Re: [asterisk-users] Multiple registrations to the same asterisk server
Marco Mouta escribió:
Hi ,
Please post here your extensions.conf in your central server only with
that i can figured out or at least try to help u.
Best regards,
Marco Mouta
On 8/15/06, * Juan Luis Moyano* <[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>> wrote:
Hi All, I have the following scenario: A central Asterisk server where
all the ATAs register themselves. This server runs Asterisk 1.2.5 and
ATAs are SPA-2002. So far everything is OK. Now I have another
location
where I want to connect 4 analog phones. I thought setting up 2
SPA-2002
but as I already have a TDM400P card and I want to use it, I had
configured asterisk 1.0.7 on the second machine. So far I can place
calls from the second server to any extension on the central
server. But
I cant get an ATA on the central server to reach an extension on the
second server. Please help me solve this situation. Thanks in advance.
Juan Luis Moyano
The configs are as follows:
Central Server
--
-sip.conf
[40019]
username=USER1
callerid=40019
type=friend
host=dynamic
secret=
mailbox=40019
accountcode=USER1
[40028]
username=USER2
callerid=40028
type=friend
host=dynamic
secret=
mailbox=40028
accountcode=USER2
[4]
username=USER3
callerid=4
type=friend
host=dynamic
secret=
mailbox=4
accountcode=USER3
[40023]
username=USER4
callerid=40023
type=friend
host=dynamic
secret=
mailbox=40023
accountcode=USER4
-extensions.conf
[clientes-sip]
exten => _4.,1,Macro(stdexten,SIP/${EXTEN},${EXTEN})
[macro-stdexten]
exten => s,1,Dial(${ARG1},30,Tr)
exten => s,2,Voicemail(u${ARG2})
exten => s,3,Hangup
exten => s,102,Voicemail(b${ARG2})
exten => s,103,Hangup
Second Server
-
-sip.conf
register => 40019:[EMAIL PROTECTED]/40019
register => 40028:[EMAIL PROTECTED]/40028
register => 4:[EMAIL PROTECTED]/4
register => 40023:[EMAIL PROTECTED]/40023
[40019]
type=friend
secret=
username=40019
host=10.32.1.16 <http://10.32.1.16>
insecure=very
[4]
type=friend
secret=
username=4
host=10.32.1.16 <http://10.32.1.16>
insecure=very
[40028]
type=friend
secret=
username=40028
host=10.32.1.16 <http://10.32.1.16>
insecure=very
[40023]
type=friend
secret=
username=40023
host= 10.32.1.16 <http://10.32.1.16>
insecure=very
-extensions.conf
[globals]
USER1=Zap/2
USER2=Zap/3
USER3=Zap/4
USER4=Zap/5
[extensions]
exten => 40019,1,Dial(${USER1})
exten => 40023,1,Dial(${USER2})
exten => 40028,1,Dial(${USER3})
exten => 4,1,Dial(${USER4})
[outbound]
exten => _.,1,Dial(SIP/[EMAIL PROTECTED])
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Com os melhores cumprimentos,
Marco Mouta
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[asterisk-users] Multiple registrations to the same asterisk server
Hi All, I have the following scenario: A central Asterisk server where
all the ATAs register themselves. This server runs Asterisk 1.2.5 and
ATAs are SPA-2002. So far everything is OK. Now I have another location
where I want to connect 4 analog phones. I thought setting up 2 SPA-2002
but as I already have a TDM400P card and I want to use it, I had
configured asterisk 1.0.7 on the second machine. So far I can place
calls from the second server to any extension on the central server. But
I cant get an ATA on the central server to reach an extension on the
second server. Please help me solve this situation. Thanks in advance.
