > > -- Called 4 > > -- Zap/4-1 is ringing > > -- Zap/4-1 is ringing > > -- Nobody picked up in 5000 ms > > -- Hungup 'Zap/4-1' > > -- Executing Answer("Zap/7-1", "") in new stack > > -- Executing Wait("Zap/7-1", "1") in new stack > > -- Executing Playback("Zap/7-1", "personal/hello") in new stack > > -- Playing 'personal/hello' (language 'en') > > -- Executing Playback("Zap/7-1", "personal/i-am-not-in-at-the-moment") > > in new stack > > -- Playing 'personal/i-am-not-in-at-the-moment' (language 'en') > > -- Executing VoiceMail2("Zap/7-1", "u2129") in new stack > > -- Playing 'vm-theperson' (language 'en') > > -- Playing 'digits/2' (language 'en') > > -- Playing 'digits/1' (language 'en') > > -- Playing 'digits/2' (language 'en') > > -- Playing 'digits/9' (language 'en') > > -- Playing 'vm-isunavail' (language 'en') > > -- Playing 'vm-intro' (language 'en') > > -- Playing 'beep' (language 'en') > > -- Recording the message > > > > It goes to the unavailable voice mail box. > > > > According to the documentation and my understanding: > > > > > > LookupBlacklist: Looks up the Caller*ID number on the active channel in the > > Asterisk database (family 'blacklist'). If the number is found, and if there > > exists a priority n + 101, where 'n' is the priority of the current > > instance, then the channel will be setup to continue at that priority level. > > Otherwise, it returns 0. Does nothing if no Caller*ID was received on the > > channel. > > Example: database put blacklist <name/number> 1 > > > > > > Could someone tell me what I am doing wrong that it won't go to Priority 106 > > and Playback black-list-blocked. > > > > Would someone share their context that is using blacklist to show me how > > they are doing it? > > > > Thanks. > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > --__--__-- > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
