Re: [Asterisk-Users] testing asterisk on FXS lines

Michael George Wed, 26 May 2004 04:37:04 -0700

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 -----
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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 <[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
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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
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
<P
align=left>[mydomain] type=peer host=myproxy context=sip us ernam e=myuser secret=mypass fromuser=myuser fromdomain=mydomain
<P
align=left>[user1] type=friend host=dynamic defaultip=default. IP<BR
username=user1 secret=secret1 dtmfmode=rfc2833 context=users ca
llerid="User 1" nat=yes
&nbsp;
Here is part of the content of extensions.conf.
;This part is working fine [sip] exten =&gt;
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 
<hr size=1>Do you Yahoo!? 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
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=09
=09
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Yahoo! Domains - Claim yours for only $14.70/year
http://smallbusiness.promotions.yahoo.com/offer=20
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--__--__--
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.
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Re: [Asterisk-Users] testing asterisk on FXS lines

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