Juan Luis Moyano
The configs are as follows:
Central Server
--
-sip.conf
[40019]
username=USER1
callerid=40019
type=friend
host=dynamic
secret=
mailbox=40019
accountcode=USER1
[40028]
username=USER2
callerid=40028
type=friend
host=dynamic
secret=
mailbox=40028
accountcode=USER2
[4]
username=USER3
callerid=4
type=friend
host=dynamic
secret=
mailbox=4
accountcode=USER3
[40023]
username=USER4
callerid=40023
type=friend
host=dynamic
secret=
mailbox=40023
accountcode=USER4
Second Server
-
-sip.conf
register => 40019:[EMAIL PROTECTED]/40019
register => 40028:[EMAIL PROTECTED]/40028
register => 4:[EMAIL PROTECTED]/4
register => 40023:[EMAIL PROTECTED]/40023
[40019]
type=friend
secret=
username=40019
host=10.32.1.16
insecure=very
[4]
type=friend
secret=
username=4
host=10.32.1.16
insecure=very
[40028]
type=friend
secret=
username=40028
host=10.32.1.16
insecure=very
[40023]
type=friend
secret=
username=40023
host=10.32.1.16
insecure=very
-extensions.conf
[globals]
USER1=Zap/2
USER2=Zap/3
USER3=Zap/4
USER4=Zap/5
[extensions]
exten => 40019,1,Dial(${USER1})
exten => 40023,1,Dial(${USER2})
exten => 40028,1,Dial(${USER3})
exten => 4,1,Dial(${USER4})
[outbound]
exten => _.,1,Dial(SIP/[EMAIL PROTECTED])
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[Asterisk-Users] Soekris net4801-50 + IAXY
Hi, I'm having an issue with a soekris net4801 board and a S101i "IAXy" device. When I connect a successfully provisioned IAXy directly via a crossover cable into an ethernet port of the soekris, the link led turns on orange so i'ts 10Mb and the activity led blinks like if there is some action going on but when I try 'tcpdump -nettti sis1' I see nothing going on, no received packets. When I plug a regular PC on the same ethernet port there I can see all the traffic going on. I'm really stuck on this one. Help me please! Regards. Juan Luis Moyano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Soekris net4801 and IAXy dhcp issue
Hi all, I have a Soekris net4801-50 board with OpenBSD 3.9 where I've
configured a dhcp server and tested it with a regular PC connected
directly via a crossover cable with success. The problem comes when I
try to connect my IAXy device instead of the PC. I can see with 'tcpdump
-nettti sis1' that the IAXy isn't sending any packets to the dhcp
server. I thought my IAXy was bad but then I configured a second dhcp
server with the exact same config file and the IAXy worked right out. So
I don't have a clue of what could be happening. Please shed me some
light on this issue. Thanks in advance.
Juan Luis Moyano
#cat /etc/dhcpd.conf
shared-network LOCAL-NET {
option domain-name "b-fon.com.ar";
option domain-name-servers 10.32.2.254, 200.69.193.1, 200.69.193.2;
subnet 10.32.2.0 netmask 255.255.255.0 {
option routers 10.32.2.254;
range 10.32.2.32 10.32.2.64;
}
}
# tcpdump -nettti sis1
tcpdump: listening on sis1, link-type EN10MB
^C
0 packets received by filter
0 packets dropped by kernel
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Re: [Asterisk-Users] Hardware dimensioning issues
Forgot to mention that ideally it will be handling about 20 concurrent users. As for the recording issue, what are you suggesting me to do? NFS? I've been told that it will make a HUGE impact on performance and voice quality. TIA. -- Juan Luis Moyano [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@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] Hardware dimensioning issues
Hi all, I'm about to set up an asterisk server to handle about 100 users. All of them will be using uLaw on IAX2 ATAs. Also the server must record all the calls on a local hard drive in GSM format. Eventually the server would be running Meetme for conferences. For timing and PSTN access it would have a TDM400P board. Could you please recommend me make and models of servers capable of accomplishing the mentioned requirements?? I was thinking of Dell PowerEdge 1850 servers but I don't know if its fully compatible with asterisk and the TDM boards. I appreciate your help. Thanks in advance. -- Juan Luis Moyano [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC
Ade Agbero wrote: Finally, We have lift off, a shaky one though. I deleted my Astcc.gi and replaced it with Darren's copy posted on his website and I have finally been able to get something recorded as BILLCOST. I got it working too here with Darren's astcc.agi. And billing as expected so finally It's working. It would be nice if someone could update the cvs with Darren's astcc.agi, because the current one doesn't work, even patched.. it gets worse. Thanks for your attention Darren! -- Juan Luis Moyano [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC
Darren Wiebe wrote:
Could you please post the output from the asterisk console when
astcc.agi crashes? I really would like to get this resolved.
