[Asterisk-Users] testing asterisk on FXS lines
Adam Goryachev
mailinglists at websitemanagers.com.au
Mon May 24 19:34:36 MST 2004
Look in your zapata.conf (hmmm, or zaptel.conf I awlays get confused,
the one in /etc/asterisk/zap???.conf)
You need to add the line:
immediate = yes
This means as soon as you pick up the line, it will follow the 's'
extension.
(You will need this defined for your fxo interface as well later)
Regards,
Adam
On Tue, 2004-05-25 at 10:46, Jason Kawakami wrote:
> 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: <asterisk-users-request at lists.digium.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Monday, May 24, 2004 5:20 PM
> Subject: Asterisk-Users digest, Vol 1 #3886 - 9 msgs
>
>
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> >
> >
> > 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 <eric at fnords.org>
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [Asterisk-Users] Re: Making a SIP call
> > Reply-To: asterisk-users at lists.digium.com
> >
> > bclark at bwkip.com 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" <jay at skimmilk.net>
> > To: <asterisk-users at lists.digium.com>
> > Subject: RE: [Asterisk-Users] testing asterisk on FXS lines
> > Date: Mon, 24 May 2004 16:29:39 -0500
> > Reply-To: asterisk-users at lists.digium.com
> >
> > 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: asterisk-users-admin at lists.digium.com
> > [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Michael
> > George
> > Sent: Monday, May 24, 2004 3:57 PM
> > To: asterisk-users at lists.digium.com
> > 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
> > Asterisk-Users at lists.digium.com
> > 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 <chuck_ramirez at yahoo.com>
> > To: asterisk-users at lists.digium.com
> > Subject: [Asterisk-Users] SIP Authentication Problem
> > Reply-To: asterisk-users at lists.digium.com
> >
> > --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:mypass at mydomain/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/${EXTEN-1}@mydomain,tr
> >
> >
> >
> > When I dial the number 812345 from my SIP Phone, this is the message
> sequence
> > Phone -> Asterisk: INVITE sip:812345@*domain SIP/2.0
> > Asterisk -> Phone: SIP/2.0 407 Proxy Authentication Required
> > Phone -> Asterisk: ACK sip:812345@*domain SIP/2.0
> > Phone -> Asterisk: INVITE sip:812345@*domain SIP/2.0 (with authentication
> header)
> > Asterisk -> Phone: SIP/2.0 100 Trying
> > Asterisk -> Proxy: INVITE sip:12345 at mydomain SIP/2.0
> > Proxy -> Asterisk: SIP/2.0 401 Unauthorized
> > Asterisk -> Proxy: ACK sip:12345 at mydomain 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:mypass at mydomain/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/${EXTEN-1}@mydomain,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:812345@*domain
> SIP/2.0<BR>Asterisk -> Phone: SIP/2.0 407 Proxy Authentication
> Required<BR>Phone -> Asterisk: ACK sip:812345@*domain
> SIP/2.0<BR>Phone -> Asterisk: INVITE sip:812345@*domain SIP/2.0 (with
> authentication header)<BR>Asterisk -> Phone: SIP/2.0 100
> Trying<BR>Asterisk -> Proxy: INVITE sip:12345 at mydomain
> SIP/2.0<BR>Proxy -> Asterisk: SIP/2.0 401 Unauthorized<BR>Asterisk ->
> Proxy: ACK sip:12345 at mydomain 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" <chad.brown at identitymine.com>
> > To: <asterisk-users at lists.digium.com>
> > Reply-To: asterisk-users at lists.digium.com
> >
> > 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: asterisk-users-admin at lists.digium.com
> > [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Shaun Dawson
> > Sent: Monday, May 24, 2004 10:03 AM
> > To: asterisk-users at lists.digium.com
> > 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 <brucek at bagel.com> 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
> > > > Asterisk-Users at lists.digium.com
> > > >
> > >
> > 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
> > > Asterisk-Users at lists.digium.com
> > >
> > 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
> > Asterisk-Users at lists.digium.com
> > 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 <flavour at partyvibe.com>
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [Asterisk-Users] extensions/sip from database?
> > Reply-To: asterisk-users at lists.digium.com
> >
> > 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" <sfrazier at fraziercorp.com>
> > To: <asterisk-users at lists.digium.com>
> > Date: Mon, 24 May 2004 17:55:17 -0400
> > Subject: [Asterisk-Users] Using Blacklist
> > Reply-To: asterisk-users at lists.digium.com
> >
> > 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 <pesb at conexion.com.py>
> > To: asterisk-users at lists.digium.com
> > Date: Mon, 24 May 2004 17:58:30 -0400
> > Subject: [Asterisk-Users] Asterisk connected to DataBase
> > Reply-To: asterisk-users at lists.digium.com
> >
> > 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" <Simon.Brown at otterson.com.au>
> > To: <asterisk-users at lists.digium.com>
> > Subject: [Asterisk-Users] mpg123
> > Reply-To: asterisk-users at lists.digium.com
> >
> > 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 <asterisk at tintar.com>
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [Asterisk-Users] Using Blacklist
> > Reply-To: asterisk-users at lists.digium.com
> >
> > 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
> > > Asterisk-Users at lists.digium.com
> > > 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
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > End of Asterisk-Users Digest
> >
>
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