[Asterisk-Users] testing asterisk on FXS lines
Michael George
george at auroravideosys.com
Wed May 26 04:20:06 MST 2004
On May 24, 2004, at 8:46 PM, 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.
Hmm, very interesting idea. Similar to putting "misc." buttons on
applications when testing esoteric functionality.
Thanks for the tip!
> 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>us
> ernam
> 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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>
-Michael
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