Darren Wiebe
[EMAIL PROTECTED]
Darren here I post you the output from asterisk console and the mysql
daemon log. After hanging the phone the field inuse stays '1' and I get
no cdr record. I'm using the cvs astcc.agi with astcc.patch applied.
//ASTCC agi debug
-- Executing DeadAGI("Zap/2-1", "astcc.agi|11|615") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
AGI Tx >> agi_request: astcc.agi
AGI Tx >> agi_channel: Zap/2-1
AGI Tx >> agi_language: en
AGI Tx >> agi_type: Zap
AGI Tx >> agi_uniqueid: 1120221737.19
AGI Tx >> agi_callerid: "CMW" <11>
AGI Tx >> agi_dnid: unknown
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: highclient
AGI Tx >> agi_extension: 77615
AGI Tx >> agi_priority: 3
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode: CMW
AGI Tx >>
AGI Rx << ANSWER
AGI Tx >> 200 result=0
AGI Rx << STREAM FILE astcc-tone 0123456789
AGI Tx >> 200 result=0 endpos=11200
AGI Rx << STREAM FILE astcc-youhave 0123456789
AGI Tx >> 200 result=0 endpos=6400
AGI Rx << SAY NUMBER 25 0123456789
-- Playing 'digits/20' (language 'en')
-- Playing 'digits/5' (language 'en')
AGI Tx >> 200 result=0
AGI Rx << STREAM FILE astcc-dollars 0123456789
AGI Tx >> 200 result=0 endpos=7200
AGI Rx << STREAM FILE astcc-remaining 0123456789
AGI Tx >> 200 result=0 endpos=3360
AGI Rx << STREAM FILE astcc-willcost 0123456789
AGI Tx >> 200 result=0 endpos=14240
AGI Rx << SAY NUMBER 50 0123456789
-- Playing 'digits/50' (language 'en')
AGI Tx >> 200 result=0
AGI Rx << STREAM FILE astcc-perminute 0123456789
AGI Tx >> 200 result=0 endpos=14240
AGI Rx << STREAM FILE astcc-pleasewait 0123456789
AGI Tx >> 200 result=0 endpos=15840
AGI Rx << EXEC DIAL
IAX2/657XXX:[EMAIL PROTECTED]/615|30|HL(300:6:3) ""
-- AGI Script Executing Application: (DIAL) Options:
(IAX2/657XXX:[EMAIL PROTECTED]/615|30|HL(300:6:3))
-- Limit Data:
-- timelimit=300
-- play_warning=6
-- play_to_caller=yes
-- play_to_callee=no
-- warning_freq=3
-- start_sound=UNDEF
-- warning_sound=timeleft
-- end_sound=UNDEF
-- Called 657XXX:[EMAIL PROTECTED]/615
-- Call accepted by 65.39.205.121 (format ulaw)
-- Format for call is ulaw
-- IAX2/65.39.205.121:4569/1 is making progress passing it to Zap/2-1
-- IAX2/65.39.205.121:4569/1 answered Zap/2-1
-- Hungup 'IAX2/65.39.205.121:4569/1'
AGI Tx >> 200 result=-1
AGI Rx << GET VARIABLE ANSWEREDTIME
AGI Tx >> 200 result=1 (24)
AGI Rx << GET VARIABLE DIALSTATUS
AGI Tx >> 200 result=1 (ANSWER)
-- AGI Script astcc.agi completed, returning 0
//MYSQL
050701 12:54:42 120 Connect [EMAIL PROTECTED] on astcc
050701 12:54:44 120 Query SELECT * FROM cards WHERE number='11'
120 Query SELECT * FROM cards WHERE number='11'
120 Query SELECT * FROM cards WHERE number='11'
120 Query SELECT * FROM cards WHERE number='11'
120 Query UPDATE cards SET used='0' WHERE
number='11'
120 Query UPDATE cards SET inuse='1' WHERE
number='11'
050701 12:54:47 120 Query SELECT * FROM routes WHERE '615'
RLIKE pattern ORDER BY LENGTH(pattern) DESC
050701 12:54:53 120 Query SELECT * FROM cards WHERE number='11'
120 Query SELECT * FROM trunks WHERE name='FWD'
050701 12:55:18 120 Quit
--
Juan Luis Moyano
[EMAIL PROTECTED]
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Re: [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC
Ade Agbero wrote: > I tried using your working astcc.agi file instead of mine, but that > failed to work too. > Having the same issues here.. it seems astcc.agi is crashing. -- Juan Luis Moyano [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC
Darren Wiebe wrote: > As I said I would, I have posted some screen shots and my astcc database > dump in the wiki. Please see: > http://www.voip-info.org/tiki-index.php?page=ASTCCGuide for links to the > info. > > Darren Wiebe > [EMAIL PROTECTED] Darren, I'm very thankful you could take a look at the code and find that annoying bug that was turning us mad!! Also I see in the wiki pages you posted, that you've corrected the code in the cdrs table creation (cardnum as PRIMARY key), that was also very annoying. Right now I'm installing asterisk from scratch, so I couldn't apply the patch you submitted but as soon as I get it applied I'm posting my feedback. Again, thank you very much! PS: Do you really thought that I was switching to AreskiCC?? You fool..;) -- Juan Luis Moyano [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC
Ade Agbero wrote: > Almost a month working on BILLCOST problem (ASTCC does not charge for > calls) with no progress in sight. > > I hate to admit it, but I am about to give up on ASTCC. > > Zero charge for calls is not what a billing system should do regardless > of how many minutes successful call one makes. > > This is one frustrated Asterisk user. > This is another one!! I'm about to switch to ARESKI. It's a shame that we could'n get some support from ASTCC's developers. -- Juan Luis Moyano [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] ASTCC not billing
Also while looking at the cards table I noticed that the creation field on a card is always changing to the date I made the last call from that card. Is this the way it has to be? -- Juan Luis Moyano [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] ASTCC not billing
Has anyone noticed that the primary key in the cdrs table is cardnum? so it won't record more than the first call made by different cards. Perhaps I'm not understanding the purpose of de cdrs table. Maybe one solution is to add an auto_increment uniqueid field like in the asteriskcdrdb cdr table. Can anyone point me in the right direction on this one? -- Juan Luis Moyano [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] ASTCC not billing
Bernard Cresencia wrote:
> sorry, I meant my.cnf, not my.conf.
>
> Once logging is enabled, I would do tail -f
> /var/log/myslqd.log and watch as the database is being
> accessed during a call.
I've done what Bernard suggested and this is my output from mysql.log on
a successful call to number 612 on FWD. I'd like to know if any of you
see something wrong or rare. Thanks a lot.
Time Id CommandArgument
050629 1:02:02 1 Connect [EMAIL PROTECTED] on astcc
050629 1:02:04 1 Query SELECT * FROM cards WHERE number='21'
1 Query SELECT * FROM cards WHERE number='21'
1 Query SELECT * FROM cards WHERE number='21'
1 Query SELECT * FROM cards WHERE number='21'
1 Query UPDATE cards SET used='1801' WHERE
number='21'
1 Query UPDATE cards SET inuse='1' WHERE
number='21'
050629 1:02:10 1 Query SELECT * FROM routes WHERE '612'
RLIKE pattern ORDER BY LENGTH(pattern) DESC
050629 1:02:25 1 Query SELECT * FROM cards WHERE number='21'
1 Query SELECT * FROM trunks WHERE name='FWD'
050629 1:02:37 1 Query INSERT INTO cdrs
(cardnum,callerid,callednum,trunk,disposition,billseconds,billcost,callstart)
VALUES ('21', '\"Coco\" <21>', '612', 'FWD', 'ANSWER', '9', '150', 'Wed
Jun 29 01:02:37 2005')
1 Query UPDATE cards SET used='1951' WHERE
number='21'
1 Query UPDATE cards SET inuse='0' WHERE
number='21'
1 Query SELECT * FROM cards WHERE number='21'
1 Query UPDATE cards SET used='1951' WHERE
number='21'
1 Query UPDATE cards SET inuse='0' WHERE
number='21'
1 Quit
--
Juan Luis Moyano
[EMAIL PROTECTED]
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Re: [Asterisk-Users] ASTCC not billing
Darren Wiebe wrote: > We are using it on [EMAIL PROTECTED] version 1.1 > > Darren Wiebe > [EMAIL PROTECTED] > > > Darren Wiebe wrote: > Darren, could you post one working example of brands, trunks and routes table? Thanks in advance. -- Juan Luis Moyano [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] ASTCC not billing
On Sab, 25 de Junio de 2005, 5:20 pm, Darren Wiebe dijo: > Do you have the notransfer and reinvite lines set properly? I had this > same problem with ASTCC but found that if I removed asterisk including > the source and did a clean reinstall it worked suddenly. > > Darren > Darren, how is the proper way of setting notransfer and canreinvite lines on IAX. TIA. -- Juan Luis Moyano [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] ASTCC not making calls
jltaylor wrote:
> Doesn't the ASTCC require 12 digit pins?
