[Asterisk-Users] testing asterisk on FXS lines
Michael George
george at auroravideosys.com
Wed May 26 04:25:29 MST 2004
On May 24, 2004, at 10:34 PM, Adam Goryachev wrote:
> 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.
This is what I happened upon myself at the end of the day when I posted
the question. I like the cleanliness of the other two solutions better
but your response allowed me to learn more about the workings of
asterisk!
> (You will need this defined for your fxo interface as well later)
That is what I would expect, but the sample files I have, as well as
the one I am running, have "immediate=no" before the FXO or FXS lines
and does not change it. I'm thinking that the fxs_ks signalling must
override the immediate mode.
> 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>u
>> sernam
>> 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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>
> _______________________________________________
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>
-Michael
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