[asterisk-users] RE: RTPTIMEOUT Configuration
Hello, I see the following descriptions on HYPERLINK http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+rtptimeoutrt ptimeout = Number : Number of seconds, to wait for RTP traffic before classify the connection as discontinued. Default 0 (no RTP timeout). (New in v1.2.x). HYPERLINK http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.confh ttp://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf HYPERLINK http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+rtptimeoutht tp://www.voip-info.org/wiki/index.php?page=Asterisk+sip+rtptimeout Valid only in [general] section and type=peer My question is, How can I set this variable RTPTIMEOUT for a given extension terminate the call after a pre-defined time of no RTP traffic I am trying to use like [general] rtptimeout=30 [1770] type=peer username=1770 secret=desrrr nat=yes context=brasil host=dynamic dtmfmode=rfc2833 rtptimeout=30 disallow=all allow=g729 allow=g723 allow=ulaw allow=ilbc [1771] type=peer username=1771 secret=alkmv nat=yes context=brasil host=dynamic dtmfmode=rfc2833 rtptimeout=30 disallow=all allow=g729 allow=g723 allow=ulaw allow=ilbc In this case, how can I set calls to terminate on extensions 1770 and 1771 after 30 seconds of no RTP traffic ??? Atenciosamente, __ Uniglobo Telecomunicacoes Ltda Diretoria Comercial - Newton Medina PBX 55-11-4082-3555 FAX 55-11-4581-9659 MSN [EMAIL PROTECTED] VISITE HYPERLINK file:///C:\\Documents%20and%20Settings\\Uniglobo%20Telecom\\Application %20Data\\Microsoft\\Signatures\\www.uniglobo.com.brwww.uniglobo.com.br -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.409 / Virus Database: 268.15.26/594 - Release Date: 12/20/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.409 / Virus Database: 268.15.26/597 - Release Date: 12/21/2006 ___ --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] RES: DISA Password Authenntication with Grandstream 488
Hi I can use now DISA settings like this one when I set E1 card connected directly to Asterisk. In this way every call dialed with pass 29 will be accepted. I have a billing which filters caller ID number and address calls to each account with same caller ID number previously set [frommt] exten = 1536,1,Answer exten = 1536,2,DigitTimeout(5) exten = 1536,3,ResponseTimeout(10) exten = 1536,4,Authenticate(29) exten = 1536,5,DISA(no-password|brasil) exten = 1536,6,Hangup Now I need to add a Grandstream 488 for DISA to remote landlines. So asterisk will receive phone number from the landline connected to this grandstream and also the sip account which is linked to Asterisk. But I cant decode caller phone number who dialed to the landline connected to asterisk. Is that possible with Asterisk to create a variable to collect a dialed password and then present that password which I can read it and then manipulate that pass ? Regards from Brazil Kind Regards, Diretoria Comercial - Newton Medina PABX 11.3085.1536 MSN[EMAIL PROTECTED] Rua Augusta 2.212 SL 26 Jardins 01412001 São Paulo - Brasil ___ --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] DISA Password Authenntication with Grandstream 488
Hi I can use now DISA settings like this one when I set E1 card connected directly to Asterisk. In this way every call dialed with pass 29 will be accepted. I have a billing which filters caller ID number and address calls to each account with same caller ID number previously set [frommt] exten = 1536,1,Answer exten = 1536,2,DigitTimeout(5) exten = 1536,3,ResponseTimeout(10) exten = 1536,4,Authenticate(29) exten = 1536,5,DISA(no-password|brasil) exten = 1536,6,Hangup Now I need to add a Grandstream 488 for DISA to remote landlines. So asterisk will receive phone number from the landline connected to this grandstream and also the sip account which is linked to Asterisk. But I cant decode caller phone number who dialed to the landline connected to asterisk. Is that possible with Asterisk to create a variable to collect a dialed password and then present that password which I can read it and then manipulate that pass ? Regards from Brazil Kind Regards, Diretoria Comercial - Newton Medina PABX 11.3085.1536 MSN[EMAIL PROTECTED] Rua Augusta 2.212 SL 26 Jardins 01412001 São Paulo - Brasil ___ --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
RES: [Asterisk-Users] ASTERISK DISA FOR INCOMING DID CALL
Hi
Tks for your info.
I can t set that
exten = s,1,Gotoif($[${CALLERIDNUM} = 1130851536 ]?10)
exten = s,2,Goto(from-pstn,s,1)
exten = s,10,disa(no-password,from-internal)
to work ok yet. I don t know
what are those contexts to (from-pstn) and(from-internal).
I cant complete disa calls
I am using
exten =
itn123456,1,Answer
exten =
itn123456,2,DigitTimeout(5)
exten =
itn123456,3,ResponseTimeout(10)
exten =
itn123456,4,Authenticate(3)
exten =
itn123456,5,DISA(no-password|hanna)
exten = .,6,Hangup
and this is working well but
I need to dial 3# as pass and I cant authenticate from 1130851536
only
Brazil Market is very booming.