No, the recording says enter 12 digit pin, but you can choose the length.
Darren Wiebe wrote:
> could you change this to ^4.* and see if that helps?
I've changed it and no luck.
///
Also, if I dial like this:
exten => _4.,1,Dial(IAX2/657XXX:[EMAIL PROTECTED]/${EXTEN:1},45,Ttm)
exten => _4.,2,Hangup()
I can connet succesfully to the desired number, but if I dial like this:
exten => _4.,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN:1})
exten => _4.,2,Hangup()
I have no success.
Still needing some help here. TIA.
--
Juan Luis Moyano
[EMAIL PROTECTED]
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[Asterisk-Users] ASTCC not making calls
Hi, im trying to setup ASTCC but I'm getting it difficult. I've
correctly set up the mysql database astcc and added a brand, trunk,
route and a card as follows:
brands
+--+--+--+--+--++--+--+
| name | language | inc | publishednum | did | markup | days | fee |
+--+--+--+--+--++--+--+
| FWD | es | 6| 4| 4| 0 | 30 |0 |
+--+--+--+--+--++--+--+
trunks
+--+--+-+
| name | tech | path|
+--+--+-+
| FWD | IAX2 | 657XXX:[EMAIL PROTECTED] |
+--+--+-+
routes
+-+---++-+-+--+
| pattern | comment | trunks | connectcost | includedseconds | cost |
+-+---++-+-+--+
| ^4. | FWD | FWD| 0 | 0 | 150 |
+-+---++-+-+--+
-Added a card with $25 credit, using 'FWD' brand.
extensions.conf
---
[outbound-fwd]
;
exten => _4.,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN:1})
exten => _4.,2,Hangup()
iax.conf
register => 657XXX:[EMAIL PROTECTED]
The problem is that when, for example, I dial '4612' i get:
-- Executing DeadAGI("IAX2/[EMAIL PROTECTED]/3", "astcc.agi|21|612") in new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
-- Playing 'digits/1' (language 'en')
-- AGI Script astcc.agi completed, returning 0
-- Executing Hangup("IAX2/[EMAIL PROTECTED]/3", "") in new stack
and i hear allison saying "I'm sorry that is not a recognized phone
number, goodbye".
Anyone knows what could be happening right here?
Many thanks in advance.
--
Juan Luis Moyano
[EMAIL PROTECTED]
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Re: [Asterisk-Users] ASTCC not making calls
Sorry 4 a.m. I'm kind of tired and I slipped a password. :S
Already changed it. Sorry!
Juan Luis Moyano wrote:
> Hi, im trying to setup ASTCC but I'm getting it difficult. I've
> correctly set up the mysql database astcc and added a brand, trunk,
> route and a card as follows:
>
> brands
> +--+--+--+--+--++--+--+
> | name | language | inc | publishednum | did | markup | days | fee |
> +--+--+--+--+--++--+--+
> | FWD | es | 6| 4| 4| 0 | 30 |0 |
> +--+--+--+--+--++--+--+
>
> trunks
> +--+--+-+
> | name | tech | path|
> +--+--+-+
> | FWD | IAX2 | 657XXX:[EMAIL PROTECTED] |
> +--+--+-+
>
> routes
> +-+---++-+-+--+
> | pattern | comment | trunks | connectcost | includedseconds | cost |
> +-+---++-+-+--+
> | 4. | FWD | FWD| 0 | 0 | 150 |
> +-+---++-+-+--+
>
> -Added a card with $1 credit and using 'FWD' brand.
>
> extensions.conf
> ---
> [outbound-fwd]
> ;
> exten => _4.,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN:1})
> exten => _4.,2,Hangup()
>
> iax.conf
>
> register => 657050:[EMAIL PROTECTED]
>
>
> The problem is that when, for example, I dial '4612' i get:
>
> -- Executing DeadAGI("IAX2/[EMAIL PROTECTED]/3", "astcc.agi|21|612") in new
> stack
> -- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
> -- Playing 'digits/1' (language 'en')
> -- AGI Script astcc.agi completed, returning 0
> -- Executing Hangup("IAX2/[EMAIL PROTECTED]/3", "") in new stack
>
> and i hear allison saying "I'm sorry that is not a recognized phone
> number, goodbye".
>
> Anyone knows what could be happening right here?
>
> Many thanks in advance.
>
--
Juan Luis Moyano
[EMAIL PROTECTED]
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[Asterisk-Users] ASTCC not making calls
Hi, im trying to setup ASTCC but I'm getting it difficult. I've
correctly set up the mysql database astcc and added a brand, trunk,
route and a card as follows:
brands
+--+--+--+--+--++--+--+
| name | language | inc | publishednum | did | markup | days | fee |
+--+--+--+--+--++--+--+
| FWD | es | 6| 4| 4| 0 | 30 |0 |
+--+--+--+--+--++--+--+
trunks
+--+--+-+
| name | tech | path|
+--+--+-+
| FWD | IAX2 | 657XXX:[EMAIL PROTECTED] |
+--+--+-+
routes
+-+---++-+-+--+
| pattern | comment | trunks | connectcost | includedseconds | cost |
+-+---++-+-+--+
| 4. | FWD | FWD| 0 | 0 | 150 |
+-+---++-+-+--+
-Added a card with $1 credit and using 'FWD' brand.
extensions.conf
---
[outbound-fwd]
;
exten => _4.,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN:1})
exten => _4.,2,Hangup()
iax.conf
register => 657050:[EMAIL PROTECTED]
The problem is that when, for example, I dial '4612' i get:
-- Executing DeadAGI("IAX2/[EMAIL PROTECTED]/3", "astcc.agi|21|612") in new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
-- Playing 'digits/1' (language 'en')
-- AGI Script astcc.agi completed, returning 0
-- Executing Hangup("IAX2/[EMAIL PROTECTED]/3", "") in new stack
and i hear allison saying "I'm sorry that is not a recognized phone
number, goodbye".
Anyone knows what could be happening right here?
Many thanks in advance.