I am searching good quality cellular calls to Brazil (any suggestion ).
What else you need to know
about brazil market ? Pls tell me [EMAIL PROTECTED]
ok
I am using also this settings
bellow but other CID numbers can also complete calls.
Do I need to use these REDS
too ?
exten =
itn123456,1,Gotoif($[${CALLERIDNUM} = 1130851536]?10)
exten =
itn123456,2,Answer
exten =
itn123456,3,DigitTimeout(5)
exten =
itn123456,4,ResponseTimeout(10)
exten =
itn123456,5,Authenticate(3)
exten =
itn123456,6,DISA(no-password|hanna)
exten = .,7,Hangup
Atenciosamente,
ITN Info
Newton Medina
PABX 11-3085-1536
-Mensagem original-
De: Tele Cost Price Reducer [mailto:[EMAIL PROTECTED]
Enviada em: sábado, 6 de maio de
2006 16:44
Para: [EMAIL PROTECTED]com.br
Cc: Asterisk Users Mailing List -
Non-Commercial Discussion
Assunto: Re: [Asterisk-Users]
ASTERISK DISA FOR INCOMING DID CALL
hi,
you
can try the following:
exten
= s,1,Gotoif($[${CALLERIDNUM} = 1130851536
]?10)
exten
= s,2,Goto(from-pstn,s,1)
exten = s,10,disa(no-password,from-internal)
it works for me.
if
you need further help, let me know.
BTW, i am very interested
in the Brazilian Market so i would like to get as much as possible info about
Brazil.
good
luck,
Mickey
On 5/6/06, ITN Info - 11-3898-0112 [EMAIL PROTECTED] wrote:
Hi,
I am trying
to create a situation where I call the DID number which is 1140636249 and I
receive a dial tone to dial. I d like also to autenticate the number
1130851536.
I can see
that asterisk decode this number but I dont know how to authenticate this
number only. This is what I am doing
Sip.conf
[globo]
type=friend
username=itn111
fromuser=itn111
secret=123456
insecure=very
host= globo.net.br
context=fromttt
fromdomain= globo.net.br
dtmfmode=rfc2833
disallow=all
allow=g729
register
= itn111:[EMAIL PROTECTED]:5060/itn111
where itn111
is the LOGIN for DID and the virtual extension for Extensions.conf file
Extensions.conf
[fromttt]
exten =
itn111,1,Dial(SIP/29650,60,Ttr)
exten =
itn111,2,Hangup()
This
settings above can can garantee that every call to 1140636249 goes to extension
29650. Do DID part is working ok.
Now I would
like to get a second dial tone when I call 1140636249 for asterisk DISA.
I d like
also to autenticate the number 1130851536 (caller number) and only this number
can receive the call
This is what
I am trying to do
exten =
itn111,1,Dial(SIP/29650,60,Ttr)
exten =
29650,2,DISA(no-password|brasil) ; I use no-password for this for now and
Brasil context
exten =
29650,3,Hangup()
or
exten =
itn111 ,1,DISA(no-password|brasil)
In sip show
channels I see
SIP/itn123456-cdfe� (fromgvt��� itn123456��� 1�� ) � Up DISA�
no-password|brasil
But there s
no dial tone. And I don t know how to authenticate this number 1130851536. I
see that asterisk collect this number
Can you pls
help me to do this settings ?