--
Juan Luis Moyano
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Compilation Problem with asterisk-addons
On Lun, 20 de Junio de 2005, 6:49 pm, Nico Giefing dijo: > Hello, i have a little Problem with compiling asterisk-addons > > > the failure is: > > app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 > arguments, but only 3 given > app_addon_sql_mysql.c: In function `del_identifier': > app_addon_sql_mysql.c:164: error: ÀSR_LIST_REMOVE' undecalred (first use > in this function) > app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported > only once > app_addon_sql_mysql.c:164: error: for each function it appears in.) > > > Does anybody know anything about this problem? > > > Thank you for your help. > > > Nico > ___ > Asterisk-Users mailing list > Asterisk-Users@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 Nico, I'm having the same issue while compiling asterisk-addons. Here I post the error I get: app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 Also it's important to mention that I'm running asterisk-1.0.7 compiled from the ebuild on a Gentoo (kernel 2.6), also I've merged latest mysql, perl and DBD-mysql. I don't know what is the best way to compile asterisk-addons on a gentoo system so if someone had accomplished this, please let me know. Thanks in advance. -- Juan Luis Moyano [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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] MySQL to static .conf
Hello, I'm running asterisk 1.0.7 on a kernel 2.4 linux box. I have my SIP users database on MySQL, also I'm running asterisk from static .conf files. What I want to get done is that after a pre defined period of time, my SIP users table must be dumped into a .conf file. Perhaps with some script and a cron entry. Please shed me some light on this. Thanks in advance. -- Juan Luis Moyano [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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] MySQL to static .conf
Hello, I'm running asterisk 1.0.7 on a kernel 2.4 linux box. I have my SIP users database on MySQL, also I'm running asterisk from static .conf files. What I want to get done is that after a pre defined period of time, my SIP users table must be converted to a .conf file. Perhaps with a script and cron entry. Please shed me some light on this. Thanks in advance. -- Juan Luis Moyano [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Bill seconds
Americo Sanchez C. wrote: > > Hi all, > > We've installed Asterisk on a rural development project and we're > testing a prepaid phone service. Americo, what prepaid solution are you using at your asterisk box? Thanks in advance. -- Juan Luis Moyano [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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] NVFaxDetect on Gentoo
Hi, I've merged asterisk-0.9.0 on a gentoo (kernel 2.4) system using portage ebuilds. I've just got NVFaxDetect .c files from Justin Newman and I'm about to install them. I want to know which is the best way to accomplish this. Thanks in advance. -- Juan Luis Moyano [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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] NVFaxDetect on Gentoo
Hi, I've merged asterisk-0.9.0 on a gentoo (kernel 2.4) system using portage ebuilds. I've just got NVFaxDetect .c files from Justin Newman and I'm about to install them. I want to know which is the best way to accomplish this. Thanks in advance. -- Juan Luis Moyano [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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] IAX Dual Servers
Hello, I'm experiencing some problems while setting up my asterisk PBX.
What I want to get done is that every incoming call to SRV_A must be
routed to inbound context at SRV_B. That works fine actually, the only
thing is that if the called party stays on the phone and doesn't hang up
after the conversation has finished, the call between SRV_A and SRV_B
stays alive even if the calling party hung up. I attach my config files.
Thanks in advance.
--
Juan Luis Moyano
[EMAIL PROTECTED]
SRV_A extensions.conf
-
[general]
static=yes
writeprotect=yes
[inbound]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Dial(IAX2/SRV_A:[EMAIL PROTECTED]/[EMAIL PROTECTED],60,tr)
exten => s,4,Hangup()
exten => h,1,Hangup()
SRV_A iax.conf
--
[general]
;port=5036
;bindaddr=192.168.0.1
;iaxcompat=yes
delayreject=yes
amaflags=billing
;accountcode=lss0101
;language=en
bandwidth=high
;allow=all ; same as bandwidth=high
disallow=g723.1 ; Hm... Proprietary, don't use it...
disallow=lpc10 ; Icky sound quality... Mr. Roboto.
;allow=gsm ; Always allow GSM, it's cool :)
jitterbuffer=no
;dropcount=2^B
;maxjitterbuffer=500
;maxexcessbuffer=80
;minexcessbuffer=10
;jittershrinkrate=1
;trunkfreq=20 ; How frequently to send trunk msgs (in ms)
;authdebug=no
tos=lowdelay
;mailboxdetail=yes
[SRV_B]
type=user
host=192.168.1.69
auth=rsa
inkey=SRV_B
context=inbound
trunk=yes
[SRV_B]
type=peer
host=192.168.1.69
auth=rsa
outkey=SRV_A
trunk=yes
SRV_B extensions.conf
-
[inbound]
exten => s,1,Answer
exten => s,2,SetMusicOnHold(default)
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,10
exten => s,5,Background(vm-extension)
include => extensions
exten => h,1,Macro(hangup)
[macro-hangup]
exten => s,1,ResetCDR(w)
exten => s,2,NoCDR()
exten => s,3,Wait(1)
exten => s,4,Hangup()
[extensions]
exten => 11,1,Macro(stdexten,${INT1},${EXTEN})
SRV_B iax.conf
--
[general]
;port=5036
;bindaddr=192.168.0.1
;iaxcompat=yes
delayreject=yes
amaflags=billing
;accountcode=lss0101
;language=en
bandwidth=high
;allow=all ; same as bandwidth=high
disallow=g723.1 ; Hm... Proprietary, don't use it...
disallow=lpc10 ; Icky sound quality... Mr. Roboto.
;allow=gsm ; Always allow GSM, it's cool :)
jitterbuffer=no
;dropcount=2
;maxjitterbuffer=500
;maxexcessbuffer=80
;minexcessbuffer=10
;jittershrinkrate=1
;trunkfreq=20 ; How frequently to send trunk msgs (in ms)
;authdebug=no
tos=lowdelay
mailboxdetail=yes
[SRV_A]
type=user
host=192.168.1.72
auth=rsa
inkey=SRV_A
context=inbound
trunk=yes
[SRV_A]
type=peer
auth=rsa
outkey=SRV_B
host=192.168.1.72
trunk=yes
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[Asterisk-Users] Asterisk Dual Servers
Hi all, I am trying to set up two asterisk servers (SrvA and SrvB), and what I want to get done is that if I dial 1X on SrvB the call must be routed to extension X on SrvA and if I dial 2X on SrvA the call must be routed to extension X on SrvB. I've read the www.voip-info.org wiki abouta sterisk dual servers but couldn't succeed on get it working. Perhaps someone that has a working dialplan similar to what I want to do could post his config files or explain what to do. Thanks in advance. -- Juan Luis Moyano [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Unable to open master device '/dev/zap/ctl'
Sorry for the inconvenience, problem solved! I had left by mistake the defs26 USE flag enabled at /etc/portage/package.use, oops! Thanks for your help! -- Juan Luis Moyano Ben Hencke wrote: Does anything show up in /var/log/messages when you do the "modprobe zaptel" ? I get this line: Zapata Telephony Interface Registered on major 196 That may give you some hint as to why it is not loading. - Ben ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Unable to open master device '/dev/zap/ctl'
I don't get the /dev/zap* files created.. I think the problem is in the loading of the zaptel module: # modprobe zaptel (no messages) #/etc/init.d/zaptel stop * ERROR: "zaptel" has not yet been started. so the module isn't loading, perhaps the problem is in the zaptel instalation. Any suggestion? -- Juan Luis Moyano [EMAIL PROTECTED] Ben Hencke wrote: > Do you have /dev/zap* files? (not in subdirectory) > If so, maybe something is wrong with the udev configs. > If not, maybe the zaptel module is not loading. > > The /dev/zap/* files are created with the zaptel module before any card > specific module. > > Do you have any messages when starting the zaptel module? > /etc/init.d/zaptel stop > modprobe zaptel > > - Ben > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com <mailto:Asterisk-Users@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@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] Unable to open master device '/dev/zap/ctl'
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, I installed asterisk (1.0.7) on my Gentoo (2.6.11-gentoo-r3) box with udev support, also installed zaptel (1.0.7). I have a TDM31B correctly installed. My problem comes right after I modprobe the card and I execute 'ztcfg -vv', it gives me the following error: line 3: Unable to open master device '/dev/zap/ctl' It seems that the system is not creating the /zap devices. I made all the modifications to the /etc/udev/rules.d and permissions.d as stated on README.udev, and it is still not creating the devices. I don't know what else to do. Please shed me some light on this. Below I post the my /var/log/messages. Thanks in advance. Apr 7 22:34:14 vocero kernel: Module 0: Installed -- AUTO FXS/DPO Apr 7 22:34:14 vocero kernel: Module 1: Installed -- AUTO FXS/DPO Apr 7 22:34:14 vocero kernel: Module 2: Installed -- AUTO FXS/DPO Apr 7 22:34:14 vocero kernel: Module 3: Installed -- AUTO FXO (FCC mode) Apr 7 22:34:14 vocero kernel: Found a Wildcard TDM: Wildcard TDM400P REV H (4 modules) - -- Juan Luis Moyano [EMAIL PROTECTED] -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFCVqB6cpv/tMr+H20RAi4cAJ0dHIgPzCD0dKcANoOhYowQrmKRYACg1c8R 99VZ+yvJVki+/O6MfJ7/D2I= =2FGC -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@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