Atenciosamente
Diretoria
Comercial - Newton Medina
PABX
11.3085.1536
MSN [EMAIL PROTECTED]
Rua Augusta 2.212
SL 26 Jardins 01412001
S�o Paulo - Brasil
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[Asterisk-Users] ASTERISK DISA FOR INCOMING DID CALL
Hi, I am trying to create a situation where I call the DID number which is 1140636249 and I receive a dial tone to dial. I d like also to autenticate the number 1130851536. I can see that asterisk decode this number but I dont know how to authenticate this number only. This is what I am doing Sip.conf [globo] type=friend username=itn111 fromuser=itn111 secret=123456 insecure=very host= globo.net.br context=fromttt fromdomain= globo.net.br dtmfmode=rfc2833 disallow=all allow=g729 register = itn111:[EMAIL PROTECTED]:5060/itn111 where itn111 is the LOGIN for DID and the virtual extension for Extensions.conf file Extensions.conf [fromttt] exten = itn111,1,Dial(SIP/29650,60,Ttr) exten = itn111,2,Hangup() This settings above can can garantee that every call to 1140636249 goes to extension 29650. Do DID part is working ok. Now I would like to get a second dial tone when I call 1140636249 for asterisk DISA. I d like also to autenticate the number 1130851536 (caller number) and only this number can receive the call This is what I am trying to do exten = itn111,1,Dial(SIP/29650,60,Ttr) exten = 29650,2,DISA(no-password|brasil) ; I use no-password for this for now and Brasil context exten = 29650,3,Hangup() or exten = itn111,1,DISA(no-password|brasil) In sip show channels I see SIP/itn123456-cdfe (fromgvt itn123456 1 ) Up DISA no-password|brasil But there s no dial tone. And I don t know how to authenticate this number 1130851536. I see that asterisk collect this number Can you pls help me to do this settings ? Atenciosamente Diretoria Comercial - Newton Medina PABX 11.3085.1536 MSN[EMAIL PROTECTED] Rua Augusta 2.212 SL 26 Jardins 01412001 São Paulo - Brasil ___ --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] Incoming Asterisk SIP DID Calls
Hello All, I am using incoming DIDs for the first time. I ll very happy if someone can help me on that serttings ... I need to know how to answer calls from IP 200.123.123.1 with credentials abc123456:123456 and I d like to address to extention 29650 incoming calls from that number which is 1140636249. Also for out going calls I d like to use my own context as I use now. So I need to know how to add this incoming calls to extention 29650 keeping the existing out going dial plan Sip.conf register = abc123456:[EMAIL PROTECTED] [in-did] type=friend username=abc123456 fromuser=abc123456 secret=123456 host=200.123.123.1 fromdomain=200.123.123.1 context=from-mysipprovider port =5060 dtmfmode=rfc2833 disallow=all allow=g729 allow=g723 allow=ulaw Extensions.conf [from-mysipprovider] exten = 29650,1,Answer ; 29650 is the contact extension set on pap2 exten = 29650,2,Hangup Newton ___ --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] Frequently Showed Info Messages
Hi When Asterisk is running those messages are frequently showed Mar 2 17:12:17 NOTICE[14355]: frame.c:128 ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Mar 2 17:12:17 NOTICE[14355]: frame.c:128 ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Is that possible to disable Asterisk info messages ? If so .. what file can I edit in order to turn this off ? Regards Newton ___ --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] RES: RTP and Signalling
Hi, I need to send RTP from asterisk to one IP and signalling to another IP. In this case, can you help me to arrange that configuration on sip.conf [] type=friend username= secret= host= dtmfmode=rfc2833 disallow=all allow=g729 Atenciosamente Diretoria Comercial - Newton Medina PABX 11.3898.0112 Fax 11.38980112 MSN[EMAIL PROTECTED] Rua Augusta 2.212 SL 26 Jardins 01412001 São Paulo - Brasil ___ --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
RES: [Asterisk-Users] Billing - Disable accounts when balance gets0 value
[EMAIL PROTECTED] Simoni, Thank you for your copersation. If you need routes in Brazil I have very high quality ones ok... Atenciosamente Reduzimos ao mínimo a sua conta de Telefone Liguetel - ITN Info - 15 anos em Telecomunicações Diretoria Comercial - Newton Medina PABX11.3891.2434 Fax 11.38980112 msn [EMAIL PROTECTED] Rua Augusta 2.212 SL 26 Jardins 01412001 São Paulo - Brasil Visite a Loja www.liguetel.com.br ou www.liguetel.com e conheça produtos e serviços para reduzir definitivamente a sua conta de telefone. -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Simone Cittadini Enviada em: segunda-feira, 5 de setembro de 2005 05:54 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [Asterisk-Users] Billing - Disable accounts when balance gets0 value This billing is also able to set accounts balance and for each call. Now I need to disable accounts which balance gets a determined value. I was thinking on changing account pass for that specif account which we need to disable. And then in the sip.com reload info. Can you help me with new (new ways for doing so) or programing ideas too once billing server has not the same public IP than Asterisk server. I ll appreciate your comments ok. I use ser+radius to do authentication, this way I can disable users or groups of users in a standard way, without using tricks like changing passwords. (when your customer pays he expect to have the same password as before, have you saved it ? where ? in a safe way ?) radius has a mysql backend, so also no need to reload config files. Asterisk and radius share the same db, with some not-too-complex agi before the actual Dial you can do stuff like setting the call timeout based on the remaining credit, blocking the call if the credit is too much in the red, and so on... ___ --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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.18/89 - Release Date: 2/9/2005 ___ --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] (no subject)
Hi, I am Newton from Liguetel in Brazil. I have now a billing system based on SQLPostgress which is able to collect real time CDRs and present in a web site all the accounts and CDRs related to their calls. This billing is also able to set accounts balance and for each call balance goes down as calls are made. Now I need to disable accounts which balance gets a determined value. I was thinking on changing account pass for that specif account which we need to disable. And then in the sip.conf reload info. Can you help me with new (new ways for doing so) or programing ideas too once billing server has not the same public IP than Asterisk server. I ll appreciate your comments ok. Kind Regards Newton ___ --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
